[Asterisk-Users] Grandstream / SIP - IAX2 / Voicepulse
Dear Folks, Trying to make calls from a GS behind NAT using SIP through my * server talking IAX2 to Voicepulse and no success. From GS to Zap/PSTN is ok and vice-versa. From ZAP to Voicepulse(IAX2) no problem... but.. not getting to connect SIP-IAX2 and the problem is not only with VoicePulse but with another provider as well in the same situation, GS(SIP)- * - IAX2 - ITSP -- Call accepted by 66.234.228.132 (format G729A) -- Format for call is G729A -- IAX2[voicepulse]/2 is busy -- Hungup 'IAX2[voicepulse]/2' == Everyone is busy at this time -- Executing Congestion(SIP/1604-4f72, ) in new stack What should it be? Thanks in advance, Isamar Maia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Subject: [Asterisk-Users] Grandstream / SIP - IAX2 / Voicepulse
but.. not getting to connect SIP-IAX2 and the problem is not only with VoicePulse but with another provider as well in the same situation, GS(SIP)- * - IAX2 - ITSP -- Call accepted by 66.234.228.132 (format G729A) -- Format for call is G729A -- IAX2[voicepulse]/2 is busy -- Hungup 'IAX2[voicepulse]/2' == Everyone is busy at this time -- Executing Congestion(SIP/1604-4f72, ) in new stack This is codec incompatibility. Either you/your provider cannot use G729 to connect. Check your iax.conf and check the [general] context as this is the one for outgoing codec negotiation. There is bug in the codec negotiation process, you can refer to previous posts. Do you have licence to use g729, otherwise unless voicepluse accept gsm calls. -- David Kwok FWD#/IAXTEL# : 17001813482 ext 1002 smime.p7s Description: S/MIME Cryptographic Signature
