[Asterisk-Users] G729 troubles

[Anton V Kirichenko]

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:peterabrown at froggy.com.au] 
> Sent: Thursday, December 25, 2003 2:50 AM
> To: asterisk-users at lists.digium.com
> 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:1}@IP.IP.IP.IP
> >
> >Some logs from Asterisk:
> >
> >First mgcp CRCX after hang up:
> >Posting Request:
> >CRCX 323 aaln/1 at DLINK 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/1 at DLINK-0", 
> >"SIP/3632034 at IP.IP.IP.IP") in new stack
> >
> >INVITE sip:3632034 at IP.IP.IP.IP 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/1 at DLINK-1
> >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:3632034 at IP.IP.IP.IP:5060 SIP/2.0
> >
> >Sip read:
> >SIP/2.0 487 Request Cancelled
> >....
> >
> >--
> >Antonio
> >_______________________________________________
> >Asterisk-Users mailing list
> >Asterisk-Users at lists.digium.com
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> Peter Brown
> CEO
> IP Telephonics 
> 
> 
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