[asterisk-users] G.729 transcoder beetween asterisk to avaya

[satish patel]

Dear all

               I have Asterisk which is connected with avaya through E1 back 2 back now i have on asterisk side G.711 codec and Avaya also useing G.711 codec everything fine.

                I need G.729 on my asterisk side. can i have lots of SIP phone on my lan and issue is i have 2 to 3 building so problem is LAN is congested thats why i need G.729 Now testing perpose i have download Free G.729 codec from digium and install on asterisk and configure SIP device to use codec G.729 now when i make call from asterisk to avaya i got this error message


I have also done entry in sip.config

allow=g729

PBX>

    -- Executing [1648 at from-avaya:1] Dial("SIP/5450-a079d200", "Zap/g1/1648|60") in new stack
[Oct 30 21:54:26] WARNING[1318]: channel.c:3216 ast_request: No translator path exists for channel type Zap (native 76) to 256
[Oct 30 21:54:26] WARNING[1318]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1648 at from-avaya:2] Playback("SIP/5450-a079d200", "vm-nobodyavail") in new stack
[Oct 30 21:54:26] WARNING[1318]: channel.c:2991 set_format: Unable to find a codec translation path from g729 to gsm
[Oct 30 21:54:26] WARNING[1318]: file.c:813 ast_streamfile: Unable to open vm-nobodyavail (format 0x100 (g729)): No such file or directory
[Oct 30 21:54:26] WARNING[1318]: app_playback.c:437 playback_exec: ast_streamfile failed on SIP/5450-a079d200 for vm-nobodyavail
  == Auto fallthrough, channel 'SIP/5450-a079d200' status is 'CHANUNAVAIL'
[Oct 30 21:54:26] WARNING[1318]: channel.c:2991 set_format: Unable to find a codec translation path from g729 to slin
[Oct 30 21:54:26] WARNING[1318]: indications.c:121 playtones_alloc: Unable to set 'SIP/5450-a079d200' to signed linear format (write)
    -- Hungup 'Zap/32-1'





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Satish Patel
mobile:- +91-9818875535

http://www.linuxbug.org
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