extensions.conf: --------------- [outbound0] exten => 123/4166666666,1,Dial(SIP/foo) exten => 123,1,NoOp() [authority0] include => outbound0 sip.conf: -------- [budge2] secret=budge2 type=friend host=dynamic ; This device needs to register canreinvite=no ; Typically set to NO if behind NAT disallow=all allow=ulaw context=authority0 callerid=left <4166666666> mailbox=123@default [foo] secret=wordpass type=friend host=dynamic ; This device needs to register canreinvite=no ; Typically set to NO if behind NAT disallow=all allow=ulaw context=authority0 callerid=right <4165551212> mailbox=456@default ------------------- *CLI> show dialplan [ Context 'authority0' created by 'pbx_config' ] Include => 'outbound0' [pbx_config] [ Context 'outbound0' created by 'pbx_config' ] '123' => 1. Dial(SIP/foo) [pbx_config] '123' => 1. NoOp() [pbx_config] [ Context 'parkedcalls' created by 'res_features' ] '700' => 1. Park() [res_features] -= 3 extensions (3 priorities) in 3 contexts. =- ------------------- <-- SIP read from 192.168.20.85:5060: INVITE sip:123@192.168.20.86:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.85;branch=z9hG4bK07a915135e5a7a98 From: ;tag=dc7f184063fea0ef To: Contact: Supported: replaces Call-ID: 271dddfe3d646cbe@192.168.20.85 CSeq: 21452 INVITE User-Agent: Grandstream BT110 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 386 v=0 o=budge2 8000 8000 IN IP4 192.168.20.85 s=SIP Call c=IN IP4 192.168.20.85 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 97 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (13 headers 18 lines) --- Using INVITE request as basis request - 271dddfe3d646cbe@192.168.20.85 Sending to 192.168.20.85 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.20.85:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.20.85;branch=z9hG4bK07a915135e5a7a98;received=192.168.20.85 From: ;tag=dc7f184063fea0ef To: ;tag=as60dee4f9 Call-ID: 271dddfe3d646cbe@192.168.20.85 CSeq: 21452 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ebe0865" Content-Length: 0 --- Scheduling destruction of call '271dddfe3d646cbe@192.168.20.85' in 15000 ms Found user 'budge2' <-- SIP read from 192.168.20.85:5060: ACK sip:123@192.168.20.86:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.85;branch=z9hG4bK07a915135e5a7a98 From: ;tag=dc7f184063fea0ef To: ;tag=as60dee4f9 Contact: Call-ID: 271dddfe3d646cbe@192.168.20.85 CSeq: 21452 ACK User-Agent: Grandstream BT110 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 --- (11 headers 0 lines) --- <-- SIP read from 192.168.20.85:5060: INVITE sip:123@192.168.20.86:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.85;branch=z9hG4bKd3f4f110fab1205c From: ;tag=dc7f184063fea0ef To: Contact: Supported: replaces Proxy-Authorization: Digest username="budge2", realm="asterisk", algorithm=MD5, uri="sip:123@192.168.20.86:5060", nonce="2ebe0865", response="30e1b38f8ad698e929a57ffe2a2ed2b8" Call-ID: 271dddfe3d646cbe@192.168.20.85 CSeq: 21453 INVITE User-Agent: Grandstream BT110 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 386 v=0 o=budge2 8000 8001 IN IP4 192.168.20.85 s=SIP Call c=IN IP4 192.168.20.85 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 97 9 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (14 headers 18 lines) --- Using INVITE request as basis request - 271dddfe3d646cbe@192.168.20.85 Sending to 192.168.20.85 : 5060 (non-NAT) Found user 'budge2' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 97 Found RTP audio format 9 Found RTP audio format 101 Peer audio RTP is at port 192.168.20.85:5004 Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G726-32 Found description format iLBC Found description format G722 Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 123 in authority0 (domain 192.168.20.86) list_route: hop: Transmitting (no NAT) to 192.168.20.85:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.20.85;branch=z9hG4bKd3f4f110fab1205c;received=192.168.20.85 From: ;tag=dc7f184063fea0ef To: Call-ID: 271dddfe3d646cbe@192.168.20.85 CSeq: 21453 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Dial("SIP/budge2-081d3528", "SIP/foo") in new stack We're at 192.168.20.86 port 15098 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 192.168.20.170:5062: INVITE sip:foo@192.168.20.170:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.86:5060;branch=z9hG4bK7289eaac;rport From: "left" ;tag=as68a3ac6c To: Contact: Call-ID: 1ed1c2ac62c60baa75ed22c93ad7cd56@192.168.20.86 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 07 Nov 2007 16:14:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 218 v=0 o=root 10943 10943 IN IP4 192.168.20.86 s=session c=IN IP4 192.168.20.86 t=0 0 m=audio 15098 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called foo <-- SIP read from 192.168.20.170:5062: SIP/2.0 100 Trying To: From: "left" ;tag=as68a3ac6c Call-ID: 1ed1c2ac62c60baa75ed22c93ad7cd56@192.168.20.86 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.20.86:5060;branch=z9hG4bK7289eaac Server: Linksys/SPA941-5.1.8 Content-Length: 0 --- (8 headers 0 lines) --- <-- SIP read from 192.168.20.170:5062: SIP/2.0 180 Ringing To: ;tag=1a172b8194b8799fi2 From: "left" ;tag=as68a3ac6c Call-ID: 1ed1c2ac62c60baa75ed22c93ad7cd56@192.168.20.86 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.20.86:5060;branch=z9hG4bK7289eaac Server: Linksys/SPA941-5.1.8 Content-Length: 0 --- (8 headers 0 lines) --- -- SIP/foo-081d8a68 is ringing Transmitting (no NAT) to 192.168.20.85:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.20.85;branch=z9hG4bKd3f4f110fab1205c;received=192.168.20.85 From: ;tag=dc7f184063fea0ef To: ;tag=as0d33fe29 Call-ID: 271dddfe3d646cbe@192.168.20.85 CSeq: 21453 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- <-- SIP read from 192.168.20.170:5062: SIP/2.0 200 OK To: ;tag=1a172b8194b8799fi2 From: "left" ;tag=as68a3ac6c Call-ID: 1ed1c2ac62c60baa75ed22c93ad7cd56@192.168.20.86 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.20.86:5060;branch=z9hG4bK7289eaac Contact: Server: Linksys/SPA941-5.1.8 Content-Length: 212 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces Content-Type: application/sdp v=0 o=- 6349997 6349997 IN IP4 192.168.20.170 s=- c=IN IP4 192.168.20.170 t=0 0 m=audio 16394 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (12 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.20.170:16394 Found description format PCMU Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.20.170, port 5062 Transmitting (no NAT) to 192.168.20.170:5062: ACK sip:foo@192.168.20.170:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.86:5060;branch=z9hG4bK689e647e;rport From: "left" ;tag=as68a3ac6c To: ;tag=1a172b8194b8799fi2 Contact: Call-ID: 1ed1c2ac62c60baa75ed22c93ad7cd56@192.168.20.86 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/foo-081d8a68 answered SIP/budge2-081d3528 We're at 192.168.20.86 port 13552 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.20.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.85;branch=z9hG4bKd3f4f110fab1205c;received=192.168.20.85 From: ;tag=dc7f184063fea0ef To: ;tag=as0d33fe29 Call-ID: 271dddfe3d646cbe@192.168.20.85 CSeq: 21453 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 218 v=0 o=root 10943 10943 IN IP4 192.168.20.86 s=session c=IN IP4 192.168.20.86 t=0 0 m=audio 13552 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/budge2-081d3528 and SIP/foo-081d8a68 <-- SIP read from 192.168.20.85:5060: ACK sip:123@192.168.20.86 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.85;branch=z9hG4bKd999f83884498e08 From: ;tag=dc7f184063fea0ef To: ;tag=as0d33fe29 Contact: Proxy-Authorization: Digest username="budge2", realm="asterisk", algorithm=MD5, uri="sip:123@192.168.20.86", nonce="2ebe0865", response="7100abe579b397fd91627ec4b5f69cc1" Call-ID: 271dddfe3d646cbe@192.168.20.85 CSeq: 21453 ACK User-Agent: Grandstream BT110 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 --- (12 headers 0 lines) --- <-- SIP read from 192.168.20.170:5062: BYE sip:4166666666@192.168.20.86 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.170:5062;branch=z9hG4bK-3cea4f5a From: ;tag=1a172b8194b8799fi2 To: "left" ;tag=as68a3ac6c Call-ID: 1ed1c2ac62c60baa75ed22c93ad7cd56@192.168.20.86 CSeq: 101 BYE Max-Forwards: 70 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 0 --- (9 headers 0 lines) --- Sending to 192.168.20.170 : 5062 (non-NAT) Transmitting (no NAT) to 192.168.20.170:5062: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.170:5062;branch=z9hG4bK-3cea4f5a;received=192.168.20.170 From: ;tag=1a172b8194b8799fi2 To: "left" ;tag=as68a3ac6c Call-ID: 1ed1c2ac62c60baa75ed22c93ad7cd56@192.168.20.86 CSeq: 101 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- == Spawn extension (authority0, 123, 1) exited non-zero on 'SIP/budge2-081d3528' Scheduling destruction of call '271dddfe3d646cbe@192.168.20.85' in 32000 ms set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.20.85, port 5060 Reliably Transmitting (no NAT) to 192.168.20.85:5060: BYE sip:budge2@192.168.20.85 SIP/2.0 Via: SIP/2.0/UDP 192.168.20.86:5060;branch=z9hG4bK49823b4e;rport From: ;tag=as0d33fe29 To: ;tag=dc7f184063fea0ef Call-ID: 271dddfe3d646cbe@192.168.20.85 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <-- SIP read from 192.168.20.85:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.20.86:5060;branch=z9hG4bK49823b4e;rport From: ;tag=as0d33fe29 To: ;tag=dc7f184063fea0ef Call-ID: 271dddfe3d646cbe@192.168.20.85 CSeq: 102 BYE User-Agent: Grandstream BT110 1.0.8.33 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 --- (11 headers 0 lines) --- Destroying call '1ed1c2ac62c60baa75ed22c93ad7cd56@192.168.20.86' Destroying call '271dddfe3d646cbe@192.168.20.85' <-- SIP read from 192.168.20.85:5060: --- (0 headers 0 lines) Nat keepalive ---