Mailing List Archive: Cisco: VOIP

DTMF SIP to Verizon, wrong payload type...

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jonvoip at gmail

May 17, 2012, 10:01 AM

Post #1 of 20 (2060 views)
Permalink

DTMF SIP to Verizon, wrong payload type...

We have a SIP trunk to Verizon, Long Distance, Local and international work
fine, however, for toll free calls, DTMF does not function.

We are set to send RTP-NTE, but Verizon is saying that we are sending this:

a=rtpmap:101 X-NSE/8000

And it should be:

telephone-event/8000

And that is why it is failing.

What configuration change can we do to force it to send the right DTMF
method?

This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's request),
there is a software MTP and Transcoder on the router (both in use)...
Verizon says it is not their problem and closed their ticket.

Relevant SIP Config:

!
voice call send-alert
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip refer
redirect ip2ip
h323
h225 display-ie ccm-compatible
modem passthrough nse payload-type 101 codec g711ulaw
sip
bind media source-interface MFR1
early-offer forced
midcall-signaling passthru
!
!

dial-peer voice 800 voip
description OUTBOUND Voice SIP calls to VzB
destination-pattern 1800[2-9]......
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
no vad

!
sip-ua
retry invite 2
retry bye 2
retry cancel 2
registrar dns:verizonsipgateway expires 3600
sip-server dns:verizonsipgateway:5071
g729-annexb override
!

Jonathan

avholloway+cisco-voip at gmail

May 17, 2012, 10:49 AM

Post #2 of 20 (2008 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

I see you are setting EO = Forced on the CUBE, which the telco requires,
but are you using EO on the SIP trunk form CUCM to the CUBE? What is your
DTMF Signaling Method set to on that Trunk?

The only command I run which I can see is missing from your config is:

voice service voip
dtmf-interworking rtp-nte

But I'm not positive that's your problem.

-Anthony

On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <jonvoip [at] gmail>wrote:

> We have a SIP trunk to Verizon, Long Distance, Local and international
> work fine, however, for toll free calls, DTMF does not function.
>
> We are set to send RTP-NTE, but Verizon is saying that we are sending this:
>
> a=rtpmap:101 X-NSE/8000
>
> And it should be:
>
> telephone-event/8000
>
> And that is why it is failing.
>
>
>
> What configuration change can we do to force it to send the right DTMF
> method?
>
>
> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's request),
> there is a software MTP and Transcoder on the router (both in use)...
> Verizon says it is not their problem and closed their ticket.
>
> Relevant SIP Config:
>
>
> !
> voice call send-alert
> voice rtp send-recv
> !
> voice service voip
> allow-connections h323 to h323
> allow-connections h323 to sip
> allow-connections sip to h323
> allow-connections sip to sip
> no supplementary-service sip refer
> redirect ip2ip
> h323
> h225 display-ie ccm-compatible
> modem passthrough nse payload-type 101 codec g711ulaw
> sip
> bind media source-interface MFR1
> early-offer forced
> midcall-signaling passthru
> !
> !
>
> dial-peer voice 800 voip
> description OUTBOUND Voice SIP calls to VzB
> destination-pattern 1800[2-9]......
> voice-class sip dtmf-relay force rtp-nte
> session protocol sipv2
> session target sip-server
> incoming called-number .
> dtmf-relay rtp-nte
> codec g711ulaw
> no vad
>
>
> !
> sip-ua
> retry invite 2
> retry bye 2
> retry cancel 2
> registrar dns:verizonsipgateway expires 3600
> sip-server dns:verizonsipgateway:5071
> g729-annexb override
> !
>
>
> Jonathan
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

avholloway+cisco-voip at gmail

May 17, 2012, 10:51 AM

Post #3 of 20 (2011 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

Clarification.

there is a software MTP and Transcoder on the router

Does that mean you are using the "mode border-element" and
"telephony-service" commands on the CUBE to register the resources locally
for CUBE's use?

-Anthony

On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <jonvoip [at] gmail>wrote:

> We have a SIP trunk to Verizon, Long Distance, Local and international
> work fine, however, for toll free calls, DTMF does not function.
>
> We are set to send RTP-NTE, but Verizon is saying that we are sending this:
>
> a=rtpmap:101 X-NSE/8000
>
> And it should be:
>
> telephone-event/8000
>
> And that is why it is failing.
>
>
>
> What configuration change can we do to force it to send the right DTMF
> method?
>
>
> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's request),
> there is a software MTP and Transcoder on the router (both in use)...
> Verizon says it is not their problem and closed their ticket.
>
> Relevant SIP Config:
>
>
> !
> voice call send-alert
> voice rtp send-recv
> !
> voice service voip
> allow-connections h323 to h323
> allow-connections h323 to sip
> allow-connections sip to h323
> allow-connections sip to sip
> no supplementary-service sip refer
> redirect ip2ip
> h323
> h225 display-ie ccm-compatible
> modem passthrough nse payload-type 101 codec g711ulaw
> sip
> bind media source-interface MFR1
> early-offer forced
> midcall-signaling passthru
> !
> !
>
> dial-peer voice 800 voip
> description OUTBOUND Voice SIP calls to VzB
> destination-pattern 1800[2-9]......
> voice-class sip dtmf-relay force rtp-nte
> session protocol sipv2
> session target sip-server
> incoming called-number .
> dtmf-relay rtp-nte
> codec g711ulaw
> no vad
>
>
> !
> sip-ua
> retry invite 2
> retry bye 2
> retry cancel 2
> registrar dns:verizonsipgateway expires 3600
> sip-server dns:verizonsipgateway:5071
> g729-annexb override
> !
>
>
> Jonathan
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

miken at sisna

May 17, 2012, 11:07 AM

Post #4 of 20 (2009 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

Configuration examples and explanations for all of the primary North
America SIP providers can be found on this link. You need partner access to
view it.

http://www.cisco.com/en/US/partner/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html

Thank you
MikeN

On Thu, May 17, 2012 at 11:01 AM, Jonathan Charles <jonvoip [at] gmail>wrote:

> We have a SIP trunk to Verizon, Long Distance, Local and international
> work fine, however, for toll free calls, DTMF does not function.
>
> We are set to send RTP-NTE, but Verizon is saying that we are sending this:
>
> a=rtpmap:101 X-NSE/8000
>
> And it should be:
>
> telephone-event/8000
>
> And that is why it is failing.
>
>
>
> What configuration change can we do to force it to send the right DTMF
> method?
>
>
> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's request),
> there is a software MTP and Transcoder on the router (both in use)...
> Verizon says it is not their problem and closed their ticket.
>
> Relevant SIP Config:
>
>
> !
> voice call send-alert
> voice rtp send-recv
> !
> voice service voip
> allow-connections h323 to h323
> allow-connections h323 to sip
> allow-connections sip to h323
> allow-connections sip to sip
> no supplementary-service sip refer
> redirect ip2ip
> h323
> h225 display-ie ccm-compatible
> modem passthrough nse payload-type 101 codec g711ulaw
> sip
> bind media source-interface MFR1
> early-offer forced
> midcall-signaling passthru
> !
> !
>
> dial-peer voice 800 voip
> description OUTBOUND Voice SIP calls to VzB
> destination-pattern 1800[2-9]......
> voice-class sip dtmf-relay force rtp-nte
> session protocol sipv2
> session target sip-server
> incoming called-number .
> dtmf-relay rtp-nte
> codec g711ulaw
> no vad
>
>
> !
> sip-ua
> retry invite 2
> retry bye 2
> retry cancel 2
> registrar dns:verizonsipgateway expires 3600
> sip-server dns:verizonsipgateway:5071
> g729-annexb override
> !
>
>
> Jonathan
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

jonvoip at gmail

May 17, 2012, 11:10 AM

Post #5 of 20 (2010 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

No, it is in a MRGL used by everyone.

On Thu, May 17, 2012 at 12:51 PM, Anthony Holloway <
avholloway+cisco-voip [at] gmail> wrote:

> Clarification.
>
> there is a software MTP and Transcoder on the router
>
>
> Does that mean you are using the "mode border-element" and
> "telephony-service" commands on the CUBE to register the resources locally
> for CUBE's use?
>
> -Anthony
>
> On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>
>> We have a SIP trunk to Verizon, Long Distance, Local and international
>> work fine, however, for toll free calls, DTMF does not function.
>>
>> We are set to send RTP-NTE, but Verizon is saying that we are sending
>> this:
>>
>> a=rtpmap:101 X-NSE/8000
>>
>> And it should be:
>>
>> telephone-event/8000
>>
>> And that is why it is failing.
>>
>>
>>
>> What configuration change can we do to force it to send the right DTMF
>> method?
>>
>>
>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's request),
>> there is a software MTP and Transcoder on the router (both in use)...
>> Verizon says it is not their problem and closed their ticket.
>>
>> Relevant SIP Config:
>>
>>
>> !
>> voice call send-alert
>> voice rtp send-recv
>> !
>> voice service voip
>> allow-connections h323 to h323
>> allow-connections h323 to sip
>> allow-connections sip to h323
>> allow-connections sip to sip
>> no supplementary-service sip refer
>> redirect ip2ip
>> h323
>> h225 display-ie ccm-compatible
>> modem passthrough nse payload-type 101 codec g711ulaw
>> sip
>> bind media source-interface MFR1
>> early-offer forced
>> midcall-signaling passthru
>> !
>> !
>>
>> dial-peer voice 800 voip
>> description OUTBOUND Voice SIP calls to VzB
>> destination-pattern 1800[2-9]......
>> voice-class sip dtmf-relay force rtp-nte
>> session protocol sipv2
>> session target sip-server
>> incoming called-number .
>> dtmf-relay rtp-nte
>> codec g711ulaw
>> no vad
>>
>>
>> !
>> sip-ua
>> retry invite 2
>> retry bye 2
>> retry cancel 2
>> registrar dns:verizonsipgateway expires 3600
>> sip-server dns:verizonsipgateway:5071
>> g729-annexb override
>> !
>>
>>
>> Jonathan
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>

jonvoip at gmail

May 17, 2012, 11:10 AM

Post #6 of 20 (2006 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

I used that to configure the gateway originally... if you use it you will
break your network.

On Thu, May 17, 2012 at 1:07 PM, miken miken <miken [at] sisna> wrote:

> Configuration examples and explanations for all of the primary North
> America SIP providers can be found on this link. You need partner access to
> view it.
>
>
> http://www.cisco.com/en/US/partner/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html
>
> Thank you
> MikeN
>
> On Thu, May 17, 2012 at 11:01 AM, Jonathan Charles <jonvoip [at] gmail>wrote:
>
>> We have a SIP trunk to Verizon, Long Distance, Local and international
>> work fine, however, for toll free calls, DTMF does not function.
>>
>> We are set to send RTP-NTE, but Verizon is saying that we are sending
>> this:
>>
>> a=rtpmap:101 X-NSE/8000
>>
>> And it should be:
>>
>> telephone-event/8000
>>
>> And that is why it is failing.
>>
>>
>>
>> What configuration change can we do to force it to send the right DTMF
>> method?
>>
>>
>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's request),
>> there is a software MTP and Transcoder on the router (both in use)...
>> Verizon says it is not their problem and closed their ticket.
>>
>> Relevant SIP Config:
>>
>>
>> !
>> voice call send-alert
>> voice rtp send-recv
>> !
>> voice service voip
>> allow-connections h323 to h323
>> allow-connections h323 to sip
>> allow-connections sip to h323
>> allow-connections sip to sip
>> no supplementary-service sip refer
>> redirect ip2ip
>> h323
>> h225 display-ie ccm-compatible
>> modem passthrough nse payload-type 101 codec g711ulaw
>> sip
>> bind media source-interface MFR1
>> early-offer forced
>> midcall-signaling passthru
>> !
>> !
>>
>> dial-peer voice 800 voip
>> description OUTBOUND Voice SIP calls to VzB
>> destination-pattern 1800[2-9]......
>> voice-class sip dtmf-relay force rtp-nte
>> session protocol sipv2
>> session target sip-server
>> incoming called-number .
>> dtmf-relay rtp-nte
>> codec g711ulaw
>> no vad
>>
>>
>> !
>> sip-ua
>> retry invite 2
>> retry bye 2
>> retry cancel 2
>> registrar dns:verizonsipgateway expires 3600
>> sip-server dns:verizonsipgateway:5071
>> g729-annexb override
>> !
>>
>>
>> Jonathan
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>

jonvoip at gmail

May 17, 2012, 11:13 AM

Post #7 of 20 (2041 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

Added it, no change.

v=0
o=CiscoSystemsSIP-GW-UserAgent 2264 8655 IN IP4 157.130.97.178
s=SIP Call
c=IN IP4 1.1.1.1
t=0 0
m=audio 18130 RTP/AVP 0 101
c=IN IP4 157.130.97.178
a=rtpmap:0 PCMU/8000
a=rtpmap:101 X-NSE/8000 <------------- this needs to be
telephone-event/8000
a=fmtp:101 192-194
a=ptime:20

On Thu, May 17, 2012 at 12:49 PM, Anthony Holloway <
avholloway+cisco-voip [at] gmail> wrote:

> I see you are setting EO = Forced on the CUBE, which the telco requires,
> but are you using EO on the SIP trunk form CUCM to the CUBE? What is your
> DTMF Signaling Method set to on that Trunk?
>
> The only command I run which I can see is missing from your config is:
>
> voice service voip
> dtmf-interworking rtp-nte
>
> But I'm not positive that's your problem.
>
> -Anthony
>
> On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>
>> We have a SIP trunk to Verizon, Long Distance, Local and international
>> work fine, however, for toll free calls, DTMF does not function.
>>
>> We are set to send RTP-NTE, but Verizon is saying that we are sending
>> this:
>>
>> a=rtpmap:101 X-NSE/8000
>>
>> And it should be:
>>
>> telephone-event/8000
>>
>> And that is why it is failing.
>>
>>
>>
>> What configuration change can we do to force it to send the right DTMF
>> method?
>>
>>
>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's request),
>> there is a software MTP and Transcoder on the router (both in use)...
>> Verizon says it is not their problem and closed their ticket.
>>
>> Relevant SIP Config:
>>
>>
>> !
>> voice call send-alert
>> voice rtp send-recv
>> !
>> voice service voip
>> allow-connections h323 to h323
>> allow-connections h323 to sip
>> allow-connections sip to h323
>> allow-connections sip to sip
>> no supplementary-service sip refer
>> redirect ip2ip
>> h323
>> h225 display-ie ccm-compatible
>> modem passthrough nse payload-type 101 codec g711ulaw
>> sip
>> bind media source-interface MFR1
>> early-offer forced
>> midcall-signaling passthru
>> !
>> !
>>
>> dial-peer voice 800 voip
>> description OUTBOUND Voice SIP calls to VzB
>> destination-pattern 1800[2-9]......
>> voice-class sip dtmf-relay force rtp-nte
>> session protocol sipv2
>> session target sip-server
>> incoming called-number .
>> dtmf-relay rtp-nte
>> codec g711ulaw
>> no vad
>>
>>
>> !
>> sip-ua
>> retry invite 2
>> retry bye 2
>> retry cancel 2
>> registrar dns:verizonsipgateway expires 3600
>> sip-server dns:verizonsipgateway:5071
>> g729-annexb override
>> !
>>
>>
>> Jonathan
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>

avholloway+cisco-voip at gmail

May 17, 2012, 11:16 AM

Post #8 of 20 (2006 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

Small trick I learned a few years back: remove the "partner" from the URL
so non-partners can view it:

http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html

-Anthony

On Thu, May 17, 2012 at 1:07 PM, miken miken <miken [at] sisna> wrote:

> Configuration examples and explanations for all of the primary North
> America SIP providers can be found on this link. You need partner access to
> view it.
>
>
> http://www.cisco.com/en/US/partner/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html
>
> Thank you
> MikeN
>
> On Thu, May 17, 2012 at 11:01 AM, Jonathan Charles <jonvoip [at] gmail>wrote:
>
>> We have a SIP trunk to Verizon, Long Distance, Local and international
>> work fine, however, for toll free calls, DTMF does not function.
>>
>> We are set to send RTP-NTE, but Verizon is saying that we are sending
>> this:
>>
>> a=rtpmap:101 X-NSE/8000
>>
>> And it should be:
>>
>> telephone-event/8000
>>
>> And that is why it is failing.
>>
>>
>>
>> What configuration change can we do to force it to send the right DTMF
>> method?
>>
>>
>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's request),
>> there is a software MTP and Transcoder on the router (both in use)...
>> Verizon says it is not their problem and closed their ticket.
>>
>> Relevant SIP Config:
>>
>>
>> !
>> voice call send-alert
>> voice rtp send-recv
>> !
>> voice service voip
>> allow-connections h323 to h323
>> allow-connections h323 to sip
>> allow-connections sip to h323
>> allow-connections sip to sip
>> no supplementary-service sip refer
>> redirect ip2ip
>> h323
>> h225 display-ie ccm-compatible
>> modem passthrough nse payload-type 101 codec g711ulaw
>> sip
>> bind media source-interface MFR1
>> early-offer forced
>> midcall-signaling passthru
>> !
>> !
>>
>> dial-peer voice 800 voip
>> description OUTBOUND Voice SIP calls to VzB
>> destination-pattern 1800[2-9]......
>> voice-class sip dtmf-relay force rtp-nte
>> session protocol sipv2
>> session target sip-server
>> incoming called-number .
>> dtmf-relay rtp-nte
>> codec g711ulaw
>> no vad
>>
>>
>> !
>> sip-ua
>> retry invite 2
>> retry bye 2
>> retry cancel 2
>> registrar dns:verizonsipgateway expires 3600
>> sip-server dns:verizonsipgateway:5071
>> g729-annexb override
>> !
>>
>>
>> Jonathan
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

jonvoip at gmail

May 17, 2012, 11:16 AM

Post #9 of 20 (2003 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

I am a partner.

The Verizon config has a call policer on it that blocks all calls after 10.

On Thu, May 17, 2012 at 1:16 PM, Anthony Holloway <
avholloway+cisco-voip [at] gmail> wrote:

> Small trick I learned a few years back: remove the "partner" from the URL
> so non-partners can view it:
>
>
> http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html
>
> -Anthony
>
> On Thu, May 17, 2012 at 1:07 PM, miken miken <miken [at] sisna> wrote:
>
>> Configuration examples and explanations for all of the primary North
>> America SIP providers can be found on this link. You need partner access to
>> view it.
>>
>>
>> http://www.cisco.com/en/US/partner/solutions/ns340/ns414/ns728/networking_solutions_products_genericcontent0900aecd805bd13d.html
>>
>> Thank you
>> MikeN
>>
>> On Thu, May 17, 2012 at 11:01 AM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>
>>> We have a SIP trunk to Verizon, Long Distance, Local and international
>>> work fine, however, for toll free calls, DTMF does not function.
>>>
>>> We are set to send RTP-NTE, but Verizon is saying that we are sending
>>> this:
>>>
>>> a=rtpmap:101 X-NSE/8000
>>>
>>> And it should be:
>>>
>>> telephone-event/8000
>>>
>>> And that is why it is failing.
>>>
>>>
>>>
>>> What configuration change can we do to force it to send the right DTMF
>>> method?
>>>
>>>
>>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's request),
>>> there is a software MTP and Transcoder on the router (both in use)...
>>> Verizon says it is not their problem and closed their ticket.
>>>
>>> Relevant SIP Config:
>>>
>>>
>>> !
>>> voice call send-alert
>>> voice rtp send-recv
>>> !
>>> voice service voip
>>> allow-connections h323 to h323
>>> allow-connections h323 to sip
>>> allow-connections sip to h323
>>> allow-connections sip to sip
>>> no supplementary-service sip refer
>>> redirect ip2ip
>>> h323
>>> h225 display-ie ccm-compatible
>>> modem passthrough nse payload-type 101 codec g711ulaw
>>> sip
>>> bind media source-interface MFR1
>>> early-offer forced
>>> midcall-signaling passthru
>>> !
>>> !
>>>
>>> dial-peer voice 800 voip
>>> description OUTBOUND Voice SIP calls to VzB
>>> destination-pattern 1800[2-9]......
>>> voice-class sip dtmf-relay force rtp-nte
>>> session protocol sipv2
>>> session target sip-server
>>> incoming called-number .
>>> dtmf-relay rtp-nte
>>> codec g711ulaw
>>> no vad
>>>
>>>
>>> !
>>> sip-ua
>>> retry invite 2
>>> retry bye 2
>>> retry cancel 2
>>> registrar dns:verizonsipgateway expires 3600
>>> sip-server dns:verizonsipgateway:5071
>>> g729-annexb override
>>> !
>>>
>>>
>>> Jonathan
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip [at] puck
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>

avholloway+cisco-voip at gmail

May 17, 2012, 11:17 AM

Post #10 of 20 (2008 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

I'm glad you posted that.

The m= is the actual setting for that call. The a= are the available
settings. And you can see in the m=, you have codec 0 (g711) and DTMF 101
(telephony).

This looks correct.

-Anthony

On Thu, May 17, 2012 at 1:13 PM, Jonathan Charles <jonvoip [at] gmail> wrote:

> Added it, no change.
>
>
> v=0
> o=CiscoSystemsSIP-GW-UserAgent 2264 8655 IN IP4 157.130.97.178
> s=SIP Call
> c=IN IP4 1.1.1.1
> t=0 0
> m=audio 18130 RTP/AVP 0 101
> c=IN IP4 157.130.97.178
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 X-NSE/8000 <------------- this needs to be
> telephone-event/8000
> a=fmtp:101 192-194
> a=ptime:20
>
>
>
>
> On Thu, May 17, 2012 at 12:49 PM, Anthony Holloway <
> avholloway+cisco-voip [at] gmail> wrote:
>
>> I see you are setting EO = Forced on the CUBE, which the telco requires,
>> but are you using EO on the SIP trunk form CUCM to the CUBE? What is your
>> DTMF Signaling Method set to on that Trunk?
>>
>> The only command I run which I can see is missing from your config is:
>>
>> voice service voip
>> dtmf-interworking rtp-nte
>>
>> But I'm not positive that's your problem.
>>
>> -Anthony
>>
>> On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>
>>> We have a SIP trunk to Verizon, Long Distance, Local and international
>>> work fine, however, for toll free calls, DTMF does not function.
>>>
>>> We are set to send RTP-NTE, but Verizon is saying that we are sending
>>> this:
>>>
>>> a=rtpmap:101 X-NSE/8000
>>>
>>> And it should be:
>>>
>>> telephone-event/8000
>>>
>>> And that is why it is failing.
>>>
>>>
>>>
>>> What configuration change can we do to force it to send the right DTMF
>>> method?
>>>
>>>
>>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's request),
>>> there is a software MTP and Transcoder on the router (both in use)...
>>> Verizon says it is not their problem and closed their ticket.
>>>
>>> Relevant SIP Config:
>>>
>>>
>>> !
>>> voice call send-alert
>>> voice rtp send-recv
>>> !
>>> voice service voip
>>> allow-connections h323 to h323
>>> allow-connections h323 to sip
>>> allow-connections sip to h323
>>> allow-connections sip to sip
>>> no supplementary-service sip refer
>>> redirect ip2ip
>>> h323
>>> h225 display-ie ccm-compatible
>>> modem passthrough nse payload-type 101 codec g711ulaw
>>> sip
>>> bind media source-interface MFR1
>>> early-offer forced
>>> midcall-signaling passthru
>>> !
>>> !
>>>
>>> dial-peer voice 800 voip
>>> description OUTBOUND Voice SIP calls to VzB
>>> destination-pattern 1800[2-9]......
>>> voice-class sip dtmf-relay force rtp-nte
>>> session protocol sipv2
>>> session target sip-server
>>> incoming called-number .
>>> dtmf-relay rtp-nte
>>> codec g711ulaw
>>> no vad
>>>
>>>
>>> !
>>> sip-ua
>>> retry invite 2
>>> retry bye 2
>>> retry cancel 2
>>> registrar dns:verizonsipgateway expires 3600
>>> sip-server dns:verizonsipgateway:5071
>>> g729-annexb override
>>> !
>>>
>>>
>>> Jonathan
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip [at] puck
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>
>

jonvoip at gmail

May 17, 2012, 11:20 AM

Post #11 of 20 (2015 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

It is not.

Per Verizon tech:

Octet1058 SIP Message Body: SDP

--------------------------------------------------------------------------------
........ Header Field v=0
........ o=CiscoSystemsSIP-GW-UserAgent 794
632 IN IP4 1,1,1,1
........ s=SIP Call
........ c=IN IP4 1.1.1.1
........ t=0 0
........ m=audio 17176 RTP/AVP 0 101
........ c=IN IP4 1.1.1.1
........ a=rtpmap:0 PCMU/8000
........ a=rtpmap:101 X-NSE/8000 <-- should
be telephone-event/8000
........ a=fmtp:101 192-194
........ a=ptime:20

They say the problem is on our end, and since we are sending the wrong
DTMF, they are closing their ticket.

On Thu, May 17, 2012 at 1:17 PM, Anthony Holloway <
avholloway+cisco-voip [at] gmail> wrote:

> I'm glad you posted that.
>
> The m= is the actual setting for that call. The a= are the available
> settings. And you can see in the m=, you have codec 0 (g711) and DTMF 101
> (telephony).
>
> This looks correct.
>
> -Anthony
>
>
> On Thu, May 17, 2012 at 1:13 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>
>> Added it, no change.
>>
>>
>> v=0
>> o=CiscoSystemsSIP-GW-UserAgent 2264 8655 IN IP4 157.130.97.178
>> s=SIP Call
>> c=IN IP4 1.1.1.1
>> t=0 0
>> m=audio 18130 RTP/AVP 0 101
>> c=IN IP4 157.130.97.178
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:101 X-NSE/8000 <------------- this needs to be
>> telephone-event/8000
>> a=fmtp:101 192-194
>> a=ptime:20
>>
>>
>>
>>
>> On Thu, May 17, 2012 at 12:49 PM, Anthony Holloway <
>> avholloway+cisco-voip [at] gmail> wrote:
>>
>>> I see you are setting EO = Forced on the CUBE, which the telco requires,
>>> but are you using EO on the SIP trunk form CUCM to the CUBE? What is your
>>> DTMF Signaling Method set to on that Trunk?
>>>
>>> The only command I run which I can see is missing from your config is:
>>>
>>> voice service voip
>>> dtmf-interworking rtp-nte
>>>
>>> But I'm not positive that's your problem.
>>>
>>> -Anthony
>>>
>>> On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>>
>>>> We have a SIP trunk to Verizon, Long Distance, Local and international
>>>> work fine, however, for toll free calls, DTMF does not function.
>>>>
>>>> We are set to send RTP-NTE, but Verizon is saying that we are sending
>>>> this:
>>>>
>>>> a=rtpmap:101 X-NSE/8000
>>>>
>>>> And it should be:
>>>>
>>>> telephone-event/8000
>>>>
>>>> And that is why it is failing.
>>>>
>>>>
>>>>
>>>> What configuration change can we do to force it to send the right DTMF
>>>> method?
>>>>
>>>>
>>>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's
>>>> request), there is a software MTP and Transcoder on the router (both in
>>>> use)... Verizon says it is not their problem and closed their ticket.
>>>>
>>>> Relevant SIP Config:
>>>>
>>>>
>>>> !
>>>> voice call send-alert
>>>> voice rtp send-recv
>>>> !
>>>> voice service voip
>>>> allow-connections h323 to h323
>>>> allow-connections h323 to sip
>>>> allow-connections sip to h323
>>>> allow-connections sip to sip
>>>> no supplementary-service sip refer
>>>> redirect ip2ip
>>>> h323
>>>> h225 display-ie ccm-compatible
>>>> modem passthrough nse payload-type 101 codec g711ulaw
>>>> sip
>>>> bind media source-interface MFR1
>>>> early-offer forced
>>>> midcall-signaling passthru
>>>> !
>>>> !
>>>>
>>>> dial-peer voice 800 voip
>>>> description OUTBOUND Voice SIP calls to VzB
>>>> destination-pattern 1800[2-9]......
>>>> voice-class sip dtmf-relay force rtp-nte
>>>> session protocol sipv2
>>>> session target sip-server
>>>> incoming called-number .
>>>> dtmf-relay rtp-nte
>>>> codec g711ulaw
>>>> no vad
>>>>
>>>>
>>>> !
>>>> sip-ua
>>>> retry invite 2
>>>> retry bye 2
>>>> retry cancel 2
>>>> registrar dns:verizonsipgateway expires 3600
>>>> sip-server dns:verizonsipgateway:5071
>>>> g729-annexb override
>>>> !
>>>>
>>>>
>>>> Jonathan
>>>>
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip [at] puck
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>
>>>>
>>>
>>
>

avholloway+cisco-voip at gmail

May 17, 2012, 11:30 AM

Post #12 of 20 (2006 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

I'm pretty sure he's wrong.

From the RFC:

The connection (`c=') and attribute (`a=') information in the
> session-level section applies to all the media of that session unless
> overridden by connection information or an attribute of the same name
> in the media description.
>
> The media
> description starts with an `m=' line and continues to the next media
> description or end of the whole session description. In general,
> session-level values are the default for all media unless overridden
> by an equivalent media-level value.
>
>
http://www.ietf.org/rfc/rfc2327.txt

And here's a link to the Media Description definition:

http://tools.ietf.org/html/rfc4566#page-22

Here's a Stackoverflow discussion on the m= field, and it confirms that the
trailing 101 is the telehony event for DTMF:

http://stackoverflow.com/questions/2930288/sdp-media-field-format

Since you are a partner, I will look for a recent VoE session which also
talk about the same thing...looking now.

-Anthony

On Thu, May 17, 2012 at 1:20 PM, Jonathan Charles <jonvoip [at] gmail> wrote:

> It is not.
>
> Per Verizon tech:
>
> Octet1058 SIP Message Body: SDP
>
> --------------------------------------------------------------------------------
> ........ Header Field v=0
> ........ o=CiscoSystemsSIP-GW-UserAgent 794
> 632 IN IP4 1,1,1,1
> ........ s=SIP Call
> ........ c=IN IP4 1.1.1.1
> ........ t=0 0
> ........ m=audio 17176 RTP/AVP 0 101
> ........ c=IN IP4 1.1.1.1
> ........ a=rtpmap:0 PCMU/8000
> ........ a=rtpmap:101 X-NSE/8000 <-- should
> be telephone-event/8000
> ........ a=fmtp:101 192-194
> ........ a=ptime:20
>
> They say the problem is on our end, and since we are sending the wrong
> DTMF, they are closing their ticket.
>
>
>
>
> On Thu, May 17, 2012 at 1:17 PM, Anthony Holloway <
> avholloway+cisco-voip [at] gmail> wrote:
>
>> I'm glad you posted that.
>>
>> The m= is the actual setting for that call. The a= are the available
>> settings. And you can see in the m=, you have codec 0 (g711) and DTMF 101
>> (telephony).
>>
>> This looks correct.
>>
>> -Anthony
>>
>>
>> On Thu, May 17, 2012 at 1:13 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>
>>> Added it, no change.
>>>
>>>
>>> v=0
>>> o=CiscoSystemsSIP-GW-UserAgent 2264 8655 IN IP4 157.130.97.178
>>> s=SIP Call
>>> c=IN IP4 1.1.1.1
>>> t=0 0
>>> m=audio 18130 RTP/AVP 0 101
>>> c=IN IP4 157.130.97.178
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 X-NSE/8000 <------------- this needs to be
>>> telephone-event/8000
>>> a=fmtp:101 192-194
>>> a=ptime:20
>>>
>>>
>>>
>>>
>>> On Thu, May 17, 2012 at 12:49 PM, Anthony Holloway <
>>> avholloway+cisco-voip [at] gmail> wrote:
>>>
>>>> I see you are setting EO = Forced on the CUBE, which the telco
>>>> requires, but are you using EO on the SIP trunk form CUCM to the CUBE?
>>>> What is your DTMF Signaling Method set to on that Trunk?
>>>>
>>>> The only command I run which I can see is missing from your config is:
>>>>
>>>> voice service voip
>>>> dtmf-interworking rtp-nte
>>>>
>>>> But I'm not positive that's your problem.
>>>>
>>>> -Anthony
>>>>
>>>> On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>>>
>>>>> We have a SIP trunk to Verizon, Long Distance, Local and international
>>>>> work fine, however, for toll free calls, DTMF does not function.
>>>>>
>>>>> We are set to send RTP-NTE, but Verizon is saying that we are sending
>>>>> this:
>>>>>
>>>>> a=rtpmap:101 X-NSE/8000
>>>>>
>>>>> And it should be:
>>>>>
>>>>> telephone-event/8000
>>>>>
>>>>> And that is why it is failing.
>>>>>
>>>>>
>>>>>
>>>>> What configuration change can we do to force it to send the right DTMF
>>>>> method?
>>>>>
>>>>>
>>>>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's
>>>>> request), there is a software MTP and Transcoder on the router (both in
>>>>> use)... Verizon says it is not their problem and closed their ticket.
>>>>>
>>>>> Relevant SIP Config:
>>>>>
>>>>>
>>>>> !
>>>>> voice call send-alert
>>>>> voice rtp send-recv
>>>>> !
>>>>> voice service voip
>>>>> allow-connections h323 to h323
>>>>> allow-connections h323 to sip
>>>>> allow-connections sip to h323
>>>>> allow-connections sip to sip
>>>>> no supplementary-service sip refer
>>>>> redirect ip2ip
>>>>> h323
>>>>> h225 display-ie ccm-compatible
>>>>> modem passthrough nse payload-type 101 codec g711ulaw
>>>>> sip
>>>>> bind media source-interface MFR1
>>>>> early-offer forced
>>>>> midcall-signaling passthru
>>>>> !
>>>>> !
>>>>>
>>>>> dial-peer voice 800 voip
>>>>> description OUTBOUND Voice SIP calls to VzB
>>>>> destination-pattern 1800[2-9]......
>>>>> voice-class sip dtmf-relay force rtp-nte
>>>>> session protocol sipv2
>>>>> session target sip-server
>>>>> incoming called-number .
>>>>> dtmf-relay rtp-nte
>>>>> codec g711ulaw
>>>>> no vad
>>>>>
>>>>>
>>>>> !
>>>>> sip-ua
>>>>> retry invite 2
>>>>> retry bye 2
>>>>> retry cancel 2
>>>>> registrar dns:verizonsipgateway expires 3600
>>>>> sip-server dns:verizonsipgateway:5071
>>>>> g729-annexb override
>>>>> !
>>>>>
>>>>>
>>>>> Jonathan
>>>>>
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip [at] puck
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>>
>>>>
>>>
>>
>

avholloway+cisco-voip at gmail

May 17, 2012, 11:42 AM

Post #13 of 20 (2012 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

Here is the VoE I was talking about:

https://communities.cisco.com/docs/DOC-7823

Look towards the top for "VoE - CUBE SIP Trunking"

Then download the PDF, and goto page 90. The page is also discuss in the
Webex recording @ 1h 48m 55s. For those who cannot see this, it says:

“c” parameter identifies the IP
> address (20.1.1.1) that the peer
> device should send the media to
>

> “m” parameter identifies:
> the type of call (audio)
> port number for media (16950)
> payload type for the 1st
> preferred codec (18 for G729)
> dtmf (101 for RFC2833)
>

> “a’” parameter identifies all the
> codecs and other descriptors for this
> call leg

This VoE event is very informative. Hope that helps.

-Anthony

On Thu, May 17, 2012 at 1:20 PM, Jonathan Charles <jonvoip [at] gmail> wrote:

> It is not.
>
> Per Verizon tech:
>
> Octet1058 SIP Message Body: SDP
>
> --------------------------------------------------------------------------------
> ........ Header Field v=0
> ........ o=CiscoSystemsSIP-GW-UserAgent 794
> 632 IN IP4 1,1,1,1
> ........ s=SIP Call
> ........ c=IN IP4 1.1.1.1
> ........ t=0 0
> ........ m=audio 17176 RTP/AVP 0 101
> ........ c=IN IP4 1.1.1.1
> ........ a=rtpmap:0 PCMU/8000
> ........ a=rtpmap:101 X-NSE/8000 <-- should
> be telephone-event/8000
> ........ a=fmtp:101 192-194
> ........ a=ptime:20
>
> They say the problem is on our end, and since we are sending the wrong
> DTMF, they are closing their ticket.
>
>
>
>
> On Thu, May 17, 2012 at 1:17 PM, Anthony Holloway <
> avholloway+cisco-voip [at] gmail> wrote:
>
>> I'm glad you posted that.
>>
>> The m= is the actual setting for that call. The a= are the available
>> settings. And you can see in the m=, you have codec 0 (g711) and DTMF 101
>> (telephony).
>>
>> This looks correct.
>>
>> -Anthony
>>
>>
>> On Thu, May 17, 2012 at 1:13 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>
>>> Added it, no change.
>>>
>>>
>>> v=0
>>> o=CiscoSystemsSIP-GW-UserAgent 2264 8655 IN IP4 157.130.97.178
>>> s=SIP Call
>>> c=IN IP4 1.1.1.1
>>> t=0 0
>>> m=audio 18130 RTP/AVP 0 101
>>> c=IN IP4 157.130.97.178
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:101 X-NSE/8000 <------------- this needs to be
>>> telephone-event/8000
>>> a=fmtp:101 192-194
>>> a=ptime:20
>>>
>>>
>>>
>>>
>>> On Thu, May 17, 2012 at 12:49 PM, Anthony Holloway <
>>> avholloway+cisco-voip [at] gmail> wrote:
>>>
>>>> I see you are setting EO = Forced on the CUBE, which the telco
>>>> requires, but are you using EO on the SIP trunk form CUCM to the CUBE?
>>>> What is your DTMF Signaling Method set to on that Trunk?
>>>>
>>>> The only command I run which I can see is missing from your config is:
>>>>
>>>> voice service voip
>>>> dtmf-interworking rtp-nte
>>>>
>>>> But I'm not positive that's your problem.
>>>>
>>>> -Anthony
>>>>
>>>> On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>>>
>>>>> We have a SIP trunk to Verizon, Long Distance, Local and international
>>>>> work fine, however, for toll free calls, DTMF does not function.
>>>>>
>>>>> We are set to send RTP-NTE, but Verizon is saying that we are sending
>>>>> this:
>>>>>
>>>>> a=rtpmap:101 X-NSE/8000
>>>>>
>>>>> And it should be:
>>>>>
>>>>> telephone-event/8000
>>>>>
>>>>> And that is why it is failing.
>>>>>
>>>>>
>>>>>
>>>>> What configuration change can we do to force it to send the right DTMF
>>>>> method?
>>>>>
>>>>>
>>>>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's
>>>>> request), there is a software MTP and Transcoder on the router (both in
>>>>> use)... Verizon says it is not their problem and closed their ticket.
>>>>>
>>>>> Relevant SIP Config:
>>>>>
>>>>>
>>>>> !
>>>>> voice call send-alert
>>>>> voice rtp send-recv
>>>>> !
>>>>> voice service voip
>>>>> allow-connections h323 to h323
>>>>> allow-connections h323 to sip
>>>>> allow-connections sip to h323
>>>>> allow-connections sip to sip
>>>>> no supplementary-service sip refer
>>>>> redirect ip2ip
>>>>> h323
>>>>> h225 display-ie ccm-compatible
>>>>> modem passthrough nse payload-type 101 codec g711ulaw
>>>>> sip
>>>>> bind media source-interface MFR1
>>>>> early-offer forced
>>>>> midcall-signaling passthru
>>>>> !
>>>>> !
>>>>>
>>>>> dial-peer voice 800 voip
>>>>> description OUTBOUND Voice SIP calls to VzB
>>>>> destination-pattern 1800[2-9]......
>>>>> voice-class sip dtmf-relay force rtp-nte
>>>>> session protocol sipv2
>>>>> session target sip-server
>>>>> incoming called-number .
>>>>> dtmf-relay rtp-nte
>>>>> codec g711ulaw
>>>>> no vad
>>>>>
>>>>>
>>>>> !
>>>>> sip-ua
>>>>> retry invite 2
>>>>> retry bye 2
>>>>> retry cancel 2
>>>>> registrar dns:verizonsipgateway expires 3600
>>>>> sip-server dns:verizonsipgateway:5071
>>>>> g729-annexb override
>>>>> !
>>>>>
>>>>>
>>>>> Jonathan
>>>>>
>>>>> _______________________________________________
>>>>> cisco-voip mailing list
>>>>> cisco-voip [at] puck
>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>
>>>>>
>>>>
>>>
>>
>

jonvoip at gmail

May 17, 2012, 11:49 AM

Post #14 of 20 (2011 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

Even that example shows:

a=rtpmap:101 telephone-event/8000

Whereas I am seeing

a=rtpmap:101 X-NSE/8000

A: Why?
B: How do I change it?

On Thu, May 17, 2012 at 1:42 PM, Anthony Holloway <
avholloway+cisco-voip [at] gmail> wrote:

> Here is the VoE I was talking about:
>
> https://communities.cisco.com/docs/DOC-7823
>
> Look towards the top for "VoE - CUBE SIP Trunking"
>
> Then download the PDF, and goto page 90. The page is also discuss in the
> Webex recording @ 1h 48m 55s. For those who cannot see this, it says:
>
> �c� parameter identifies the IP
>> address (20.1.1.1) that the peer
>> device should send the media to
>>
>
>
>> �m� parameter identifies:
>> the type of call (audio)
>> port number for media (16950)
>> payload type for the 1st
>> preferred codec (18 for G729)
>> dtmf (101 for RFC2833)
>>
>
>
>> �a�� parameter identifies all the
>> codecs and other descriptors for this
>> call leg
>
>
> This VoE event is very informative. Hope that helps.
>
> -Anthony
>
> On Thu, May 17, 2012 at 1:20 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>
>> It is not.
>>
>> Per Verizon tech:
>>
>> Octet1058 SIP Message Body: SDP
>>
>> --------------------------------------------------------------------------------
>> ........ Header Field v=0
>> ........ o=CiscoSystemsSIP-GW-UserAgent 794
>> 632 IN IP4 1,1,1,1
>> ........ s=SIP Call
>> ........ c=IN IP4 1.1.1.1
>> ........ t=0 0
>> ........ m=audio 17176 RTP/AVP 0 101
>> ........ c=IN IP4 1.1.1.1
>> ........ a=rtpmap:0 PCMU/8000
>> ........ a=rtpmap:101 X-NSE/8000 <--
>> should be telephone-event/8000
>> ........ a=fmtp:101 192-194
>> ........ a=ptime:20
>>
>> They say the problem is on our end, and since we are sending the wrong
>> DTMF, they are closing their ticket.
>>
>>
>>
>>
>> On Thu, May 17, 2012 at 1:17 PM, Anthony Holloway <
>> avholloway+cisco-voip [at] gmail> wrote:
>>
>>> I'm glad you posted that.
>>>
>>> The m= is the actual setting for that call. The a= are the available
>>> settings. And you can see in the m=, you have codec 0 (g711) and DTMF 101
>>> (telephony).
>>>
>>> This looks correct.
>>>
>>> -Anthony
>>>
>>>
>>> On Thu, May 17, 2012 at 1:13 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>>
>>>> Added it, no change.
>>>>
>>>>
>>>> v=0
>>>> o=CiscoSystemsSIP-GW-UserAgent 2264 8655 IN IP4 157.130.97.178
>>>> s=SIP Call
>>>> c=IN IP4 1.1.1.1
>>>> t=0 0
>>>> m=audio 18130 RTP/AVP 0 101
>>>> c=IN IP4 157.130.97.178
>>>> a=rtpmap:0 PCMU/8000
>>>> a=rtpmap:101 X-NSE/8000 <------------- this needs to be
>>>> telephone-event/8000
>>>> a=fmtp:101 192-194
>>>> a=ptime:20
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, May 17, 2012 at 12:49 PM, Anthony Holloway <
>>>> avholloway+cisco-voip [at] gmail> wrote:
>>>>
>>>>> I see you are setting EO = Forced on the CUBE, which the telco
>>>>> requires, but are you using EO on the SIP trunk form CUCM to the CUBE?
>>>>> What is your DTMF Signaling Method set to on that Trunk?
>>>>>
>>>>> The only command I run which I can see is missing from your config is:
>>>>>
>>>>> voice service voip
>>>>> dtmf-interworking rtp-nte
>>>>>
>>>>> But I'm not positive that's your problem.
>>>>>
>>>>> -Anthony
>>>>>
>>>>> On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>>>>
>>>>>> We have a SIP trunk to Verizon, Long Distance, Local and
>>>>>> international work fine, however, for toll free calls, DTMF does not
>>>>>> function.
>>>>>>
>>>>>> We are set to send RTP-NTE, but Verizon is saying that we are sending
>>>>>> this:
>>>>>>
>>>>>> a=rtpmap:101 X-NSE/8000
>>>>>>
>>>>>> And it should be:
>>>>>>
>>>>>> telephone-event/8000
>>>>>>
>>>>>> And that is why it is failing.
>>>>>>
>>>>>>
>>>>>>
>>>>>> What configuration change can we do to force it to send the right
>>>>>> DTMF method?
>>>>>>
>>>>>>
>>>>>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's
>>>>>> request), there is a software MTP and Transcoder on the router (both in
>>>>>> use)... Verizon says it is not their problem and closed their ticket.
>>>>>>
>>>>>> Relevant SIP Config:
>>>>>>
>>>>>>
>>>>>> !
>>>>>> voice call send-alert
>>>>>> voice rtp send-recv
>>>>>> !
>>>>>> voice service voip
>>>>>> allow-connections h323 to h323
>>>>>> allow-connections h323 to sip
>>>>>> allow-connections sip to h323
>>>>>> allow-connections sip to sip
>>>>>> no supplementary-service sip refer
>>>>>> redirect ip2ip
>>>>>> h323
>>>>>> h225 display-ie ccm-compatible
>>>>>> modem passthrough nse payload-type 101 codec g711ulaw
>>>>>> sip
>>>>>> bind media source-interface MFR1
>>>>>> early-offer forced
>>>>>> midcall-signaling passthru
>>>>>> !
>>>>>> !
>>>>>>
>>>>>> dial-peer voice 800 voip
>>>>>> description OUTBOUND Voice SIP calls to VzB
>>>>>> destination-pattern 1800[2-9]......
>>>>>> voice-class sip dtmf-relay force rtp-nte
>>>>>> session protocol sipv2
>>>>>> session target sip-server
>>>>>> incoming called-number .
>>>>>> dtmf-relay rtp-nte
>>>>>> codec g711ulaw
>>>>>> no vad
>>>>>>
>>>>>>
>>>>>> !
>>>>>> sip-ua
>>>>>> retry invite 2
>>>>>> retry bye 2
>>>>>> retry cancel 2
>>>>>> registrar dns:verizonsipgateway expires 3600
>>>>>> sip-server dns:verizonsipgateway:5071
>>>>>> g729-annexb override
>>>>>> !
>>>>>>
>>>>>>
>>>>>> Jonathan
>>>>>>
>>>>>> _______________________________________________
>>>>>> cisco-voip mailing list
>>>>>> cisco-voip [at] puck
>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>

avholloway+cisco-voip at gmail

May 17, 2012, 11:58 AM

Post #15 of 20 (2023 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

Do you use EO in CUCM on the Trunk's SIP profile? And what is the DTMF
setting in CUCM on the trunk? And lastly, your MTP Required check box
setting on the trunk?

-Anthony

On Thu, May 17, 2012 at 1:49 PM, Jonathan Charles <jonvoip [at] gmail> wrote:

> Even that example shows:
>
> a=rtpmap:101 telephone-event/8000
>
> Whereas I am seeing
>
> a=rtpmap:101 X-NSE/8000
>
>
> A: Why?
> B: How do I change it?
>
> On Thu, May 17, 2012 at 1:42 PM, Anthony Holloway <
> avholloway+cisco-voip [at] gmail> wrote:
>
>> Here is the VoE I was talking about:
>>
>> https://communities.cisco.com/docs/DOC-7823
>>
>> Look towards the top for "VoE - CUBE SIP Trunking"
>>
>> Then download the PDF, and goto page 90. The page is also discuss in the
>> Webex recording @ 1h 48m 55s. For those who cannot see this, it says:
>>
>> “c” parameter identifies the IP
>>> address (20.1.1.1) that the peer
>>> device should send the media to
>>>
>>
>>
>>> “m” parameter identifies:
>>> the type of call (audio)
>>> port number for media (16950)
>>> payload type for the 1st
>>> preferred codec (18 for G729)
>>> dtmf (101 for RFC2833)
>>>
>>
>>
>>> “a’” parameter identifies all the
>>> codecs and other descriptors for this
>>> call leg
>>
>>
>> This VoE event is very informative. Hope that helps.
>>
>> -Anthony
>>
>> On Thu, May 17, 2012 at 1:20 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>
>>> It is not.
>>>
>>> Per Verizon tech:
>>>
>>> Octet1058 SIP Message Body: SDP
>>>
>>> --------------------------------------------------------------------------------
>>> ........ Header Field v=0
>>> ........ o=CiscoSystemsSIP-GW-UserAgent 794
>>> 632 IN IP4 1,1,1,1
>>> ........ s=SIP Call
>>> ........ c=IN IP4 1.1.1.1
>>> ........ t=0 0
>>> ........ m=audio 17176 RTP/AVP 0 101
>>> ........ c=IN IP4 1.1.1.1
>>> ........ a=rtpmap:0 PCMU/8000
>>> ........ a=rtpmap:101 X-NSE/8000 <--
>>> should be telephone-event/8000
>>> ........ a=fmtp:101 192-194
>>> ........ a=ptime:20
>>>
>>> They say the problem is on our end, and since we are sending the wrong
>>> DTMF, they are closing their ticket.
>>>
>>>
>>>
>>>
>>> On Thu, May 17, 2012 at 1:17 PM, Anthony Holloway <
>>> avholloway+cisco-voip [at] gmail> wrote:
>>>
>>>> I'm glad you posted that.
>>>>
>>>> The m= is the actual setting for that call. The a= are the available
>>>> settings. And you can see in the m=, you have codec 0 (g711) and DTMF 101
>>>> (telephony).
>>>>
>>>> This looks correct.
>>>>
>>>> -Anthony
>>>>
>>>>
>>>> On Thu, May 17, 2012 at 1:13 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>>>
>>>>> Added it, no change.
>>>>>
>>>>>
>>>>> v=0
>>>>> o=CiscoSystemsSIP-GW-UserAgent 2264 8655 IN IP4 157.130.97.178
>>>>> s=SIP Call
>>>>> c=IN IP4 1.1.1.1
>>>>> t=0 0
>>>>> m=audio 18130 RTP/AVP 0 101
>>>>> c=IN IP4 157.130.97.178
>>>>> a=rtpmap:0 PCMU/8000
>>>>> a=rtpmap:101 X-NSE/8000 <------------- this needs to be
>>>>> telephone-event/8000
>>>>> a=fmtp:101 192-194
>>>>> a=ptime:20
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Thu, May 17, 2012 at 12:49 PM, Anthony Holloway <
>>>>> avholloway+cisco-voip [at] gmail> wrote:
>>>>>
>>>>>> I see you are setting EO = Forced on the CUBE, which the telco
>>>>>> requires, but are you using EO on the SIP trunk form CUCM to the CUBE?
>>>>>> What is your DTMF Signaling Method set to on that Trunk?
>>>>>>
>>>>>> The only command I run which I can see is missing from your config is:
>>>>>>
>>>>>> voice service voip
>>>>>> dtmf-interworking rtp-nte
>>>>>>
>>>>>> But I'm not positive that's your problem.
>>>>>>
>>>>>> -Anthony
>>>>>>
>>>>>> On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <jonvoip [at] gmail
>>>>>> > wrote:
>>>>>>
>>>>>>> We have a SIP trunk to Verizon, Long Distance, Local and
>>>>>>> international work fine, however, for toll free calls, DTMF does not
>>>>>>> function.
>>>>>>>
>>>>>>> We are set to send RTP-NTE, but Verizon is saying that we are
>>>>>>> sending this:
>>>>>>>
>>>>>>> a=rtpmap:101 X-NSE/8000
>>>>>>>
>>>>>>> And it should be:
>>>>>>>
>>>>>>> telephone-event/8000
>>>>>>>
>>>>>>> And that is why it is failing.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> What configuration change can we do to force it to send the right
>>>>>>> DTMF method?
>>>>>>>
>>>>>>>
>>>>>>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's
>>>>>>> request), there is a software MTP and Transcoder on the router (both in
>>>>>>> use)... Verizon says it is not their problem and closed their ticket.
>>>>>>>
>>>>>>> Relevant SIP Config:
>>>>>>>
>>>>>>>
>>>>>>> !
>>>>>>> voice call send-alert
>>>>>>> voice rtp send-recv
>>>>>>> !
>>>>>>> voice service voip
>>>>>>> allow-connections h323 to h323
>>>>>>> allow-connections h323 to sip
>>>>>>> allow-connections sip to h323
>>>>>>> allow-connections sip to sip
>>>>>>> no supplementary-service sip refer
>>>>>>> redirect ip2ip
>>>>>>> h323
>>>>>>> h225 display-ie ccm-compatible
>>>>>>> modem passthrough nse payload-type 101 codec g711ulaw
>>>>>>> sip
>>>>>>> bind media source-interface MFR1
>>>>>>> early-offer forced
>>>>>>> midcall-signaling passthru
>>>>>>> !
>>>>>>> !
>>>>>>>
>>>>>>> dial-peer voice 800 voip
>>>>>>> description OUTBOUND Voice SIP calls to VzB
>>>>>>> destination-pattern 1800[2-9]......
>>>>>>> voice-class sip dtmf-relay force rtp-nte
>>>>>>> session protocol sipv2
>>>>>>> session target sip-server
>>>>>>> incoming called-number .
>>>>>>> dtmf-relay rtp-nte
>>>>>>> codec g711ulaw
>>>>>>> no vad
>>>>>>>
>>>>>>>
>>>>>>> !
>>>>>>> sip-ua
>>>>>>> retry invite 2
>>>>>>> retry bye 2
>>>>>>> retry cancel 2
>>>>>>> registrar dns:verizonsipgateway expires 3600
>>>>>>> sip-server dns:verizonsipgateway:5071
>>>>>>> g729-annexb override
>>>>>>> !
>>>>>>>
>>>>>>>
>>>>>>> Jonathan
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> cisco-voip mailing list
>>>>>>> cisco-voip [at] puck
>>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>

matthnick at gmail

May 17, 2012, 12:15 PM

Post #16 of 20 (2051 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

They're actually right.

Remove this:
modem passthrough nse payload-type 101 codec g711ulaw

Or change it to something similar like:
modem passthrough nse payload-type 100 codec g711ulaw

-nick

On Thu, May 17, 2012 at 2:58 PM, Anthony Holloway <
avholloway+cisco-voip [at] gmail> wrote:

> Do you use EO in CUCM on the Trunk's SIP profile? And what is the DTMF
> setting in CUCM on the trunk? And lastly, your MTP Required check box
> setting on the trunk?
>
> -Anthony
>
>
> On Thu, May 17, 2012 at 1:49 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>
>> Even that example shows:
>>
>> a=rtpmap:101 telephone-event/8000
>>
>> Whereas I am seeing
>>
>> a=rtpmap:101 X-NSE/8000
>>
>>
>> A: Why?
>> B: How do I change it?
>>
>> On Thu, May 17, 2012 at 1:42 PM, Anthony Holloway <
>> avholloway+cisco-voip [at] gmail> wrote:
>>
>>> Here is the VoE I was talking about:
>>>
>>> https://communities.cisco.com/docs/DOC-7823
>>>
>>> Look towards the top for "VoE - CUBE SIP Trunking"
>>>
>>> Then download the PDF, and goto page 90. The page is also discuss in
>>> the Webex recording @ 1h 48m 55s. For those who cannot see this, it says:
>>>
>>> �c� parameter identifies the IP
>>>> address (20.1.1.1) that the peer
>>>> device should send the media to
>>>>
>>>
>>>
>>>> �m� parameter identifies:
>>>> the type of call (audio)
>>>> port number for media (16950)
>>>> payload type for the 1st
>>>> preferred codec (18 for G729)
>>>> dtmf (101 for RFC2833)
>>>>
>>>
>>>
>>>> �a�� parameter identifies all the
>>>> codecs and other descriptors for this
>>>> call leg
>>>
>>>
>>> This VoE event is very informative. Hope that helps.
>>>
>>> -Anthony
>>>
>>> On Thu, May 17, 2012 at 1:20 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>>
>>>> It is not.
>>>>
>>>> Per Verizon tech:
>>>>
>>>> Octet1058 SIP Message Body: SDP
>>>>
>>>> --------------------------------------------------------------------------------
>>>> ........ Header Field v=0
>>>> ........ o=CiscoSystemsSIP-GW-UserAgent
>>>> 794 632 IN IP4 1,1,1,1
>>>> ........ s=SIP Call
>>>> ........ c=IN IP4 1.1.1.1
>>>> ........ t=0 0
>>>> ........ m=audio 17176 RTP/AVP 0 101
>>>> ........ c=IN IP4 1.1.1.1
>>>> ........ a=rtpmap:0 PCMU/8000
>>>> ........ a=rtpmap:101 X-NSE/8000 <--
>>>> should be telephone-event/8000
>>>> ........ a=fmtp:101 192-194
>>>> ........ a=ptime:20
>>>>
>>>> They say the problem is on our end, and since we are sending the wrong
>>>> DTMF, they are closing their ticket.
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, May 17, 2012 at 1:17 PM, Anthony Holloway <
>>>> avholloway+cisco-voip [at] gmail> wrote:
>>>>
>>>>> I'm glad you posted that.
>>>>>
>>>>> The m= is the actual setting for that call. The a= are the available
>>>>> settings. And you can see in the m=, you have codec 0 (g711) and DTMF 101
>>>>> (telephony).
>>>>>
>>>>> This looks correct.
>>>>>
>>>>> -Anthony
>>>>>
>>>>>
>>>>> On Thu, May 17, 2012 at 1:13 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>>>>
>>>>>> Added it, no change.
>>>>>>
>>>>>>
>>>>>> v=0
>>>>>> o=CiscoSystemsSIP-GW-UserAgent 2264 8655 IN IP4 157.130.97.178
>>>>>> s=SIP Call
>>>>>> c=IN IP4 1.1.1.1
>>>>>> t=0 0
>>>>>> m=audio 18130 RTP/AVP 0 101
>>>>>> c=IN IP4 157.130.97.178
>>>>>> a=rtpmap:0 PCMU/8000
>>>>>> a=rtpmap:101 X-NSE/8000 <------------- this needs to be
>>>>>> telephone-event/8000
>>>>>> a=fmtp:101 192-194
>>>>>> a=ptime:20
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, May 17, 2012 at 12:49 PM, Anthony Holloway <
>>>>>> avholloway+cisco-voip [at] gmail> wrote:
>>>>>>
>>>>>>> I see you are setting EO = Forced on the CUBE, which the telco
>>>>>>> requires, but are you using EO on the SIP trunk form CUCM to the CUBE?
>>>>>>> What is your DTMF Signaling Method set to on that Trunk?
>>>>>>>
>>>>>>> The only command I run which I can see is missing from your config
>>>>>>> is:
>>>>>>>
>>>>>>> voice service voip
>>>>>>> dtmf-interworking rtp-nte
>>>>>>>
>>>>>>> But I'm not positive that's your problem.
>>>>>>>
>>>>>>> -Anthony
>>>>>>>
>>>>>>> On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <
>>>>>>> jonvoip [at] gmail> wrote:
>>>>>>>
>>>>>>>> We have a SIP trunk to Verizon, Long Distance, Local and
>>>>>>>> international work fine, however, for toll free calls, DTMF does not
>>>>>>>> function.
>>>>>>>>
>>>>>>>> We are set to send RTP-NTE, but Verizon is saying that we are
>>>>>>>> sending this:
>>>>>>>>
>>>>>>>> a=rtpmap:101 X-NSE/8000
>>>>>>>>
>>>>>>>> And it should be:
>>>>>>>>
>>>>>>>> telephone-event/8000
>>>>>>>>
>>>>>>>> And that is why it is failing.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> What configuration change can we do to force it to send the right
>>>>>>>> DTMF method?
>>>>>>>>
>>>>>>>>
>>>>>>>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's
>>>>>>>> request), there is a software MTP and Transcoder on the router (both in
>>>>>>>> use)... Verizon says it is not their problem and closed their ticket.
>>>>>>>>
>>>>>>>> Relevant SIP Config:
>>>>>>>>
>>>>>>>>
>>>>>>>> !
>>>>>>>> voice call send-alert
>>>>>>>> voice rtp send-recv
>>>>>>>> !
>>>>>>>> voice service voip
>>>>>>>> allow-connections h323 to h323
>>>>>>>> allow-connections h323 to sip
>>>>>>>> allow-connections sip to h323
>>>>>>>> allow-connections sip to sip
>>>>>>>> no supplementary-service sip refer
>>>>>>>> redirect ip2ip
>>>>>>>> h323
>>>>>>>> h225 display-ie ccm-compatible
>>>>>>>> modem passthrough nse payload-type 101 codec g711ulaw
>>>>>>>> sip
>>>>>>>> bind media source-interface MFR1
>>>>>>>> early-offer forced
>>>>>>>> midcall-signaling passthru
>>>>>>>> !
>>>>>>>> !
>>>>>>>>
>>>>>>>> dial-peer voice 800 voip
>>>>>>>> description OUTBOUND Voice SIP calls to VzB
>>>>>>>> destination-pattern 1800[2-9]......
>>>>>>>> voice-class sip dtmf-relay force rtp-nte
>>>>>>>> session protocol sipv2
>>>>>>>> session target sip-server
>>>>>>>> incoming called-number .
>>>>>>>> dtmf-relay rtp-nte
>>>>>>>> codec g711ulaw
>>>>>>>> no vad
>>>>>>>>
>>>>>>>>
>>>>>>>> !
>>>>>>>> sip-ua
>>>>>>>> retry invite 2
>>>>>>>> retry bye 2
>>>>>>>> retry cancel 2
>>>>>>>> registrar dns:verizonsipgateway expires 3600
>>>>>>>> sip-server dns:verizonsipgateway:5071
>>>>>>>> g729-annexb override
>>>>>>>> !
>>>>>>>>
>>>>>>>>
>>>>>>>> Jonathan
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> cisco-voip mailing list
>>>>>>>> cisco-voip [at] puck
>>>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

tman701 at gmail

May 17, 2012, 1:05 PM

Post #17 of 20 (2014 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

I have a few CUBE's using Verizon SIP trunks and the one config that has
pretty much always worked for me is the following:

voice service voip
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol pass-through g711ulaw
h323
modem passthrough nse codec g711ulaw
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
early-offer forced
midcall-signaling passthru
!
!
dial-peer voice 313 voip
description description ** OUTBOUND VOICE SIP CALLS TO VZB FROM FLBOCA**
translation-profile outgoing DIGITSTRIP-FLBOC
destination-pattern 113T
voice-class codec 1
voice-class sip early-offer forced
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte digit-drop
no vad
!
!
sip-ua
retry invite 2
retry bye 2
retry cancel 2
retry options 5
sip-server ipv4:X.X.X.X:XXXX
g729-annexb override

It's not the prettiest but it works. Haven't had any issues with DTMF that
I know of using it this way.

Joel P

On Thu, May 17, 2012 at 3:15 PM, Nick Matthews <matthnick [at] gmail> wrote:

> They're actually right.
>
> Remove this:
> modem passthrough nse payload-type 101 codec g711ulaw
>
> Or change it to something similar like:
> modem passthrough nse payload-type 100 codec g711ulaw
>
> -nick
>
>
> On Thu, May 17, 2012 at 2:58 PM, Anthony Holloway <
> avholloway+cisco-voip [at] gmail> wrote:
>
>> Do you use EO in CUCM on the Trunk's SIP profile? And what is the DTMF
>> setting in CUCM on the trunk? And lastly, your MTP Required check box
>> setting on the trunk?
>>
>> -Anthony
>>
>>
>> On Thu, May 17, 2012 at 1:49 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>
>>> Even that example shows:
>>>
>>> a=rtpmap:101 telephone-event/8000
>>>
>>> Whereas I am seeing
>>>
>>> a=rtpmap:101 X-NSE/8000
>>>
>>>
>>> A: Why?
>>> B: How do I change it?
>>>
>>> On Thu, May 17, 2012 at 1:42 PM, Anthony Holloway <
>>> avholloway+cisco-voip [at] gmail> wrote:
>>>
>>>> Here is the VoE I was talking about:
>>>>
>>>> https://communities.cisco.com/docs/DOC-7823
>>>>
>>>> Look towards the top for "VoE - CUBE SIP Trunking"
>>>>
>>>> Then download the PDF, and goto page 90. The page is also discuss in
>>>> the Webex recording @ 1h 48m 55s. For those who cannot see this, it says:
>>>>
>>>> �c� parameter identifies the IP
>>>>> address (20.1.1.1) that the peer
>>>>> device should send the media to
>>>>>
>>>>
>>>>
>>>>> �m� parameter identifies:
>>>>> the type of call (audio)
>>>>> port number for media (16950)
>>>>> payload type for the 1st
>>>>> preferred codec (18 for G729)
>>>>> dtmf (101 for RFC2833)
>>>>>
>>>>
>>>>
>>>>> �a�� parameter identifies all the
>>>>> codecs and other descriptors for this
>>>>> call leg
>>>>
>>>>
>>>> This VoE event is very informative. Hope that helps.
>>>>
>>>> -Anthony
>>>>
>>>> On Thu, May 17, 2012 at 1:20 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>>>
>>>>> It is not.
>>>>>
>>>>> Per Verizon tech:
>>>>>
>>>>> Octet1058 SIP Message Body: SDP
>>>>>
>>>>> --------------------------------------------------------------------------------
>>>>> ........ Header Field v=0
>>>>> ........ o=CiscoSystemsSIP-GW-UserAgent
>>>>> 794 632 IN IP4 1,1,1,1
>>>>> ........ s=SIP Call
>>>>> ........ c=IN IP4 1.1.1.1
>>>>> ........ t=0 0
>>>>> ........ m=audio 17176 RTP/AVP 0 101
>>>>> ........ c=IN IP4 1.1.1.1
>>>>> ........ a=rtpmap:0 PCMU/8000
>>>>> ........ a=rtpmap:101 X-NSE/8000 <--
>>>>> should be telephone-event/8000
>>>>> ........ a=fmtp:101 192-194
>>>>> ........ a=ptime:20
>>>>>
>>>>> They say the problem is on our end, and since we are sending the wrong
>>>>> DTMF, they are closing their ticket.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Thu, May 17, 2012 at 1:17 PM, Anthony Holloway <
>>>>> avholloway+cisco-voip [at] gmail> wrote:
>>>>>
>>>>>> I'm glad you posted that.
>>>>>>
>>>>>> The m= is the actual setting for that call. The a= are the available
>>>>>> settings. And you can see in the m=, you have codec 0 (g711) and DTMF 101
>>>>>> (telephony).
>>>>>>
>>>>>> This looks correct.
>>>>>>
>>>>>> -Anthony
>>>>>>
>>>>>>
>>>>>> On Thu, May 17, 2012 at 1:13 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>>>>>
>>>>>>> Added it, no change.
>>>>>>>
>>>>>>>
>>>>>>> v=0
>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 2264 8655 IN IP4 157.130.97.178
>>>>>>> s=SIP Call
>>>>>>> c=IN IP4 1.1.1.1
>>>>>>> t=0 0
>>>>>>> m=audio 18130 RTP/AVP 0 101
>>>>>>> c=IN IP4 157.130.97.178
>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>> a=rtpmap:101 X-NSE/8000 <------------- this needs to
>>>>>>> be telephone-event/8000
>>>>>>> a=fmtp:101 192-194
>>>>>>> a=ptime:20
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Thu, May 17, 2012 at 12:49 PM, Anthony Holloway <
>>>>>>> avholloway+cisco-voip [at] gmail> wrote:
>>>>>>>
>>>>>>>> I see you are setting EO = Forced on the CUBE, which the telco
>>>>>>>> requires, but are you using EO on the SIP trunk form CUCM to the CUBE?
>>>>>>>> What is your DTMF Signaling Method set to on that Trunk?
>>>>>>>>
>>>>>>>> The only command I run which I can see is missing from your config
>>>>>>>> is:
>>>>>>>>
>>>>>>>> voice service voip
>>>>>>>> dtmf-interworking rtp-nte
>>>>>>>>
>>>>>>>> But I'm not positive that's your problem.
>>>>>>>>
>>>>>>>> -Anthony
>>>>>>>>
>>>>>>>> On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <
>>>>>>>> jonvoip [at] gmail> wrote:
>>>>>>>>
>>>>>>>>> We have a SIP trunk to Verizon, Long Distance, Local and
>>>>>>>>> international work fine, however, for toll free calls, DTMF does not
>>>>>>>>> function.
>>>>>>>>>
>>>>>>>>> We are set to send RTP-NTE, but Verizon is saying that we are
>>>>>>>>> sending this:
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 X-NSE/8000
>>>>>>>>>
>>>>>>>>> And it should be:
>>>>>>>>>
>>>>>>>>> telephone-event/8000
>>>>>>>>>
>>>>>>>>> And that is why it is failing.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> What configuration change can we do to force it to send the right
>>>>>>>>> DTMF method?
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's
>>>>>>>>> request), there is a software MTP and Transcoder on the router (both in
>>>>>>>>> use)... Verizon says it is not their problem and closed their ticket.
>>>>>>>>>
>>>>>>>>> Relevant SIP Config:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> !
>>>>>>>>> voice call send-alert
>>>>>>>>> voice rtp send-recv
>>>>>>>>> !
>>>>>>>>> voice service voip
>>>>>>>>> allow-connections h323 to h323
>>>>>>>>> allow-connections h323 to sip
>>>>>>>>> allow-connections sip to h323
>>>>>>>>> allow-connections sip to sip
>>>>>>>>> no supplementary-service sip refer
>>>>>>>>> redirect ip2ip
>>>>>>>>> h323
>>>>>>>>> h225 display-ie ccm-compatible
>>>>>>>>> modem passthrough nse payload-type 101 codec g711ulaw
>>>>>>>>> sip
>>>>>>>>> bind media source-interface MFR1
>>>>>>>>> early-offer forced
>>>>>>>>> midcall-signaling passthru
>>>>>>>>> !
>>>>>>>>> !
>>>>>>>>>
>>>>>>>>> dial-peer voice 800 voip
>>>>>>>>> description OUTBOUND Voice SIP calls to VzB
>>>>>>>>> destination-pattern 1800[2-9]......
>>>>>>>>> voice-class sip dtmf-relay force rtp-nte
>>>>>>>>> session protocol sipv2
>>>>>>>>> session target sip-server
>>>>>>>>> incoming called-number .
>>>>>>>>> dtmf-relay rtp-nte
>>>>>>>>> codec g711ulaw
>>>>>>>>> no vad
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> !
>>>>>>>>> sip-ua
>>>>>>>>> retry invite 2
>>>>>>>>> retry bye 2
>>>>>>>>> retry cancel 2
>>>>>>>>> registrar dns:verizonsipgateway expires 3600
>>>>>>>>> sip-server dns:verizonsipgateway:5071
>>>>>>>>> g729-annexb override
>>>>>>>>> !
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Jonathan
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> cisco-voip mailing list
>>>>>>>>> cisco-voip [at] puck
>>>>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

jonvoip at gmail

May 17, 2012, 1:58 PM

Post #18 of 20 (1985 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

That fixed it!

Yay....

Now, will faxing still work...

On Thu, May 17, 2012 at 2:15 PM, Nick Matthews <matthnick [at] gmail> wrote:

> They're actually right.
>
> Remove this:
> modem passthrough nse payload-type 101 codec g711ulaw
>
> Or change it to something similar like:
> modem passthrough nse payload-type 100 codec g711ulaw
>
> -nick
>
>
> On Thu, May 17, 2012 at 2:58 PM, Anthony Holloway <
> avholloway+cisco-voip [at] gmail> wrote:
>
>> Do you use EO in CUCM on the Trunk's SIP profile? And what is the DTMF
>> setting in CUCM on the trunk? And lastly, your MTP Required check box
>> setting on the trunk?
>>
>> -Anthony
>>
>>
>> On Thu, May 17, 2012 at 1:49 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>
>>> Even that example shows:
>>>
>>> a=rtpmap:101 telephone-event/8000
>>>
>>> Whereas I am seeing
>>>
>>> a=rtpmap:101 X-NSE/8000
>>>
>>>
>>> A: Why?
>>> B: How do I change it?
>>>
>>> On Thu, May 17, 2012 at 1:42 PM, Anthony Holloway <
>>> avholloway+cisco-voip [at] gmail> wrote:
>>>
>>>> Here is the VoE I was talking about:
>>>>
>>>> https://communities.cisco.com/docs/DOC-7823
>>>>
>>>> Look towards the top for "VoE - CUBE SIP Trunking"
>>>>
>>>> Then download the PDF, and goto page 90. The page is also discuss in
>>>> the Webex recording @ 1h 48m 55s. For those who cannot see this, it says:
>>>>
>>>> �c� parameter identifies the IP
>>>>> address (20.1.1.1) that the peer
>>>>> device should send the media to
>>>>>
>>>>
>>>>
>>>>> �m� parameter identifies:
>>>>> the type of call (audio)
>>>>> port number for media (16950)
>>>>> payload type for the 1st
>>>>> preferred codec (18 for G729)
>>>>> dtmf (101 for RFC2833)
>>>>>
>>>>
>>>>
>>>>> �a�� parameter identifies all the
>>>>> codecs and other descriptors for this
>>>>> call leg
>>>>
>>>>
>>>> This VoE event is very informative. Hope that helps.
>>>>
>>>> -Anthony
>>>>
>>>> On Thu, May 17, 2012 at 1:20 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>>>
>>>>> It is not.
>>>>>
>>>>> Per Verizon tech:
>>>>>
>>>>> Octet1058 SIP Message Body: SDP
>>>>>
>>>>> --------------------------------------------------------------------------------
>>>>> ........ Header Field v=0
>>>>> ........ o=CiscoSystemsSIP-GW-UserAgent
>>>>> 794 632 IN IP4 1,1,1,1
>>>>> ........ s=SIP Call
>>>>> ........ c=IN IP4 1.1.1.1
>>>>> ........ t=0 0
>>>>> ........ m=audio 17176 RTP/AVP 0 101
>>>>> ........ c=IN IP4 1.1.1.1
>>>>> ........ a=rtpmap:0 PCMU/8000
>>>>> ........ a=rtpmap:101 X-NSE/8000 <--
>>>>> should be telephone-event/8000
>>>>> ........ a=fmtp:101 192-194
>>>>> ........ a=ptime:20
>>>>>
>>>>> They say the problem is on our end, and since we are sending the wrong
>>>>> DTMF, they are closing their ticket.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Thu, May 17, 2012 at 1:17 PM, Anthony Holloway <
>>>>> avholloway+cisco-voip [at] gmail> wrote:
>>>>>
>>>>>> I'm glad you posted that.
>>>>>>
>>>>>> The m= is the actual setting for that call. The a= are the available
>>>>>> settings. And you can see in the m=, you have codec 0 (g711) and DTMF 101
>>>>>> (telephony).
>>>>>>
>>>>>> This looks correct.
>>>>>>
>>>>>> -Anthony
>>>>>>
>>>>>>
>>>>>> On Thu, May 17, 2012 at 1:13 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>>>>>
>>>>>>> Added it, no change.
>>>>>>>
>>>>>>>
>>>>>>> v=0
>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 2264 8655 IN IP4 157.130.97.178
>>>>>>> s=SIP Call
>>>>>>> c=IN IP4 1.1.1.1
>>>>>>> t=0 0
>>>>>>> m=audio 18130 RTP/AVP 0 101
>>>>>>> c=IN IP4 157.130.97.178
>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>> a=rtpmap:101 X-NSE/8000 <------------- this needs to
>>>>>>> be telephone-event/8000
>>>>>>> a=fmtp:101 192-194
>>>>>>> a=ptime:20
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Thu, May 17, 2012 at 12:49 PM, Anthony Holloway <
>>>>>>> avholloway+cisco-voip [at] gmail> wrote:
>>>>>>>
>>>>>>>> I see you are setting EO = Forced on the CUBE, which the telco
>>>>>>>> requires, but are you using EO on the SIP trunk form CUCM to the CUBE?
>>>>>>>> What is your DTMF Signaling Method set to on that Trunk?
>>>>>>>>
>>>>>>>> The only command I run which I can see is missing from your config
>>>>>>>> is:
>>>>>>>>
>>>>>>>> voice service voip
>>>>>>>> dtmf-interworking rtp-nte
>>>>>>>>
>>>>>>>> But I'm not positive that's your problem.
>>>>>>>>
>>>>>>>> -Anthony
>>>>>>>>
>>>>>>>> On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <
>>>>>>>> jonvoip [at] gmail> wrote:
>>>>>>>>
>>>>>>>>> We have a SIP trunk to Verizon, Long Distance, Local and
>>>>>>>>> international work fine, however, for toll free calls, DTMF does not
>>>>>>>>> function.
>>>>>>>>>
>>>>>>>>> We are set to send RTP-NTE, but Verizon is saying that we are
>>>>>>>>> sending this:
>>>>>>>>>
>>>>>>>>> a=rtpmap:101 X-NSE/8000
>>>>>>>>>
>>>>>>>>> And it should be:
>>>>>>>>>
>>>>>>>>> telephone-event/8000
>>>>>>>>>
>>>>>>>>> And that is why it is failing.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> What configuration change can we do to force it to send the right
>>>>>>>>> DTMF method?
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's
>>>>>>>>> request), there is a software MTP and Transcoder on the router (both in
>>>>>>>>> use)... Verizon says it is not their problem and closed their ticket.
>>>>>>>>>
>>>>>>>>> Relevant SIP Config:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> !
>>>>>>>>> voice call send-alert
>>>>>>>>> voice rtp send-recv
>>>>>>>>> !
>>>>>>>>> voice service voip
>>>>>>>>> allow-connections h323 to h323
>>>>>>>>> allow-connections h323 to sip
>>>>>>>>> allow-connections sip to h323
>>>>>>>>> allow-connections sip to sip
>>>>>>>>> no supplementary-service sip refer
>>>>>>>>> redirect ip2ip
>>>>>>>>> h323
>>>>>>>>> h225 display-ie ccm-compatible
>>>>>>>>> modem passthrough nse payload-type 101 codec g711ulaw
>>>>>>>>> sip
>>>>>>>>> bind media source-interface MFR1
>>>>>>>>> early-offer forced
>>>>>>>>> midcall-signaling passthru
>>>>>>>>> !
>>>>>>>>> !
>>>>>>>>>
>>>>>>>>> dial-peer voice 800 voip
>>>>>>>>> description OUTBOUND Voice SIP calls to VzB
>>>>>>>>> destination-pattern 1800[2-9]......
>>>>>>>>> voice-class sip dtmf-relay force rtp-nte
>>>>>>>>> session protocol sipv2
>>>>>>>>> session target sip-server
>>>>>>>>> incoming called-number .
>>>>>>>>> dtmf-relay rtp-nte
>>>>>>>>> codec g711ulaw
>>>>>>>>> no vad
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> !
>>>>>>>>> sip-ua
>>>>>>>>> retry invite 2
>>>>>>>>> retry bye 2
>>>>>>>>> retry cancel 2
>>>>>>>>> registrar dns:verizonsipgateway expires 3600
>>>>>>>>> sip-server dns:verizonsipgateway:5071
>>>>>>>>> g729-annexb override
>>>>>>>>> !
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Jonathan
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> cisco-voip mailing list
>>>>>>>>> cisco-voip [at] puck
>>>>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>

csvoip at googlemail

May 18, 2012, 3:39 AM

Post #19 of 20 (1976 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

I have faxing and dmtf working perfectly over Verizon using the following

!
voice service voip
address-hiding
dtmf-interworking rtp-nte
mode border-element
allow-connections sip to sip
redundancy
fax protocol pass-through g711ulaw
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
early-offer forced
midcall-signaling passthru

On 17 May 2012 21:58, Jonathan Charles <jonvoip [at] gmail> wrote:

> That fixed it!
>
> Yay....
>
> Now, will faxing still work...
>
> On Thu, May 17, 2012 at 2:15 PM, Nick Matthews <matthnick [at] gmail>wrote:
>
>> They're actually right.
>>
>> Remove this:
>> modem passthrough nse payload-type 101 codec g711ulaw
>>
>> Or change it to something similar like:
>> modem passthrough nse payload-type 100 codec g711ulaw
>>
>> -nick
>>
>>
>> On Thu, May 17, 2012 at 2:58 PM, Anthony Holloway <
>> avholloway+cisco-voip [at] gmail> wrote:
>>
>>> Do you use EO in CUCM on the Trunk's SIP profile? And what is the DTMF
>>> setting in CUCM on the trunk? And lastly, your MTP Required check box
>>> setting on the trunk?
>>>
>>> -Anthony
>>>
>>>
>>> On Thu, May 17, 2012 at 1:49 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>>
>>>> Even that example shows:
>>>>
>>>> a=rtpmap:101 telephone-event/8000
>>>>
>>>> Whereas I am seeing
>>>>
>>>> a=rtpmap:101 X-NSE/8000
>>>>
>>>>
>>>> A: Why?
>>>> B: How do I change it?
>>>>
>>>> On Thu, May 17, 2012 at 1:42 PM, Anthony Holloway <
>>>> avholloway+cisco-voip [at] gmail> wrote:
>>>>
>>>>> Here is the VoE I was talking about:
>>>>>
>>>>> https://communities.cisco.com/docs/DOC-7823
>>>>>
>>>>> Look towards the top for "VoE - CUBE SIP Trunking"
>>>>>
>>>>> Then download the PDF, and goto page 90. The page is also discuss in
>>>>> the Webex recording @ 1h 48m 55s. For those who cannot see this, it says:
>>>>>
>>>>> �c� parameter identifies the IP
>>>>>> address (20.1.1.1) that the peer
>>>>>> device should send the media to
>>>>>>
>>>>>
>>>>>
>>>>>> �m� parameter identifies:
>>>>>> the type of call (audio)
>>>>>> port number for media (16950)
>>>>>> payload type for the 1st
>>>>>> preferred codec (18 for G729)
>>>>>> dtmf (101 for RFC2833)
>>>>>>
>>>>>
>>>>>
>>>>>> �a�� parameter identifies all the
>>>>>> codecs and other descriptors for this
>>>>>> call leg
>>>>>
>>>>>
>>>>> This VoE event is very informative. Hope that helps.
>>>>>
>>>>> -Anthony
>>>>>
>>>>> On Thu, May 17, 2012 at 1:20 PM, Jonathan Charles <jonvoip [at] gmail>wrote:
>>>>>
>>>>>> It is not.
>>>>>>
>>>>>> Per Verizon tech:
>>>>>>
>>>>>> Octet1058 SIP Message Body: SDP
>>>>>>
>>>>>> --------------------------------------------------------------------------------
>>>>>> ........ Header Field v=0
>>>>>> ........ o=CiscoSystemsSIP-GW-UserAgent
>>>>>> 794 632 IN IP4 1,1,1,1
>>>>>> ........ s=SIP Call
>>>>>> ........ c=IN IP4 1.1.1.1
>>>>>> ........ t=0 0
>>>>>> ........ m=audio 17176 RTP/AVP 0 101
>>>>>> ........ c=IN IP4 1.1.1.1
>>>>>> ........ a=rtpmap:0 PCMU/8000
>>>>>> ........ a=rtpmap:101 X-NSE/8000 <--
>>>>>> should be telephone-event/8000
>>>>>> ........ a=fmtp:101 192-194
>>>>>> ........ a=ptime:20
>>>>>>
>>>>>> They say the problem is on our end, and since we are sending the
>>>>>> wrong DTMF, they are closing their ticket.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, May 17, 2012 at 1:17 PM, Anthony Holloway <
>>>>>> avholloway+cisco-voip [at] gmail> wrote:
>>>>>>
>>>>>>> I'm glad you posted that.
>>>>>>>
>>>>>>> The m= is the actual setting for that call. The a= are the
>>>>>>> available settings. And you can see in the m=, you have codec 0 (g711) and
>>>>>>> DTMF 101 (telephony).
>>>>>>>
>>>>>>> This looks correct.
>>>>>>>
>>>>>>> -Anthony
>>>>>>>
>>>>>>>
>>>>>>> On Thu, May 17, 2012 at 1:13 PM, Jonathan Charles <jonvoip [at] gmail
>>>>>>> > wrote:
>>>>>>>
>>>>>>>> Added it, no change.
>>>>>>>>
>>>>>>>>
>>>>>>>> v=0
>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 2264 8655 IN IP4 157.130.97.178
>>>>>>>> s=SIP Call
>>>>>>>> c=IN IP4 1.1.1.1
>>>>>>>> t=0 0
>>>>>>>> m=audio 18130 RTP/AVP 0 101
>>>>>>>> c=IN IP4 157.130.97.178
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>> a=rtpmap:101 X-NSE/8000 <------------- this needs to
>>>>>>>> be telephone-event/8000
>>>>>>>> a=fmtp:101 192-194
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Thu, May 17, 2012 at 12:49 PM, Anthony Holloway <
>>>>>>>> avholloway+cisco-voip [at] gmail> wrote:
>>>>>>>>
>>>>>>>>> I see you are setting EO = Forced on the CUBE, which the telco
>>>>>>>>> requires, but are you using EO on the SIP trunk form CUCM to the CUBE?
>>>>>>>>> What is your DTMF Signaling Method set to on that Trunk?
>>>>>>>>>
>>>>>>>>> The only command I run which I can see is missing from your config
>>>>>>>>> is:
>>>>>>>>>
>>>>>>>>> voice service voip
>>>>>>>>> dtmf-interworking rtp-nte
>>>>>>>>>
>>>>>>>>> But I'm not positive that's your problem.
>>>>>>>>>
>>>>>>>>> -Anthony
>>>>>>>>>
>>>>>>>>> On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <
>>>>>>>>> jonvoip [at] gmail> wrote:
>>>>>>>>>
>>>>>>>>>> We have a SIP trunk to Verizon, Long Distance, Local and
>>>>>>>>>> international work fine, however, for toll free calls, DTMF does not
>>>>>>>>>> function.
>>>>>>>>>>
>>>>>>>>>> We are set to send RTP-NTE, but Verizon is saying that we are
>>>>>>>>>> sending this:
>>>>>>>>>>
>>>>>>>>>> a=rtpmap:101 X-NSE/8000
>>>>>>>>>>
>>>>>>>>>> And it should be:
>>>>>>>>>>
>>>>>>>>>> telephone-event/8000
>>>>>>>>>>
>>>>>>>>>> And that is why it is failing.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> What configuration change can we do to force it to send the right
>>>>>>>>>> DTMF method?
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's
>>>>>>>>>> request), there is a software MTP and Transcoder on the router (both in
>>>>>>>>>> use)... Verizon says it is not their problem and closed their ticket.
>>>>>>>>>>
>>>>>>>>>> Relevant SIP Config:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> !
>>>>>>>>>> voice call send-alert
>>>>>>>>>> voice rtp send-recv
>>>>>>>>>> !
>>>>>>>>>> voice service voip
>>>>>>>>>> allow-connections h323 to h323
>>>>>>>>>> allow-connections h323 to sip
>>>>>>>>>> allow-connections sip to h323
>>>>>>>>>> allow-connections sip to sip
>>>>>>>>>> no supplementary-service sip refer
>>>>>>>>>> redirect ip2ip
>>>>>>>>>> h323
>>>>>>>>>> h225 display-ie ccm-compatible
>>>>>>>>>> modem passthrough nse payload-type 101 codec g711ulaw
>>>>>>>>>> sip
>>>>>>>>>> bind media source-interface MFR1
>>>>>>>>>> early-offer forced
>>>>>>>>>> midcall-signaling passthru
>>>>>>>>>> !
>>>>>>>>>> !
>>>>>>>>>>
>>>>>>>>>> dial-peer voice 800 voip
>>>>>>>>>> description OUTBOUND Voice SIP calls to VzB
>>>>>>>>>> destination-pattern 1800[2-9]......
>>>>>>>>>> voice-class sip dtmf-relay force rtp-nte
>>>>>>>>>> session protocol sipv2
>>>>>>>>>> session target sip-server
>>>>>>>>>> incoming called-number .
>>>>>>>>>> dtmf-relay rtp-nte
>>>>>>>>>> codec g711ulaw
>>>>>>>>>> no vad
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> !
>>>>>>>>>> sip-ua
>>>>>>>>>> retry invite 2
>>>>>>>>>> retry bye 2
>>>>>>>>>> retry cancel 2
>>>>>>>>>> registrar dns:verizonsipgateway expires 3600
>>>>>>>>>> sip-server dns:verizonsipgateway:5071
>>>>>>>>>> g729-annexb override
>>>>>>>>>> !
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Jonathan
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> cisco-voip mailing list
>>>>>>>>>> cisco-voip [at] puck
>>>>>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip [at] puck
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>>
>>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

matthnick at gmail

May 18, 2012, 9:22 AM

Post #20 of 20 (1988 views)
Permalink

Re: DTMF SIP to Verizon, wrong payload type... [In reply to]

A little more background information for the curious:

NSE - this is a cisco proprietary RTP packet type that we've used for
a long time to signal certain events from device to device.� Cisco Fax
Relay uses this (defunct), as well as modem passthrough.� Basically
two cisco TDM termination devices that need to switch to faxing or
modem.

So in theory it should only have an effect if your ISP supported Cisco
specific modem passthrough (not likely) or you had TDM ports on your
CUBE that were interoperating with other Cisco gateways.

I filed a bug on this a while back: CSCtc00564. It should warn you
when you do this, though I can't remember if it blocks the
configuration or not. You shouldn't be able to assign both DTMF and
modem passthrough to the same RTP payload type.

-nick

On Thu, May 17, 2012 at 4:58 PM, Jonathan Charles <jonvoip [at] gmail> wrote:
>
> That fixed it!
>
> Yay....
>
> Now, will faxing still work...
>
> On Thu, May 17, 2012 at 2:15 PM, Nick Matthews <matthnick [at] gmail> wrote:
>>
>> They're actually right.
>>
>> Remove this:
>> modem passthrough nse payload-type 101 codec g711ulaw
>>
>> Or change it to something similar like:
>> modem passthrough nse payload-type 100 codec g711ulaw
>>
>> -nick
>>
>>
>> On Thu, May 17, 2012 at 2:58 PM, Anthony Holloway <avholloway+cisco-voip [at] gmail> wrote:
>>>
>>> Do you use EO in CUCM on the Trunk's SIP profile? �And what is the DTMF setting in CUCM on the trunk? �And lastly, your MTP Required check box setting on the trunk?
>>>
>>> -Anthony
>>>
>>>
>>> On Thu, May 17, 2012 at 1:49 PM, Jonathan Charles <jonvoip [at] gmail> wrote:
>>>>
>>>> Even that example shows:
>>>>
>>>> a=rtpmap:101 telephone-event/8000
>>>>
>>>> Whereas I am seeing
>>>>
>>>> a=rtpmap:101 X-NSE/8000
>>>>
>>>>
>>>> A: Why?
>>>> B: How do I change it?
>>>>
>>>> On Thu, May 17, 2012 at 1:42 PM, Anthony Holloway <avholloway+cisco-voip [at] gmail> wrote:
>>>>>
>>>>> Here is the VoE I was talking about:
>>>>>
>>>>> https://communities.cisco.com/docs/DOC-7823
>>>>>
>>>>> Look towards the top for "VoE - CUBE SIP Trunking"
>>>>>
>>>>> Then download the PDF, and goto page 90. �The page is also discuss in the Webex recording @ 1h 48m 55s. �For those who cannot see this, it says:
>>>>>
>>>>>> �c� parameter identifies the IP
>>>>>> address (20.1.1.1) that the peer
>>>>>> device should send the media to
>>>>>
>>>>>
>>>>>>
>>>>>> �m� parameter identifies:
>>>>>> the type of call (audio)
>>>>>> port number for media (16950)
>>>>>> payload type for the 1st
>>>>>> preferred codec (18 for G729)
>>>>>> dtmf (101 for RFC2833)
>>>>>
>>>>>
>>>>>>
>>>>>> �a�� parameter identifies all the
>>>>>> codecs and other descriptors for this
>>>>>> call leg
>>>>>
>>>>>
>>>>> This VoE event is very informative. �Hope that helps.
>>>>>
>>>>> -Anthony
>>>>>
>>>>> On Thu, May 17, 2012 at 1:20 PM, Jonathan Charles <jonvoip [at] gmail> wrote:
>>>>>>
>>>>>> It is not.
>>>>>>
>>>>>> Per Verizon tech:
>>>>>>
>>>>>> � � �Octet1058 SIP Message Body: SDP
>>>>>> �� --------------------------------------------------------------------------------
>>>>>> �� ........� Header Field�� v=0
>>>>>> �� ........�� o=CiscoSystemsSIP-GW-UserAgent 794 632 IN IP4 1,1,1,1
>>>>>> �� ........�� s=SIP Call
>>>>>> �� ........�� c=IN IP4 1.1.1.1
>>>>>> �� ........�� t=0 0
>>>>>> �� ........�� m=audio 17176 RTP/AVP 0 101
>>>>>> �� ........�� c=IN IP4 1.1.1.1
>>>>>> �� ........�� a=rtpmap:0 PCMU/8000
>>>>>> �� ........�� a=rtpmap:101 X-NSE/8000�� <-- should be telephone-event/8000
>>>>>> �� ........�� a=fmtp:101 192-194
>>>>>> �� ........�� a=ptime:20
>>>>>>
>>>>>>
>>>>>> They say the problem is on our end, and since we are sending the wrong DTMF, they are closing their ticket.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, May 17, 2012 at 1:17 PM, Anthony Holloway <avholloway+cisco-voip [at] gmail> wrote:
>>>>>>>
>>>>>>> I'm glad you posted that.
>>>>>>>
>>>>>>> The m= is the actual setting for that call. �The a= are the available settings. �And you can see in the m=, you have codec 0 (g711) and DTMF 101 (telephony).
>>>>>>>
>>>>>>> This looks correct.
>>>>>>>
>>>>>>> -Anthony
>>>>>>>
>>>>>>>
>>>>>>> On Thu, May 17, 2012 at 1:13 PM, Jonathan Charles <jonvoip [at] gmail> wrote:
>>>>>>>>
>>>>>>>> Added it, no change.
>>>>>>>>
>>>>>>>>
>>>>>>>> v=0
>>>>>>>> o=CiscoSystemsSIP-GW-UserAgent 2264 8655 IN IP4 157.130.97.178
>>>>>>>> s=SIP Call
>>>>>>>> c=IN IP4 1.1.1.1
>>>>>>>> t=0 0
>>>>>>>> m=audio 18130 RTP/AVP 0 101
>>>>>>>> c=IN IP4 157.130.97.178
>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>> a=rtpmap:101 X-NSE/8000 � � � � � � � <------------- this needs to be�telephone-event/8000
>>>>>>>> a=fmtp:101 192-194
>>>>>>>> a=ptime:20
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Thu, May 17, 2012 at 12:49 PM, Anthony Holloway <avholloway+cisco-voip [at] gmail> wrote:
>>>>>>>>>
>>>>>>>>> I see you are setting EO = Forced on the CUBE, which the telco requires, but are you using EO on the SIP trunk form CUCM to the CUBE? �What is your DTMF Signaling Method set to on that Trunk?
>>>>>>>>>
>>>>>>>>> The only command I run which I can see is missing from your config is:
>>>>>>>>>
>>>>>>>>> voice service voip
>>>>>>>>> ��dtmf-interworking rtp-nte
>>>>>>>>>
>>>>>>>>> But I'm not positive that's your problem.
>>>>>>>>>
>>>>>>>>> -Anthony
>>>>>>>>>
>>>>>>>>> On Thu, May 17, 2012 at 12:01 PM, Jonathan Charles <jonvoip [at] gmail> wrote:
>>>>>>>>>>
>>>>>>>>>> We have a SIP trunk to Verizon, Long Distance, Local and international work fine, however, for toll free calls, DTMF does not function.
>>>>>>>>>>
>>>>>>>>>> We are set to send RTP-NTE, but Verizon is saying that we are sending this:
>>>>>>>>>>
>>>>>>>>>> a=rtpmap:101 X-NSE/8000
>>>>>>>>>>
>>>>>>>>>> And it should be:
>>>>>>>>>>
>>>>>>>>>> telephone-event/8000
>>>>>>>>>>
>>>>>>>>>> And that is why it is failing.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> What configuration change can we do to force it to send the right DTMF method?
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> This is on a Cisco 3825 CUBE running 12.2.20.T4 (per Verizon's request), there is a software MTP and Transcoder on the router (both in use)... Verizon says it is not their problem and closed their ticket.
>>>>>>>>>>
>>>>>>>>>> Relevant SIP Config:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> !
>>>>>>>>>> voice call send-alert
>>>>>>>>>> voice rtp send-recv
>>>>>>>>>> !
>>>>>>>>>> voice service voip
>>>>>>>>>> �allow-connections h323 to h323
>>>>>>>>>> �allow-connections h323 to sip
>>>>>>>>>> �allow-connections sip to h323
>>>>>>>>>> �allow-connections sip to sip
>>>>>>>>>> �no supplementary-service sip refer
>>>>>>>>>> �redirect ip2ip
>>>>>>>>>> �h323
>>>>>>>>>> � h225 display-ie ccm-compatible
>>>>>>>>>> �modem passthrough nse payload-type 101 codec g711ulaw
>>>>>>>>>> �sip
>>>>>>>>>> � bind media source-interface MFR1
>>>>>>>>>> � early-offer forced
>>>>>>>>>> � midcall-signaling passthru
>>>>>>>>>> !
>>>>>>>>>> !
>>>>>>>>>>
>>>>>>>>>> dial-peer voice 800 voip
>>>>>>>>>> �description OUTBOUND Voice SIP calls to VzB
>>>>>>>>>> �destination-pattern 1800[2-9]......
>>>>>>>>>> �voice-class sip dtmf-relay force rtp-nte
>>>>>>>>>> �session protocol sipv2
>>>>>>>>>> �session target sip-server
>>>>>>>>>> �incoming called-number .
>>>>>>>>>> �dtmf-relay rtp-nte
>>>>>>>>>> �codec g711ulaw
>>>>>>>>>> �no vad
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> !
>>>>>>>>>> sip-ua
>>>>>>>>>> �retry invite 2
>>>>>>>>>> �retry bye 2
>>>>>>>>>> �retry cancel 2
>>>>>>>>>> �registrar dns:verizonsipgateway expires 3600
>>>>>>>>>> �sip-server dns:verizonsipgateway:5071
>>>>>>>>>> �g729-annexb override
>>>>>>>>>> !
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Jonathan
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> cisco-voip mailing list
>>>>>>>>>> cisco-voip [at] puck
>>>>>>>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip [at] puck
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>

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