[Asterisk-Users] G729 troubles

Anton V Kirichenko akirichenko at bsh.ru
Wed Dec 24 17:49:01 MST 2003 

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:scheesman at gdsworks.com] 
> Sent: Thursday, December 25, 2003 3:34 AM
> To: 'asterisk-users at lists.digium.com'
> 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: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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