I'm having a problem with Asterisk choosing the wrong peer entry from the sip.conf file. Based off the debug below Asterisk should see the message coming from TNT3 (see the sip.conf below) and not from SER_FAX (which it shows in the debug below "Found peer 'SER_FAX'"). Based off what I have here it seems to me that since the From is 198.88.216.30 it should match with the TNT3 entry in my sip.conf which is what I want it to do, instead it is matching with the SIP proxy that is proxying it the SIP message. Is there a way to get Asterisk to lookup based of the originator of the INVITE instead of by who last proxied it the INVITE?
Basically here is my setup: 198.88.216.84 = SER 198.88.216.30 = TNT3 (My PSTN Gateway) 198.88.216.85 = Asterisk TNT3 <---> SER Proxy <---> Asterisk ----- Here is the debug from Asterisk: Sip read: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Record-Route: <sip:[EMAIL PROTECTED]:5060;ftag=46cae657-1bf8d0c3-1ed858c6;lr=on> To: <sip:[EMAIL PROTECTED]:5060;user=phone> From: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=46cae657-1bf8d0c3-1ed858c 6 Remote-Party-Id: <sip:[EMAIL PROTECTED]:5060;user=phone>;screen=yes;id-type=subscriber ;party=calling;privacy=off Proxy-Require: privacy Call-ID: [EMAIL PROTECTED] CSeq: 93145626 INVITE Via: SIP/2.0/UDP 198.88.216.84;branch=z9hG4bK4aca.dc7b3842.0 Via: SIP/2.0/UDP 198.88.216.30:5060;rport=5060 Max-Forwards: 69 Contact: <sip:[EMAIL PROTECTED]:5060;user=phone> Supported: replaces Supported: 100rel Content-Type: application/sdp Content-Length: 272 v=0 o=TNT3 469291203 469291203 IN IP4 198.88.216.30 s=Session SDP c=IN IP4 198.88.216.30 t=0 0 m=audio 44518 RTP/AVP 18 0 8 96 a=silenceSupp:off a=ecan:b on g168 a=rtpmap:96 telephone-event/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 16 headers, 12 lines Using latest request as basis request Sending to 198.88.216.84 : 5060 (non-NAT) Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 96 Peer audio RTP is at port 198.88.216.30:44518 Found description format telephone-event Found description format PCMA Found description format PCMU Found description format G729 Capabilities: us - 0x10c(ULAW|ALAW|G729A), peer - audio=0x10c(ULAW|ALAW|G729A)/video=0x0(EMPTY), combined - 0x10c(ULAW|ALAW|G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found peer 'SER_FAX' Looking for 4444444444 in FROM_PSTN list_route: hop: <sip:[EMAIL PROTECTED]:5060;ftag=46cae657-1bf8d0c3-1ed858c6;lr=on> list_route: hop: <sip:[EMAIL PROTECTED]:5060;user=phone> Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 198.88.216.84;branch=z9hG4bK4aca.dc7b3842.0;received=198.88.216.84;rport=506 0 Via: SIP/2.0/UDP 198.88.216.30:5060 From: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=46cae657-1bf8d0c3-1ed858c 6 To: <sip:[EMAIL PROTECTED]:5060;user=phone>;tag=as62f171b3 Call-ID: [EMAIL PROTECTED] CSeq: 93145626 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 ----- Here is my sip.conf: [general] port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind to context=FROM_PSTN ; Default for incoming calls rtptimeout=30 ; Terminate call if 60 seconds of no RTP activity canreinvite=no disallow=all allow=g729 allow=ulaw allow=alaw [SER] type=friend host=198.88.216.84 port=5060 qualify=no nat=yes disallow=all allow=g729 allow=ulaw allow=alaw [SER_FAX] type=friend host=198.88.216.84 port=5060 qualify=no nat=yes disallow=all allow=ulaw [TNT3] type=friend host=198.88.216.30 port=5060 dtmfmode=rfc2833 qualify=no canreinvite=no nat=no context=FROM_PSTN deny=0.0.0.0 permit=198.88.216.30/255.255.255.255 disallow=all allow=g729 allow=ulaw allow=alaw ---------------------------------------- Michael Shuler _______________________________________________ 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
