One note on the config here: The peer name for the network box here is "customerbox". It's a little ambiguous, bit it keeps things nice and clean on the networkbox side. -- Executing Dial("SIP/02015-08c56618", "SIP/8005551212@customerbox|60|j") in new stack We're at 10.2.1.1 port 16528 Adding codec 0x10 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 10.1.1.1:5060: INVITE sip:8005551212@sip1.networkbox.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK4ae307c3;rport From: "Jeff Pyle" ;tag=as3aa69fd2 To: Contact: Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 102 INVITE User-Agent: Asterisk PBX brns Max-Forwards: 70 Date: Tue, 26 Sep 2006 16:56:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 217 v=0 o=root 2460 2460 IN IP4 10.2.1.1 s=session c=IN IP4 10.2.1.1 t=0 0 m=audio 16528 RTP/AVP 111 101 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 8005551212@customerbox hare*CLI> <-- SIP read from 10.1.1.1:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK4ae307c3;rport;received=10.2.1.1 From: "Jeff Pyle" ;tag=as3aa69fd2 To: ;tag=as141c4902 Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest algorithm=MD5, realm="funhouse", nonce="653fc1b1" Content-Length: 0 --- (11 headers 0 lines)--- Transmitting (no NAT) to 10.1.1.1:5060: ACK sip:8005551212@sip1.networkbox.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK4ae307c3;rport From: "Jeff Pyle" ;tag=as3aa69fd2 To: ;tag=as141c4902 Contact: Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 102 ACK User-Agent: Asterisk PBX brns Max-Forwards: 70 Content-Length: 0 --- We're at 10.2.1.1 port 16528 Adding codec 0x10 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.1.1.1:5060: INVITE sip:8005551212@sip1.networkbox.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK12132483;rport From: "Jeff Pyle" ;tag=as3aa69fd2 To: Contact: Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 103 INVITE User-Agent: Asterisk PBX brns Max-Forwards: 70 Proxy-Authorization: Digest username="customerbox", realm="funhouse", algorithm=MD5, uri="sip:8005551212@sip1.networkbox.com;user=phone", nonce="653fc1b1", response="aeb14bd594969777d59cfd8d0368f164", opaque="" Date: Tue, 26 Sep 2006 16:56:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 217 v=0 o=root 2460 2461 IN IP4 10.2.1.1 s=session c=IN IP4 10.2.1.1 t=0 0 m=audio 16528 RTP/AVP 111 101 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- hare*CLI> <-- SIP read from 10.1.1.1:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK12132483;rport;received=10.2.1.1 From: "Jeff Pyle" ;tag=as3aa69fd2 To: Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- Scheduling destruction of call '267123ab71c907d24adb6dac1c8e0937@10.2.1.1' in 32000 ms == Spawn extension (macro-dialout, s, 21) exited non-zero on 'SIP/02015-08c56618' in macro 'dialout' == Spawn extension (macro-dialout, s, 21) exited non-zero on 'SIP/02015-08c56618' hare*CLI> <-- SIP read from 10.1.1.1:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK12132483;rport;received=10.2.1.1 From: "Jeff Pyle" ;tag=as3aa69fd2 To: ;tag=as1a44a341 Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- Reliably Transmitting (no NAT) to 10.1.1.1:5060: BYE sip:8005551212@sip1.networkbox.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK62b86bdb;rport From: "Jeff Pyle" ;tag=as3aa69fd2 To: ;tag=as1a44a341 Contact: Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 104 BYE User-Agent: Asterisk PBX brns Max-Forwards: 70 Proxy-Authorization: Digest username="customerbox", realm="funhouse", algorithm=MD5, uri="sip:8005551212@sip1.networkbox.com;user=phone", nonce="653fc1b1", response="ebea8e32927b05f5c83a27ae5e6fb574", opaque="" Content-Length: 0 --- Scheduling destruction of call '267123ab71c907d24adb6dac1c8e0937@10.2.1.1' in 32000 ms <-- SIP read from 10.1.1.1:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK12132483;rport;received=10.2.1.1 From: "Jeff Pyle" ;tag=as3aa69fd2 To: ;tag=as1a44a341 Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 225 v=0 o=root 16674 16674 IN IP4 10.1.1.1 s=session c=IN IP4 10.1.1.1 t=0 0 m=audio 40210 RTP/AVP 111 101 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (11 headers 10 lines)--- Found RTP audio format 111 Found RTP audio format 101 Peer audio RTP is at port 10.1.1.1:40210 Found description format G726-32 Found description format telephone-event Capabilities: us - 0x10 (g726), peer - audio=0x10 (g726)/video=0x0 (nothing), combined - 0x10 (g726) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) hare*CLI> <-- SIP read from 10.1.1.1:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK12132483;rport;received=10.2.1.1 From: "Jeff Pyle" ;tag=as3aa69fd2 To: ;tag=as1a44a341 Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- Transmitting (no NAT) to 10.1.1.1:5060: ACK sip:8005551212@sip1.networkbox.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK62b86bdb;rport From: "Jeff Pyle" ;tag=as3aa69fd2 To: ;tag=as1a44a341 Contact: Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 103 ACK User-Agent: Asterisk PBX brns Max-Forwards: 70 Content-Length: 0 --- hare*CLI> <-- SIP read from 10.1.1.1:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK62b86bdb;received=10.2.1.1;rport=5060 From: "Jeff Pyle" ;tag=as3aa69fd2 To: ;tag=as1a44a341 Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 104 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '267123ab71c907d24adb6dac1c8e0937@10.2.1.1' hare*CLI> <-- SIP read from 10.1.1.1:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK12132483;rport;received=10.2.1.1 From: "Jeff Pyle" ;tag=as3aa69fd2 To: ;tag=as1a44a341 Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '267123ab71c907d24adb6dac1c8e0937@10.2.1.1' hare*CLI> <-- SIP read from 10.1.1.1:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK12132483;rport;received=10.2.1.1 From: "Jeff Pyle" ;tag=as3aa69fd2 To: ;tag=as1a44a341 Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '267123ab71c907d24adb6dac1c8e0937@10.2.1.1' hare*CLI> <-- SIP read from 10.1.1.1:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK12132483;rport;received=10.2.1.1 From: "Jeff Pyle" ;tag=as3aa69fd2 To: ;tag=as1a44a341 Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '267123ab71c907d24adb6dac1c8e0937@10.2.1.1' hare*CLI> <-- SIP read from 10.1.1.1:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK12132483;rport;received=10.2.1.1 From: "Jeff Pyle" ;tag=as3aa69fd2 To: ;tag=as1a44a341 Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '267123ab71c907d24adb6dac1c8e0937@10.2.1.1' hare*CLI> <-- SIP read from 10.1.1.1:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK12132483;rport;received=10.2.1.1 From: "Jeff Pyle" ;tag=as3aa69fd2 To: ;tag=as1a44a341 Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '267123ab71c907d24adb6dac1c8e0937@10.2.1.1' hare*CLI> <-- SIP read from 10.1.1.1:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.2.1.1:5060;branch=z9hG4bK12132483;rport;received=10.2.1.1 From: "Jeff Pyle" ;tag=as3aa69fd2 To: ;tag=as1a44a341 Call-ID: 267123ab71c907d24adb6dac1c8e0937@10.2.1.1 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '267123ab71c907d24adb6dac1c8e0937@10.2.1.1'