h7-ast3*CLI> sip debug SIP Debugging Enabled h7-ast3*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP [Asterisk IP address]:5060;branch=z9hG4bK6a23ccfd;rport From: "Jon Brüel" ;tag=as4810e82a To: ;tag=1c25247 Call-ID: 6da3e76f084570156f90d8597aa68af0@[Asterisk IP address] CSeq: 102 INVITE Contact: Supported: em,timer,100rel Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE Server: Audiocodes-Sip-Gateway/MP-104 FXS/v.4.30.19.7 Content-Type: application/sdp Content-Length: 210 v=0 o=AudiocodesGW 70467 75160 IN IP4 [Audiocode IP address] s=Phone-Call c=IN IP4 [Audiocode IP address] t=0 0 m=audio 4000 RTP/AVP 8 96 a=rtpmap:8 pcma/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 12 headers, 10 lines Found RTP audio format 8 Found RTP audio format 96 Peer audio RTP is at port [Audiocode IP address]:4000 Found description format pcma Found description format telephone-event Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to [Audiocode IP address], port 5060 Transmitting: ACK sip:100000001720@[Audiocode IP address];user=phone SIP/2.0 Via: SIP/2.0/UDP [Asterisk IP address]:5060;branch=z9hG4bK51722206;rport From: "Jon Brüel" ;tag=as4810e82a To: ;tag=1c25247 Contact: Call-ID: 6da3e76f084570156f90d8597aa68af0@[Asterisk IP address] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (NAT) to [Audiocode IP address]:5060 -- SIP/100000001720-d26f answered SIP/100000001707-afc9 We're at [Asterisk IP address] port 10478 Answering with preferred capability 0x8 (alaw) Answering with preferred capability 0x100 (g729) Answering with preferred capability 0x2 (gsm) Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 83.91.245.154:10954;branch=z9hG4bK-z0enc5ksi0z8;received=83.91.245.154;rport=10954 From: "Jon Brüel" ;tag=0nlcat16wf To: ;tag=as0ca655ac Call-ID: 3c26d7385e58-9xc2zuoc7dgt@192-168-100-107 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 295 v=0 o=root 21968 21968 IN IP4 [Asterisk IP address] s=session c=IN IP4 [Asterisk IP address] t=0 0 m=audio 10478 RTP/AVP 8 18 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 83.91.245.154:10954 -- Attempting native bridge of SIP/100000001707-afc9 and SIP/100000001720-d26f h7-ast3*CLI> Sip read: ACK sip:720@[Asterisk IP address] SIP/2.0 Via: SIP/2.0/UDP 83.91.245.154:10954;branch=z9hG4bK-mpi0ehd5lnr2;rport From: "Jon Brüel" ;tag=0nlcat16wf To: ;tag=as0ca655ac Call-ID: 3c26d7385e58-9xc2zuoc7dgt@192-168-100-107 CSeq: 2 ACK Max-Forwards: 70 Contact: Content-Length: 0 9 headers, 0 lines 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:31pwmq0oKyAI@130.228.38.63 SIP/2.0 Via: SIP/2.0/UDP [Asterisk IP address]:5060;branch=z9hG4bK68b67f3d;rport From: "asterisk" ;tag=as30f86cc9 To: Contact: Call-ID: 6192b1fe1da62ae009747773364ea4eb@[Asterisk IP address] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 42 Messages-Waiting: no Voice-Message: 0/0 (NAT) to 130.228.38.63:5060 Scheduling destruction of call '6192b1fe1da62ae009747773364ea4eb@[Asterisk IP address]' in 15000 ms h7-ast3*CLI> Sip read: SIP/2.0 200 Ok Via: SIP/2.0/UDP [Asterisk IP address]:5060;branch=z9hG4bK68b67f3d;received=[Asterisk IP address];rport=5060 From: "asterisk" ;tag=as30f86cc9 To: Call-ID: 6192b1fe1da62ae009747773364ea4eb@[Asterisk IP address] CSeq: 102 NOTIFY Content-Length: 0 7 headers, 0 lines Destroying call '6192b1fe1da62ae009747773364ea4eb@[Asterisk IP address]' Destroying call '2297362386238wyKo@[Audiocode IP address]' Destroying call '173853110631106QRpG@[Audiocode IP address]' h7-ast3*CLI> Sip read: SUBSCRIBE sip:100000001720@[Asterisk IP address] SIP/2.0 Via: SIP/2.0/UDP [Audiocode IP address];branch=z9hG4bKacKegsXiL From: ;tag=1c25247 To: "Jon Brüel" ;tag=as4810e82a Call-ID: 6da3e76f084570156f90d8597aa68af0@[Asterisk IP address] CSeq: 169837 SUBSCRIBE Supported: em,timer,100rel Expires: 9999 Event: telephone-event User-Agent: Audiocodes-Sip-Gateway/MP-104 FXS/v.4.30.19.7 Content-Length: 0 11 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to [Audiocode IP address] : 5060 (NAT) Transmitting (NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP [Audiocode IP address];branch=z9hG4bKacKegsXiL;received=[Audiocode IP address];rport=5060 From: ;tag=1c25247 To: "Jon Brüel" ;tag=as4810e82a Call-ID: 6da3e76f084570156f90d8597aa68af0@[Asterisk IP address] CSeq: 169837 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 9999 Contact: ;expires=9999 Content-Length: 0 to [Audiocode IP address]:5060 Scheduling destruction of call '6da3e76f084570156f90d8597aa68af0@[Asterisk IP address]' in 10009000 ms set_destination: Parsing for address/port to send to set_destination: set destination to [Audiocode IP address], port 5060 Reliably Transmitting: NOTIFY sip:100000001720@[Audiocode IP address] SIP/2.0 Via: SIP/2.0/UDP [Asterisk IP address]:5060;branch=z9hG4bK7a4cb600;rport From: "Jon Brüel" ;tag=as4810e82a To: ;tag=1c25247 Contact: Call-ID: 6da3e76f084570156f90d8597aa68af0@[Asterisk IP address] CSeq: 103 NOTIFY User-Agent: Asterisk PBX Event: dialog Content-Type: application/dialog-info+xml Content-Length: 221 confirmed (NAT) to [Audiocode IP address]:5060 h7-ast3*CLI> Disconnected from Asterisk server Executing last minute cleanups Asterisk cleanly ending (0).