Hi
I’ve got an issue with codecs in asterisk 1.8.4.4. The remote peer only supports alaw, which asterisk identifies, but for some reason read/write format is set to g726 anyway. This results in audio working from peer to asterisk but not the otherway.
Any help is much appreciated!
sip.conf:
disallow=all
allow=g726
allow=alaw
allow=ulaw
debug log (call from asterisk to peer):
Capabilities: us - 0x80c (ulaw|alaw|g726), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 8 09:07:06] DEBUG[2148]: res_rtp_asterisk.c:2391 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x5f9090’
Peer audio RTP is at port 192.168.0.10:36832
[Nov 8 09:07:06] DEBUG[2148]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0x7e1fb910 to 0x5f9240
[Nov 8 09:07:06] DEBUG[2148]: rtp_engine.c:516 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7e1fb910 to 0x5f9240
[Nov 8 09:07:06] DEBUG[2148]: chan_sip.c:8863 process_sdp: We’re settling with these formats: 0x8 (alaw)
[Nov 8 09:07:06] DEBUG[2148]: chan_sip.c:8868 process_sdp: We have an owner, now see if we need to change this call
[Nov 8 09:07:06] DEBUG[29496]: channel.c:4693 ast_write: Deadlock avoided for write to channel ‘SIP/mypeer-00000053’
[Nov 8 09:07:06] DEBUG[2148]: chan_sip.c:8875 process_sdp: Oooh, we need to change our audio formats since our peer supports only 0x8 (alaw) and not 0x800 (g726)
[Nov 8 09:07:06] DEBUG[2148]: channel.c:5048 set_format: Set channel SIP/mypeer-00000053 to read format g726
[Nov 8 09:07:06] DEBUG[2148]: channel.c:5048 set_format: Set channel SIP/mypeer-00000053 to write format g726