Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: -- Accepting AUTHENTICATED call from 192.168.112.99, requested format = 512, actual format = 512 -- Called 0031651931985@mutualphone -- SIP/mutualphone-6b26 is ringing -- SIP/mutualphone-6b26 answered IAX2/iaxrene@iaxrene/2 The BT101 gives this: -- Called 003165193XXXX@mutualphone -- SIP/mutualphone-2de1 is ringing -- SIP/mutualphone-2de1 answered SIP/chimit01-6013 -- Attempting native bridge of SIP/chimit01-6013 and SIP/mutualphone-2de1 Jan 16 18:50:41 WARNING[18631600]: chan_sip.c:2804 process_sdp: No compatible codecs! -- Got SIP response 488 "Not Acceptable Here" back from 209.250.147.116 show translation (I figure this has anything to do with it) shows that all paths are supported: G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - 4 2 2 3 2 1 4 13 35 19 GSM 15 - 2 2 3 2 1 4 13 35 19 ULAW 15 4 - 1 3 2 1 4 13 35 19 ALAW 15 4 1 - 3 2 1 4 13 35 19 G726 17 6 4 4 - 4 3 6 15 37 21 ADPCM 15 4 2 2 3 - 1 4 13 35 19 SLINR 14 3 1 1 2 1 - 3 12 34 18 LPC10 17 6 4 4 5 4 3 - 15 37 21 G729A 17 6 4 4 5 4 3 6 - 37 21 SPEEX 16 5 3 3 4 3 2 5 14 - 20 ILBC 17 6 4 4 5 4 3 6 15 37 - The first preferred Vocoder configured in the BT101 is PCMU, but changing this to G729 (the one that mutualphone is using) won't make it work. I changed the option back again because all other services (FWD, BRI, IAX2) work like this and I don't want to break them. Any suggestions about what I can change to make this work? Cheers! Rene Kluwen Chimit -----William Suffil's Comment----- I've heard problems with the Grandstream G729 and the new digium G729 by MAC ID. Could be a compatibility issue with the implementations. Did you ever use the Grandstream against asterisk with the old Voiceage G729? I've heard that works just fine. -- William This is not true. I use Grandstream with Digium Codec G729 just fine. The Old Voiceage codec infact has the problem where the calls do not connect and when they connect, the quality is horrendous. My guess is that the entries in SIP.CONF have not been setup properly to use the available codecs. Best is to post the SIP.CONF entries here to see what is missing. By the where did you get the G723 and G729 from? If you have compiled them on your own, did you statically link the libraries? Or just copied the .SO files from another dude's Asterisk box? Post all the details Seshu Kanuri -------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited.
this may help you billing.mutualphone.com/phpBB2/viewtopic.php?t=78&postdays=0&postorder=asc&start=15 On Tue, 18 Jan 2005 10:23:45 -0500, Kanuri, Seshu (Company IT) <Seshu.Kanuri@morganstanley.com> wrote:> > Original Post > ---------------- > I have an Asterisk related problem with mutualphone. > I can connect to any number with any softphone that I am using (iaxcomm, > SJPhone, and a few others). > Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to > mutualphone destinations. Other destinations go fine. > > A working phone call (e.g. from iaxcomm) gives the following on the > console: > > -- Accepting AUTHENTICATED call from 192.168.112.99, requested > format = 512, actual format = 512 > -- Called 0031651931985@mutualphone > -- SIP/mutualphone-6b26 is ringing > -- SIP/mutualphone-6b26 answered IAX2/iaxrene@iaxrene/2 > > The BT101 gives this: > > -- Called 003165193XXXX@mutualphone > -- SIP/mutualphone-2de1 is ringing > -- SIP/mutualphone-2de1 answered SIP/chimit01-6013 > -- Attempting native bridge of SIP/chimit01-6013 and > SIP/mutualphone-2de1 > Jan 16 18:50:41 WARNING[18631600]: chan_sip.c:2804 process_sdp: No > compatible codecs! > -- Got SIP response 488 "Not Acceptable Here" back from > 209.250.147.116 > > show translation (I figure this has anything to do with it) shows > that all paths are supported: > > G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX > ILBC > G723 - 4 2 2 3 2 1 4 13 35 > 19 > GSM 15 - 2 2 3 2 1 4 13 35 > 19 > ULAW 15 4 - 1 3 2 1 4 13 35 > 19 > ALAW 15 4 1 - 3 2 1 4 13 35 > 19 > G726 17 6 4 4 - 4 3 6 15 37 > 21 > ADPCM 15 4 2 2 3 - 1 4 13 35 > 19 > SLINR 14 3 1 1 2 1 - 3 12 34 > 18 > LPC10 17 6 4 4 5 4 3 - 15 37 > 21 > G729A 17 6 4 4 5 4 3 6 - 37 > 21 > SPEEX 16 5 3 3 4 3 2 5 14 - > 20 > ILBC 17 6 4 4 5 4 3 6 15 37 > - > > The first preferred Vocoder configured in the BT101 is PCMU, but > changing this to G729 (the one that mutualphone is using) won't make it > work. I changed the option back again because all other services (FWD, > BRI, IAX2) work like this and I don't want to break them. > > Any suggestions about what I can change to make this work? > > Cheers! > > Rene Kluwen > Chimit > -----William Suffil's Comment----- > I've heard problems with the Grandstream G729 and the new digium G729 by > MAC ID. Could be a compatibility issue with the implementations. > Did you ever use the Grandstream against asterisk with the old Voiceage > G729? I've heard that works just fine. > -- William > > This is not true. I use Grandstream with Digium Codec G729 just fine. > The Old Voiceage codec infact has the problem where the calls do not > connect and when they connect, the quality is horrendous. > > My guess is that the entries in SIP.CONF have not been setup properly to > use the available codecs. > > Best is to post the SIP.CONF entries here to see what is missing. > > By the where did you get the G723 and G729 from? If you have compiled > them on your own, did you statically link the libraries? Or just copied > the .SO files from another dude's Asterisk box? > > Post all the details > > Seshu Kanuri > -------------------------------------------------------- > > NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users >