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
