
jared at puck
Mar 15, 2004, 8:27 AM
Post #3 of 9
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SIP should be able to handle the call transfers back and forth. Depending on what phones you're using, you might want to try out the chan_skinny module in asterisk. I'm using it with mixed success on a test phone (7970). I suspect that the 7940/60 firmware works better when running SCCP. - Jared On Mon, Mar 15, 2004 at 10:28:24AM -0500, Kyle Stone wrote: > Not for a gateway.. Asterisk is essentially a self contained PBX, I'm > trying to route calls into and out of it. > > > Kyle > > > -----Original Message----- > From: Jared Mauch [mailto:jared [at] puck] > Sent: Monday, March 15, 2004 10:23 AM > To: Kyle Stone > Cc: cisco-voip [at] puck > Subject: Re: [cisco-voip] Asterisk and a CCM... > > On Mon, Mar 15, 2004 at 10:19:20AM -0500, Kyle Stone wrote: > > I'm trying to bridge Asterisk PBX and CCM. I setup a h323 gateway on > > CCM and pointed it to the Asterisk box. I build chan_oh323 for > > asterisk.. installed it, and pointed it to route to the CCM. > > I call a CCM extension from Asterisk and the phone rings 1.5 times and > > CCM ends the call. I can tell this from the Asterisk debug logs. > > > > > > Anyone have any suggestions as how to debug this on the CCM side? > > Not sure if your CM supports it, but perhaps you want to > try SIP instead of H.323. Much simpler, IMHO. > > - Jared > > -- > Jared Mauch | pgp key available via finger from jared [at] puck > clue++; | http://puck.nether.net/~jared/ My statements are only > mine. -- Jared Mauch | pgp key available via finger from jared [at] puck clue++; | http://puck.nether.net/~jared/ My statements are only mine.
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