In my case I see only g729 codec request from CPE (see mgcp CRCX) and only g729 from PGW2200 (see debug of sip messages) and I don't need and transcoding from one codec format to another codec format. Could you expain to me why asterisk starts transcoding process from g729 to alaw ?
-- antonio > -----Original Message----- > From: Sean Cheesman [mailto:[EMAIL PROTECTED] > Sent: Thursday, December 25, 2003 3:34 AM > To: '[EMAIL PROTECTED]' > Subject: RE: [Asterisk-Users] G729 troubles > > I'm going to take a stab at this, so someone correct me if > I'm wrong! If you're calling one g729 device from another, > the call is actually handed off without any decoding done, > therefore the licensing isn't needed. If * has to connect > the g729 call to another format, then the licensing comes in > to play. And it could be that even though you've configured > the disabling of the codec at one location, it still is > enabled elsewhere? Close? Anyone? > > Sean > > -----Original Message----- > From: Anton V Kirichenko [mailto:[EMAIL PROTECTED] > Sent: Wednesday, December 24, 2003 7:04 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] G729 troubles > > > No, I did't bought any license from Digium. But as I say at > my previous post, only _some part_ of my g729 calls are failed ! > I think if I need license for G729 at Asterisk then all of my > calls must to fails. Is it right ? > > -- > Antonio > > > -----Original Message----- > > From: Peter Brown [mailto:[EMAIL PROTECTED] > > Sent: Thursday, December 25, 2003 2:50 AM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] G729 troubles > > > > Have you bought G.729a from Digium which cost $10/channel? > > At 02:04 25/12/03 +0300, you wrote: > > >Hello, > > >I've successfully installed Asterisk from last CVS and > > configured it > > >for using with DLINK-DG104S as mgcp CPE and PGW2200 as > external sip > > >server. > > >All are work fine at G711 codecs, but then I disable all > > codecs except > > >g729 some calls failed (Not all calls. Some calls passed at g729 > > >succesfully). > > > All my devices configred to use only g729 and I don't see > > other codecs > > >at mgcp or sip messages, but I see strange string at > asterisks log: > > > > > >NOTICE[196633]: File rtp.c, Line 418 (ast_rtp_read): Unknown > > RTP codec > > >123 received > > >NOTICE[196633]: File channel.c, Line 1478 > > (ast_set_read_format): Unable > > >to find a path from ALAW to G729A > > >NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): > > >Unable to find a path from G729A to ALAW > > >WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to > > >transmit frame type 8, while native formats is 256 (read/write = > > >256/256) > > >WARNING[196633]: File app_dial.c, Line 279 > > (wait_for_answer): Unable to > > >forward frame > > > > > >I find similary posts at Asteris-Users mailing list, but > > don't find how > > >to resolve this trouble. Is this a bug or some > > misconfiguration at my > > >configs ? > > > > > >sip.conf: > > >[general] > > >port = 5060 > > >bindaddr = 0.0.0.0 > > >context = local > > >disallow = all > > >allow = g729 > > >mgcp.conf > > >[general] > > >port = 2427 > > >bindaddr = 0.0.0.0 > > >disallow = all > > >allow = g729 > > >[DLINK] > > >context=local > > >host=Y.Y.Y.Y > > >threewaycalling=yes > > >transfer=yes > > >line => aaln/1 > > >line => aaln/2 > > >line => aaln/3 > > >line => aaln/4 > > >extension.conf > > >[local] > > >ignorepat => 9 > > >exten => _9XXXXXXX,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED] > > > > > >Some logs from Asterisk: > > > > > >First mgcp CRCX after hang up: > > >Posting Request: > > >CRCX 323 aaln/[EMAIL PROTECTED] MGCP 1.0 > > >v=0 > > >o=root 23577 23577 IN IP4 X.X.X.X > > >s=session > > >c=IN IP4 X.X.X.X > > >t=0 0 > > >m=audio 14548 RTP/AVP 18 > > >a=rtpmap:18 G729/8000 > > > > > >After that I enter phone number and sent call to sip server: > > > > > > -- Executing Dial("MGCP/aaln/[EMAIL PROTECTED]", > > >"SIP/[EMAIL PROTECTED]") in new stack > > > > > >INVITE sip:[EMAIL PROTECTED] SIP/2.0 <skip> v=0 o=root > 16078 16078 > > >IN IP4 X.X.X.X s=session c=IN IP4 X.X.X.X t=0 0 m=audio > 18480 RTP/AVP > > >18 101 > > >a=rtpmap:18 G729/8000 > > >a=rtpmap:101 telephone-event/8000 > > >a=fmtp:101 0-16 > > > > > >Then I receive reply from SIP server: > > >Sip read: > > >SIP/2.0 100 Trying > > ><skip> > > > > > >Sip read: > > >SIP/2.0 183 Session Progress > > ><skip> > > >v=0 > > >o=- 0 0 IN IP4 Z.Z.Z.Z > > >s=- > > >c=IN IP4 Z.Z.Z.Z > > >t=0 0 > > >m=audio 49640 RTP/AVP 18 101 > > >a=rtpmap:101 telephone-event/8000 > > >a=fmtp:101 0-15 > > >a=X-sqn: 0 > > >a=X-cap: 1 image udptl t38 > > >a=sqn: 0 > > >a=cdsc: 1 image udptl t38 > > > > > >After this message sometimes Asterisk make error message > at log and > > >drop > > >call: > > > > > > -- SIP/IP.IP.IP.IP-b782 is making progress passing it to > > >MGCP/aaln/[EMAIL PROTECTED] > > >srv-5*CLI> NOTICE[196633]: File rtp.c, Line 418 > > (ast_rtp_read): Unknown > > >RTP codec 123 received > > >NOTICE[196633]: File channel.c, Line 1478 > > (ast_set_read_format): Unable > > >to find a path from ALAW to G729A > > >NOTICE[196633]: File channel.c, Line 1448 (ast_set_write_format): > > >Unable to find a path from G729A to ALAW > > >WARNING[196633]: File chan_mgcp.c, Line 757 (mgcp_write): Asked to > > >transmit frame type 8, while native formats is 256 (read/write = > > >256/256) > > >WARNING[196633]: File app_dial.c, Line 279 > > (wait_for_answer): Unable to > > >forward frame > > > > > >Reliably Transmitting: > > >CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0 > > > > > >Sip read: > > >SIP/2.0 487 Request Cancelled > > >.... > > > > > >-- > > >Antonio > > >_______________________________________________ > > >Asterisk-Users mailing list > > >[EMAIL PROTECTED] > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > Peter Brown > > CEO > > IP Telephonics > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
