Channel.c:1045 ast_best_codec

Hi everyone!

This is my first post, I hope you can help me with this problem that has presented me in my Asterisk 1.8.2.3…

Problem:

I have several Asterisk servers connected to each other, for lack of time are different versions.

When I try to make a call from the Asterisk server with Asterisk 1.4.25.1 1.8.2.3 to the server through an IAX trunk I returned the following in the CLI 1.8.2.3

-- Executing [4404@sip-normales:1] Dial("SIP/4171-00018aae", "iax2/avila-ivr/4404,300,Tt") in new stack [Jun 14 17:07:04] WARNING[21608]: channel.c:1045 ast_best_codec: Don't know any of 0x0 (nothing) formats -- Called avila-ivr/4404 -- Call accepted by XX.XX.XX.XX (format alaw) -- Format for call is alaw -- IAX2/avila-ivr-18974 is ringing -- IAX2/avila-ivr-18974 stopped sounds -- IAX2/avila-ivr-18974 answered SIP/4171-00018aae

And on the server with 1.4.25.1 shows me the following

Accepting AUTHENTICATED call from XX.XX.XX.XX: > requested format = h263, > requested prefs = (alaw|gsm), > actual format = alaw, > host prefs = (alaw|ulaw|gsm), > priority = mine -- Executing [4404@incoming-iax:1] Goto("IAX2/user-astcnti-4303", "internal|4404|1") in new stack -- Goto (internal,4404,1) -- Executing [4404@internal:1] Macro("IAX2/user-astcnti-4303", "stdexten-extra-followme|4404|SIP/4404|4404") in new stack -- Executing [s@macro-stdexten-extra-followme:1] Set("IAX2/user-astcnti-4303", "tempcfim=") in new stack -- Executing [s@macro-stdexten-extra-followme:2] Set("IAX2/user-astcnti-4303", "tempdnd=") in new stack -- Executing [s@macro-stdexten-extra-followme:3] Set("IAX2/user-astcnti-4303", "tempcfb=") in new stack -- Executing [s@macro-stdexten-extra-followme:4] Set("IAX2/user-astcnti-4303", "tempcfna=") in new stack -- Executing [s@macro-stdexten-extra-followme:5] Set("IAX2/user-astcnti-4303", "tempcfollow1=") in new stack -- Executing [s@macro-stdexten-extra-followme:6] Set("IAX2/user-astcnti-4303", "tempcfollow2=") in new stack -- Executing [s@macro-stdexten-extra-followme:7] Set("IAX2/user-astcnti-4303", "tempcfollow3=") in new stack -- Executing [s@macro-stdexten-extra-followme:8] Set("IAX2/user-astcnti-4303", "CDR(accountcode)=call-to-ext-4404") in new stack -- Executing [s@macro-stdexten-extra-followme:9] ResetCDR("IAX2/user-astcnti-4303", "") in new stack -- Executing [s@macro-stdexten-extra-followme:10] GotoIf("IAX2/user-astcnti-4303", "?s|dndyes:dndno") in new stack -- Goto (macro-stdexten-extra-followme,s,13) -- Executing [s@macro-stdexten-extra-followme:13] NoOp("IAX2/user-astcnti-4303", "we continue with cfim test") in new stack -- Executing [s@macro-stdexten-extra-followme:14] GotoIf("IAX2/user-astcnti-4303", "?s|cfimyes:cfimno") in new stack -- Goto (macro-stdexten-extra-followme,s,25) -- Executing [s@macro-stdexten-extra-followme:25] NoOp("IAX2/user-astcnti-4303", "no cfim so lets continue") in new stack -- Executing [s@macro-stdexten-extra-followme:26] Dial("IAX2/user-astcnti-4303", "SIP/4404|20|wt") in new stack -- Called 4404 -- SIP/4404-09ea1248 is ringing -- SIP/4406-b6503c08 answered IAX2/user-astcnti-4303 == Spawn extension (macro-stdexten-extra-followme, s, 26) exited non-zero on 'IAX2/user-astcnti-4303' in macro 'stdexten-extra-followme' == Spawn extension (internal, 4404, 1) exited non-zero on 'IAX2/user-astcnti-4303' -- Hungup 'IAX2/user-astcnti-4303'

Calls are made ​​from a VX-1500 phone Polycom to Polycom SoundPoint IP 331. When the call is answered you hear a lot noise in the channel and can not hear the person on the other side.

iax.conf Server with Asterisk 1.8.2.3

[avila-ivr]
type=peer
username=**********
host=XX.XX.XX.XX
secret=************
context=incoming-iax
qualify=yes
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=gsm
allow=h263
notransfer=yes
trunk=yes

iax.conf Server with Asterisk 1.4.25.1

[asterisk-cnti]
;
type=peer
username=user-ivr
host=XX.XX.XX.XX
secret=*********
context=ivr-atencion-usuarios
qualify=yes
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=h261
allow=h263


[user-astcnti]
;
type=user
secret=*********
context=incoming-iax
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=h261
allow=h263
allow=h263p
notransfer=yes

I’ve tried everything, including adding the necessary codecs but nothing worked, I hope you can somehow lead me to resolve this problem.

Thanks in advance.

hi:
what is the config for SIP/4171? it looks codec mismatch?