Hello, Could someone give me clues where to figure out this problem? If I call from a Sip client to an Firefly client running IAX, the call connects fine, no problems. I can connect to asterisk using any codec (excepting g.729) on firefly to voicemail and music-on-hold, other sip extensions and everything works fine. If I try to connect to the same client via a ZAP channel (X100P clone), via Dial(IAX2/XXXX) I get an error : Jan 10 18:07:26 WARNING[1378]: chan_iax2.c:6017 socket_read: Call rejected by xx.xxx.xxx.xxx: No compatible Codecs I just updated asterisk to cvs-head-1-10-2005. All codecs are allowed in IAX.conf and all codecs are enabled on Firefly. I have tried everything I can think of- only enable gsm, only gsm+G.711, all codecs on firefly. Same results. Anyone else with this issue? Thanks, Ernie
use ethereal or iax2 debug to see what capabilities are been set in your NEW message Ernie Ankele wrote:> Hello, > Could someone give me clues where to figure out this problem? > If I call from a Sip client to an Firefly client running IAX, the call > connects fine, no problems. > I can connect to asterisk using any codec (excepting g.729) on firefly > to voicemail and music-on-hold, other sip extensions and everything > works fine. > If I try to connect to the same client via a ZAP channel (X100P clone), > via Dial(IAX2/XXXX) I get an error : > > Jan 10 18:07:26 WARNING[1378]: chan_iax2.c:6017 socket_read: Call > rejected by xx.xxx.xxx.xxx: No compatible Codecs > > I just updated asterisk to cvs-head-1-10-2005. All codecs are allowed in > IAX.conf and all codecs are enabled on Firefly. > I have tried everything I can think of- only enable gsm, only gsm+G.711, > all codecs on firefly. Same results. > Anyone else with this issue? > Thanks, Ernie > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
On a sip to iax :
CODEC_PREFS : (gsm|ulaw|alaw|ilbc)
and
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACCEPT
Timestamp: 00000ms SCall: 19170 DCall: 00001 [xx.xxx.xxx.xxx:20406]
FORMAT : 4
-- Call accepted by xx.xxx.xxx.xxx (format ulaw)
-- Format for call is ulaw
On ZAP to IAX:
CODEC_PREFS : (gsm|ulaw|alaw|ilbc)
and
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REJECT
Timestamp: 00000ms SCall: 15725 DCall: 00003 [xx.xxx.xxx.xxx:20406]
CAUSE : No compatible Codecs
Jan 10 18:50:06 WARNING[1378]: chan_iax2.c:6017 socket_read: Call
rejected by xx.xxx.xxx.xxx: No compatible Codecs
Thanks, Ernie
On Jan 10, 2005, at 6:34 PM, Adam Hart wrote:
> use ethereal or iax2 debug to see what capabilities are been set in
> your NEW message
>
> Ernie Ankele wrote:
>> Hello,
>> Could someone give me clues where to figure out this problem?
>> If I call from a Sip client to an Firefly client running IAX, the
>> call connects fine, no problems.
>> I can connect to asterisk using any codec (excepting g.729) on
>> firefly to voicemail and music-on-hold, other sip extensions and
>> everything works fine.
>> If I try to connect to the same client via a ZAP channel (X100P
>> clone), via Dial(IAX2/XXXX) I get an error :
>> Jan 10 18:07:26 WARNING[1378]: chan_iax2.c:6017 socket_read: Call
>> rejected by xx.xxx.xxx.xxx: No compatible Codecs
>> I just updated asterisk to cvs-head-1-10-2005. All codecs are allowed
>> in IAX.conf and all codecs are enabled on Firefly.
>> I have tried everything I can think of- only enable gsm, only
>> gsm+G.711, all codecs on firefly. Same results.
>> Anyone else with this issue?
>> Thanks, Ernie
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users@lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
Can you paste the full NEW frame please. Could be Preference vs capability thanks, Adam Ernie Ankele wrote:> On a sip to iax : > CODEC_PREFS : (gsm|ulaw|alaw|ilbc) > > and > Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: > ACCEPT > Timestamp: 00000ms SCall: 19170 DCall: 00001 [xx.xxx.xxx.xxx:20406] > FORMAT : 4 > > -- Call accepted by xx.xxx.xxx.xxx (format ulaw) > -- Format for call is ulaw > > On ZAP to IAX: > CODEC_PREFS : (gsm|ulaw|alaw|ilbc) > and > Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: > REJECT > Timestamp: 00000ms SCall: 15725 DCall: 00003 [xx.xxx.xxx.xxx:20406] > CAUSE : No compatible Codecs > > Jan 10 18:50:06 WARNING[1378]: chan_iax2.c:6017 socket_read: Call > rejected by xx.xxx.xxx.xxx: No compatible Codecs > > Thanks, Ernie > > On Jan 10, 2005, at 6:34 PM, Adam Hart wrote: > >> use ethereal or iax2 debug to see what capabilities are been set in >> your NEW message >> >> Ernie Ankele wrote: >> >>> Hello, >>> Could someone give me clues where to figure out this problem? >>> If I call from a Sip client to an Firefly client running IAX, the >>> call connects fine, no problems. >>> I can connect to asterisk using any codec (excepting g.729) on >>> firefly to voicemail and music-on-hold, other sip extensions and >>> everything works fine. >>> If I try to connect to the same client via a ZAP channel (X100P >>> clone), via Dial(IAX2/XXXX) I get an error : >>> Jan 10 18:07:26 WARNING[1378]: chan_iax2.c:6017 socket_read: Call >>> rejected by xx.xxx.xxx.xxx: No compatible Codecs >>> I just updated asterisk to cvs-head-1-10-2005. All codecs are allowed >>> in IAX.conf and all codecs are enabled on Firefly. >>> I have tried everything I can think of- only enable gsm, only >>> gsm+G.711, all codecs on firefly. Same results. >>> Anyone else with this issue? >>> Thanks, Ernie >>> _______________________________________________ >>> Asterisk-Users mailing list >>> Asterisk-Users@lists.digium.com >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Adam, I think I got it worked out...
I changed disallow=723.1 to disallow=all and then accepted back in
ulaw,alaw,gsm and ilbc and
it started accepting the calls. I do not know why, but its working now.
FWIW, here is the full frame as it was before:
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 00009ms SCall: 00001 DCall: 00000 [xx.xxx.xxx.xxx:20406]
VERSION : 2
CALLED NUMBER : s
CODEC_PREFS : (gsm|ulaw|alaw|ilbc)
CALLING NUMBER : 3035520218
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
LANGUAGE : en
FORMAT : 64
CAPABILITY : 1048575
ADSICPE : 2
DATE TIME : 170564634
thanks, Ernie
On Jan 10, 2005, at 7:17 PM, Adam Hart wrote:
> Can you paste the full NEW frame please. Could be Preference vs
> capability
>
> thanks,
>
> Adam
>
>
> Ernie Ankele wrote:
>> On a sip to iax :
>> CODEC_PREFS : (gsm|ulaw|alaw|ilbc)
>> and
>> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX
>> Subclass: ACCEPT
>> Timestamp: 00000ms SCall: 19170 DCall: 00001
>> [xx.xxx.xxx.xxx:20406]
>> FORMAT : 4
>> -- Call accepted by xx.xxx.xxx.xxx (format ulaw)
>> -- Format for call is ulaw
>> On ZAP to IAX:
>> CODEC_PREFS : (gsm|ulaw|alaw|ilbc)
>> and
>> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX
>> Subclass: REJECT
>> Timestamp: 00000ms SCall: 15725 DCall: 00003
>> [xx.xxx.xxx.xxx:20406]
>> CAUSE : No compatible Codecs
>> Jan 10 18:50:06 WARNING[1378]: chan_iax2.c:6017 socket_read: Call
>> rejected by xx.xxx.xxx.xxx: No compatible Codecs
>> Thanks, Ernie
>> On Jan 10, 2005, at 6:34 PM, Adam Hart wrote:
>>> use ethereal or iax2 debug to see what capabilities are been set in
>>> your NEW message
>>>
>>> Ernie Ankele wrote:
>>>
>>>> Hello,
>>>> Could someone give me clues where to figure out this problem?
>>>> If I call from a Sip client to an Firefly client running IAX,
the
>>>> call connects fine, no problems.
>>>> I can connect to asterisk using any codec (excepting g.729) on
>>>> firefly to voicemail and music-on-hold, other sip extensions
and
>>>> everything works fine.
>>>> If I try to connect to the same client via a ZAP channel (X100P
>>>> clone), via Dial(IAX2/XXXX) I get an error :
>>>> Jan 10 18:07:26 WARNING[1378]: chan_iax2.c:6017 socket_read:
Call
>>>> rejected by xx.xxx.xxx.xxx: No compatible Codecs
>>>> I just updated asterisk to cvs-head-1-10-2005. All codecs are
>>>> allowed in IAX.conf and all codecs are enabled on Firefly.
>>>> I have tried everything I can think of- only enable gsm, only
>>>> gsm+G.711, all codecs on firefly. Same results.
>>>> Anyone else with this issue?
>>>> Thanks, Ernie
>>>> _______________________________________________
>>>> Asterisk-Users mailing list
>>>> Asterisk-Users@lists.digium.com
>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>> To UNSUBSCRIBE or update options visit:
>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>> _______________________________________________
>>> Asterisk-Users mailing list
>>> Asterisk-Users@lists.digium.com
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users@lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>