I tried to set allow=all in sip.conf. But it still doesn't work.
2011/2/16 Faisal Hanif <[email protected]> > I faced same issue for sipgate but got it resolved by allowing all codec in > sipgate peer config. > > > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Felix Dong > *Sent:* Wednesday, February 16, 2011 5:33 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] function Echo() doesn't work > > > > * == Using SIP RTP CoS mark 5* > > * -- Executing [1174614@von-voip-provider:1] > Answer("SIP/sipgate-account-00000000", "") in new stack* > > * -- Executing [1174614@von-voip-provider:2] > Echo("SIP/sipgate-account-00000000", "") in new stack* > > * == Spawn extension (von-voip-provider, 1174614, 2) exited non-zero on > 'SIP/sipgate-account-00000000'* > > > > > > here is the log. It is as same as I got from CAPI and Datacard. I just > didn't hear the echo from SIP connection. > > > 2011/2/16 Faisal Hanif <[email protected]> > > Check if you have incoming SIP call in supported codec or send CLI log for > further troubleshooting. > > > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Felix Dong > *Sent:* Wednesday, February 16, 2011 5:14 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] function Echo() doesn't work > > > > Yes, I did it. The function Echo() works both on CAPI and Datacard (UMTS > Stick). Just only no echo on SIP. Any suggestion? > > 2011/2/16 Faisal Hanif <[email protected]> > > Did you executed Answer() before it? > > > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *Felix Dong > *Sent:* Wednesday, February 16, 2011 4:48 PM > *To:* [email protected] > *Subject:* [asterisk-users] function Echo() doesn't work > > > > Hi guys, > > > > the function Echo() did work on CAPI, but doesn't work for SIP connection. > Can anybody help? > > thanks a lot. > > > > best regards, > > > > Felix > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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