[Asterisk-Users] G729 troubles

[Sean Cheesman]

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:akirichenko at bsh.ru]
Sent: Wednesday, December 24, 2003 7:04 PM
To: asterisk-users at lists.digium.com
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: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 
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 
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