this may help you http://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) <[EMAIL PROTECTED]> 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 [EMAIL PROTECTED] > -- SIP/mutualphone-6b26 is ringing > -- SIP/mutualphone-6b26 answered IAX2/[EMAIL PROTECTED]/2 > > The BT101 gives this: > > -- Called [EMAIL PROTECTED] > -- 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 > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
