Hi All, I have a weird problem. I have asterisk setup using the G729 Codec to receive Incoming calls both from a SIP Gateway (SER and Quintum) and via ISDN using i4l and have rules setup in extensions.conf for sending calls out either back via the SIP Gateway or ISDN. What I want to do is have PSTN calls come in via the SIP Gateway, be answered by the auto-attendant and then sent back out to the SIP Gateway to a PSTN number when the particular choice is made. However it gets to the point of ringing and then once the call is connected, there is no voice traffic and the following message appears:
chan_sip.c:2752 process_sdp: No compatible codecs! (Only if multiple codecs are available on the SER server) otherwise if I only have one codec allowed on the Quintum and Asterisk, it does not come up with this error (eg G729 or ALAW). However if you ring in from the PSTN (via ISDN) and select this option, it completes the call as requested, the same if I call the menu and select the option from a ip phone connected directly to the Asterisk Box. These Configurations work fine (In Easy step through): Incoming Call from ISDN --> Asterisk Menu --> Selection Made --> Call sent out to SER/Quintum --> Connected Party Incoming Call from local IP Phone --> Asterisk Menu --> Selection Made --> Call sent out to SER/Quintum --> Connected Party Incoming Call from SER from IP Phone --> Asterisk Menu --> Selection Made --> Call sent out to SER/Quintum --> Connected Party This doesn't work: Incoming Call from SER/Quintum from PSTN --> Asterisk Menu --> Selection Made --> Call sent out to SER/Quintum --> Connected Party Everything looks ok here and the configuration is correct (when I can make calls out to the SIP Gateway, both from mentioned earlier and from the IP Phone.) It appears that its only effecting incoming calls coming in from the Quintum from the PSTN to the SER gateway and then to asterisk, which are then being sent back out the SER gateway to the quintum to carry the call back to the PSTN. Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.20:5060;branch=z9hG4bK39591ff8;rport=5060 To: <sip:[EMAIL PROTECTED]>;tag=1bc039ff From: "0388016766" <sip:[EMAIL PROTECTED]>;tag=as34cebb79 Call-ID: [EMAIL PROTECTED] CSeq: 104 INVITE Record-Route: <sip:[EMAIL PROTECTED];ftag=as34cebb79;lr> Contact: <sip:[EMAIL PROTECTED]:5061> Content-Type: application/sdp Content-Length: 207 v=0 o=Quintum 13544 2493 IN IP4 192.168.1.90 s=VoipCall c=IN IP4 192.168.1.90 t=0 0 m=audio 10672 RTP/AVP 18 101 c=IN IP4 192.168.1.90 a=rtpmap:18 g729/8000/1 a=rtpmap:101 telephone-event/8000/1 10 headers, 9 lines Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.90:10672 Found description format g729 Found description format telephone-event Capabilities: us - 0x108(ALAW|G729A), peer - audio=0x100(G729A)/video=0x0(EMPTY), combined - 0x100(G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) set_destination: Parsing <sip:[EMAIL PROTECTED];lr> for address/port to send to set_destination: set destination to 192.168.1.90, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED]:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.20:5060;branch=z9hG4bK3fd2c533;rport Route: <sip:[EMAIL PROTECTED]:5061> From: "0388016766" <sip:[EMAIL PROTECTED]>;tag=as34cebb79 To: <sip:[EMAIL PROTECTED]>;tag=1bc039ff Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 104 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to 192.168.1.90:5060 The strange thing is that when this happens it appears that the RTP stream is stable and there is no indication of problems in selecting the codecs. Is there any possible cause as to why this may happen? Especially when it works correctly when I make a call in via the ISDN or an IP phone connected to the Asterisk Server. Does anyone have any pointers as to what may be causing this problem? Thanks, David _______________________________________________ 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