**** SIP.CONF **** [general] context=default bindport=5060 bindaddr=0.0.0.0 disallow=all allow=alaw [marcus] type=peer context=default subscribecontext=default notifyringing=yes secret=*** host=dynamic qualify=100 nat=never canreinvite=no dtmfmode=info call-limit=1 [mobile] type=peer context=default subscribecontext=default notifyringing=yes secret=*** host=dynamic qualify=1000 nat=never canreinvite=no dtmfmode=info call-limit=1 [1000] type=peer context=default subscribecontext=default notifyringing=yes secret=1000 host=dynamic qualify=100 nat=never canreinvite=no dtmfmode=info call-limit=4 **** EXTENSIONS.CONF **** [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [globals] [default] exten => 101,hint,SIP/marcus exten => 101,1,Dial(SIP/marcus,20,r) exten => 103,hint,SIP/mobile exten => 103,1,Dial(SIP/mobile,20,m) exten => 1000,hint,SIP/1000 exten => 1000,1,Dial(SIP/1000) **** SIP DEBUG OUTPUT **** Asterisk: 192.168.172.11 Extension 101: 192.168.172.211 Extension 103: 192.168.172.237 Extension 1000: 192.168.172.235 (the only phone subscribing for notifies) <-- SIP read from 192.168.172.235:5060: REGISTER sip:192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bKc298902018ea2c2e From: ;tag=73b5d060031ae56b To: Contact: Call-ID: d5737585ab7334bd@192.168.172.235 CSeq: 10001 REGISTER Expires: 900 User-Agent: Grandstream GXP2000 1.1.0.13 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 --- (12 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.172.235 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.172.235:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bKc298902018ea2c2e;received=192.168.172.235 From: ;tag=73b5d060031ae56b To: Call-ID: d5737585ab7334bd@192.168.172.235 CSeq: 10001 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.172.235:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bKc298902018ea2c2e;received=192.168.172.235 From: ;tag=73b5d060031ae56b To: ;tag=as76a89c26 Call-ID: d5737585ab7334bd@192.168.172.235 CSeq: 10001 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="4b92514b" Content-Length: 0 --- Scheduling destruction of call 'd5737585ab7334bd@192.168.172.235' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.172.235:5060: REGISTER sip:192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bKbb6bba21595e3b68 From: ;tag=73b5d060031ae56b To: Contact: Authorization: Digest username="1000", realm="asterisk", algorithm=MD5, uri="sip:192.168.172.11", nonce="4b92514b", response="e1436a75978c38007d1a41026693e1c7" Call-ID: d5737585ab7334bd@192.168.172.235 CSeq: 10002 REGISTER Expires: 900 User-Agent: Grandstream GXP2000 1.1.0.13 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 --- (13 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.172.235 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.172.235:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bKbb6bba21595e3b68;received=192.168.172.235 From: ;tag=73b5d060031ae56b To: Call-ID: d5737585ab7334bd@192.168.172.235 CSeq: 10002 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bKbb6bba21595e3b68;received=192.168.172.235 From: ;tag=73b5d060031ae56b To: ;tag=as76a89c26 Call-ID: d5737585ab7334bd@192.168.172.235 CSeq: 10002 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 900 Contact: ;expires=900 Date: Sat, 10 Jun 2006 12:22:00 GMT Content-Length: 0 --- Scheduling destruction of call 'd5737585ab7334bd@192.168.172.235' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SUBSCRIBE sip:101@192.168.172.11;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bK3bcdcce2e04e1a92 From: ;tag=0a646e4efb2d2f0e To: Contact: Call-ID: 2566e8c3f5ac1a4c@192.168.172.235 CSeq: 1101 SUBSCRIBE User-Agent: Grandstream GXP2000 1.1.0.13 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Event: dialog Expires: 900 Accept: application/dialog-info+xml Content-Length: 0 --- (14 headers 0 lines)--- Using latest SUBSCRIBE request as basis request Sending to 192.168.172.235 : 5060 (non-NAT) Found peer '1000' Transmitting (no NAT) to 192.168.172.235:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bK3bcdcce2e04e1a92;received=192.168.172.235 From: ;tag=0a646e4efb2d2f0e To: ;tag=as441b2fe8 Call-ID: 2566e8c3f5ac1a4c@192.168.172.235 CSeq: 1101 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="77bddcbd" Content-Length: 0 --- Scheduling destruction of call '2566e8c3f5ac1a4c@192.168.172.235' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SUBSCRIBE sip:102@192.168.172.11;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bK021104f38543116f From: ;tag=e224574d72705af2 To: Contact: Call-ID: bea52678f6e425ad@192.168.172.235 CSeq: 2201 SUBSCRIBE User-Agent: Grandstream GXP2000 1.1.0.13 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Event: dialog Expires: 900 Accept: application/dialog-info+xml Content-Length: 0 --- (14 headers 0 lines)--- Using latest SUBSCRIBE request as basis request Sending to 192.168.172.235 : 5060 (non-NAT) Found peer '1000' Transmitting (no NAT) to 192.168.172.235:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bK021104f38543116f;received=192.168.172.235 From: ;tag=e224574d72705af2 To: ;tag=as60e44d6c Call-ID: bea52678f6e425ad@192.168.172.235 CSeq: 2201 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="722ff508" Content-Length: 0 --- Scheduling destruction of call 'bea52678f6e425ad@192.168.172.235' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SUBSCRIBE sip:103@192.168.172.11;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bKc854bafa4c9a6054 From: ;tag=1db389b1ed54bf83 To: Contact: Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 3301 SUBSCRIBE User-Agent: Grandstream GXP2000 1.1.0.13 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Event: dialog Expires: 900 Accept: application/dialog-info+xml Content-Length: 0 --- (14 headers 0 lines)--- Using latest SUBSCRIBE request as basis request Sending to 192.168.172.235 : 5060 (non-NAT) Found peer '1000' Transmitting (no NAT) to 192.168.172.235:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bKc854bafa4c9a6054;received=192.168.172.235 From: ;tag=1db389b1ed54bf83 To: ;tag=as0d3e5360 Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 3301 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="1c0132f5" Content-Length: 0 --- Scheduling destruction of call 'd69bef7e20c470dd@192.168.172.235' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SUBSCRIBE sip:104@192.168.172.11;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bK7f18814e83ab05f7 From: ;tag=148162b0d6d6f108 To: Contact: Call-ID: 915776ed210a8554@192.168.172.235 CSeq: 4401 SUBSCRIBE User-Agent: Grandstream GXP2000 1.1.0.13 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Event: dialog Expires: 900 Accept: application/dialog-info+xml Content-Length: 0 --- (14 headers 0 lines)--- Using latest SUBSCRIBE request as basis request Sending to 192.168.172.235 : 5060 (non-NAT) Found peer '1000' Transmitting (no NAT) to 192.168.172.235:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bK7f18814e83ab05f7;received=192.168.172.235 From: ;tag=148162b0d6d6f108 To: ;tag=as1ef9697c Call-ID: 915776ed210a8554@192.168.172.235 CSeq: 4401 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="399c61e7" Content-Length: 0 --- Scheduling destruction of call '915776ed210a8554@192.168.172.235' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SUBSCRIBE sip:1000@192.168.172.11;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bK347a56c548d7d970 From: ;tag=63939ca009e76d0c To: Contact: Call-ID: 2b7e9b22ef9d67e5@192.168.172.235 CSeq: 5501 SUBSCRIBE User-Agent: Grandstream GXP2000 1.1.0.13 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Event: dialog Expires: 900 Accept: application/dialog-info+xml Content-Length: 0 --- (14 headers 0 lines)--- Using latest SUBSCRIBE request as basis request Sending to 192.168.172.235 : 5060 (non-NAT) Found peer '1000' Transmitting (no NAT) to 192.168.172.235:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bK347a56c548d7d970;received=192.168.172.235 From: ;tag=63939ca009e76d0c To: ;tag=as0c5c7abe all-ID: 2b7e9b22ef9d67e5@192.168.172.235 CSeq: 5501 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="5da4e71e" Content-Length: 0 --- Scheduling destruction of call '2b7e9b22ef9d67e5@192.168.172.235' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SUBSCRIBE sip:101@192.168.172.11;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bKbb897e31b8da9530 From: ;tag=0a646e4efb2d2f0e To: Contact: Authorization: Digest username="1000", realm="asterisk", algorithm=MD5, uri="sip:101@192.168.172.11;user=p", nonce="77bddcbd", response="efbf6808e2d357cf0d439cc88e4bba7a" Call-ID: 2566e8c3f5ac1a4c@192.168.172.235 CSeq: 1102 SUBSCRIBE User-Agent: Grandstream GXP2000 1.1.0.13 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Event: dialog Expires: 900 Accept: application/dialog-info+xml Content-Length: 0 --- (15 headers 0 lines)--- Found peer '1000' Looking for 101 in default (domain 192.168.172.11) Scheduling destruction of call '2566e8c3f5ac1a4c@192.168.172.235' in 910000 ms Transmitting (no NAT) to 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bKbb897e31b8da9530;received=192.168.172.235 From: ;tag=0a646e4efb2d2f0e To: ;tag=as441b2fe8 Call-ID: 2566e8c3f5ac1a4c@192.168.172.235 CSeq: 1102 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 900 Contact: ;expires=900 Content-Length: 0 --- Reliably Transmitting (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK3bfc5b8e From: ;tag=as441b2fe8 To: ;tag=0a646e4efb2d2f0e Contact: Call-ID: 2566e8c3f5ac1a4c@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- Destroying call 'dd563593b2c4b6ac@192.168.172.235' asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SUBSCRIBE sip:102@192.168.172.11;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bK9a6f80cbc00e8fa3 From: ;tag=e224574d72705af2 To: Contact: Authorization: Digest username="1000", realm="asterisk", algorithm=MD5, uri="sip:102@192.168.172.11;user=p", nonce="722ff508", response="a51f209fd25c45bfdc48fbd936601fcd" Call-ID: bea52678f6e425ad@192.168.172.235 CSeq: 2202 SUBSCRIBE User-Agent: Grandstream GXP2000 1.1.0.13 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Event: dialog Expires: 900 Accept: application/dialog-info+xml Content-Length: 0 --- (15 headers 0 lines)--- Found peer '1000' Looking for 102 in default (domain 192.168.172.11) Scheduling destruction of call 'bea52678f6e425ad@192.168.172.235' in 910000 ms Transmitting (no NAT) to 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bK9a6f80cbc00e8fa3;received=192.168.172.235 From: ;tag=e224574d72705af2 To: ;tag=as60e44d6c Call-ID: bea52678f6e425ad@192.168.172.235 CSeq: 2202 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 900 Contact: ;expires=900 Content-Length: 0 --terisk*CLI> Reliably Transmitting (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK6a4f861f From: ;tag=as60e44d6c To: ;tag=e224574d72705af2 Contact: Call-ID: bea52678f6e425ad@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 204 confirmed --- Retransmitting #1 (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK3bfc5b8e From: ;tag=as441b2fe8 To: ;tag=0a646e4efb2d2f0e Contact: Call-ID: 2566e8c3f5ac1a4c@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- Retransmitting #2 (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK3bfc5b8e From: ;tag=as441b2fe8 To: ;tag=0a646e4efb2d2f0e Contact: Call-ID: 2566e8c3f5ac1a4c@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- Destroying call '87a5727883e4c1ad@192.168.172.235' <-- SIP read from 192.168.172.235:5060: SUBSCRIBE sip:103@192.168.172.11;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bK974096b2563faaf1 From: ;tag=1db389b1ed54bf83 To: Contact: Authorization: Digest username="1000", realm="asterisk", algorithm=MD5, uri="sip:103@192.168.172.11;user=p", nonce="1c0132f5", response="61f9d9889c87b4c539edfe80115789e0" Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 3302 SUBSCRIBE User-Agent: Grandstream GXP2000 1.1.0.13 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Event: dialog Expires: 900 Accept: application/dialog-info+xml Content-Length: 0 --- (15 headers 0 lines)--- Found peer '1000' Looking for 103 in default (domain 192.168.172.11) Scheduling destruction of call 'd69bef7e20c470dd@192.168.172.235' in 910000 ms Transmitting (no NAT) to 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bK974096b2563faaf1;received=192.168.172.235 From: ;tag=1db389b1ed54bf83 To: ;tag=as0d3e5360 Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 3302 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 900 Contact: ;expires=900 Content-Length: 0 --- Reliably Transmitting (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK767086d2 From: ;tag=as0d3e5360 To: ;tag=1db389b1ed54bf83 Contact: Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- Destroying call 'ff8b981e7c933de5@192.168.172.235' <-- SIP read from 192.168.172.235:5060: SUBSCRIBE sip:104@192.168.172.11;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bK9c53813944da020b From: ;tag=148162b0d6d6f108 To: Contact: Authorization: Digest username="1000", realm="asterisk", algorithm=MD5, uri="sip:104@192.168.172.11;user=p", nonce="399c61e7", response="cfbf7cb539900d24601dc701b54f46f0" Call-ID: 915776ed210a8554@192.168.172.235 CSeq: 4402 SUBSCRIBE User-Agent: Grandstream GXP2000 1.1.0.13 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Event: dialog Expires: 900 Accept: application/dialog-info+xml Content-Length: 0 --- (15 headers 0 lines)--- Found peer '1000' Looking for 104 in default (domain 192.168.172.11) Scheduling destruction of call '915776ed210a8554@192.168.172.235' in 910000 ms Transmitting (no NAT) to 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bK9c53813944da020b;received=192.168.172.235 From: ;tag=148162b0d6d6f108 To: ;tag=as1ef9697c Call-ID: 915776ed210a8554@192.168.172.235 CSeq: 4402 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 900 Contact: ;expires=900 Content-Length: 0 --- Reliably Transmitting (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK2720b73f From: ;tag=as1ef9697c To: ;tag=148162b0d6d6f108 Contact: Call-ID: 915776ed210a8554@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- Destroying call 'e35767ed210ae554@192.168.172.235' <-- SIP read from 192.168.172.235:5060: SUBSCRIBE sip:1000@192.168.172.11;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bKc305f88483066281 From: ;tag=63939ca009e76d0c To: Contact: Authorization: Digest username="1000", realm="asterisk", algorithm=MD5, uri="sip:1000@192.168.172.11;user=p", nonce="5da4e71e", response="fc90fb9a20d66d7b87e9a05a43f527a3" Call-ID: 2b7e9b22ef9d67e5@192.168.172.235 CSeq: 5502 SUBSCRIBE User-Agent: Grandstream GXP2000 1.1.0.13 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Event: dialog Expires: 900 Accept: application/dialog-info+xml Content-Length: 0 --- (15 headers 0 lines)--- Found peer '1000' Looking for 1000 in default (domain 192.168.172.11) Scheduling destruction of call '2b7e9b22ef9d67e5@192.168.172.235' in 910000 ms Transmitting (no NAT) to 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.235:5060;branch=z9hG4bKc305f88483066281;received=192.168.172.235 From: ;tag=63939ca009e76d0c To: ;tag=as0c5c7abe Call-ID: 2b7e9b22ef9d67e5@192.168.172.235 CSeq: 5502 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 900 Contact: ;expires=900 Content-Length: 0 asterisk*CLI> --- Reliably Transmitting (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK60b07e64 From: ;tag=as0c5c7abe To: ;tag=63939ca009e76d0c Contact: Call-ID: 2b7e9b22ef9d67e5@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 207 terminated --- Destroying call '0e7e1d2250ad57ed@192.168.172.235' <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK3bfc5b8e From: ;tag=as441b2fe8 To: ;tag=0a646e4efb2d2f0e Call-ID: 2566e8c3f5ac1a4c@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK6a4f861f From: ;tag=as60e44d6c To: ;tag=e224574d72705af2 Call-ID: bea52678f6e425ad@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK3bfc5b8e From: ;tag=as441b2fe8 To: ;tag=0a646e4efb2d2f0e Call-ID: 2566e8c3f5ac1a4c@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- Retransmitting #1 (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK767086d2 From: ;tag=as0d3e5360 To: ;tag=1db389b1ed54bf83 Contact: Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 asterisk*CLI> terminated --- Retransmitting #1 (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK2720b73f From: ;tag=as1ef9697c To: ;tag=148162b0d6d6f108 Contact: all-ID: 915776ed210a8554@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- Retransmitting #1 (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK60b07e64 From: ;tag=as0c5c7abe To: ;tag=63939ca009e76d0c Contact: Call-ID: 2b7e9b22ef9d67e5@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 207 terminated --- asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK3bfc5b8e From: ;tag=as441b2fe8 To: ;tag=0a646e4efb2d2f0e Call-ID: 2566e8c3f5ac1a4c@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- Retransmitting #2 (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK767086d2 From: ;tag=as0d3e5360 To: ;tag=1db389b1ed54bf83 Contact: Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 asterisk*CLI> terminated --- Retransmitting #2 (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK2720b73f From: ;tag=as1ef9697c To: ;tag=148162b0d6d6f108 Contact: all-ID: 915776ed210a8554@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- Retransmitting #2 (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK60b07e64 From: ;tag=as0c5c7abe To: ;tag=63939ca009e76d0c Contact: Call-ID: 2b7e9b22ef9d67e5@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 207 terminated --- asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK767086d2 From: ;tag=as0d3e5360 To: ;tag=1db389b1ed54bf83 Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK2720b73f From: ;tag=as1ef9697c To: ;tag=148162b0d6d6f108 Call-ID: 915776ed210a8554@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK60b07e64 From: ;tag=as0c5c7abe To: ;tag=63939ca009e76d0c Call-ID: 2b7e9b22ef9d67e5@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 asterisk*CLI> --- (11 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK767086d2 From: ;tag=as0d3e5360 To: ;tag=1db389b1ed54bf83 Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK2720b73f From: ;tag=as1ef9697c To: ;tag=148162b0d6d6f108 Call-ID: 915776ed210a8554@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK60b07e64 From: ;tag=as0c5c7abe To: ;tag=63939ca009e76d0c Call-ID: 2b7e9b22ef9d67e5@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 asterisk*CLI> --- (11 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK767086d2 From: ;tag=as0d3e5360 To: ;tag=1db389b1ed54bf83 Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK2720b73f From: ;tag=as1ef9697c To: ;tag=148162b0d6d6f108 Call-ID: 915776ed210a8554@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK60b07e64 From: ;tag=as0c5c7abe To: ;tag=63939ca009e76d0c Call-ID: 2b7e9b22ef9d67e5@192.168.172.235 CSeq: 102 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- *** and now a call from 103 to 101 *** <-- SIP read from 192.168.172.237:5060: INVITE sip:101@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.237:5060;rport;branch=z9hG4bK3237775271 From: "Marcus" ;tag=2405749057 To: Call-ID: 285982932@192.168.172.237 CSeq: 1000 INVITE Contact: max-forwards: 70 user-agent: UTSTARCOM F1000/Device ID-0007ba25de69 Content-Type: application/sdp Content-Length: 307 v=0 o=mobile 123456 654774 IN IP4 192.168.172.237 s=none c=IN IP4 192.168.172.237 t=0 0 m=audio 10010 RTP/AVP 8 0 18 2 101 a=ptime:30 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729A/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (11 headers 14 lines)--- Using INVITE request as basis request - 285982932@192.168.172.237 Sending to 192.168.172.237 : 5060 (NAT) Found peer 'mobile' Reliably Transmitting (no NAT) to 192.168.172.237:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.172.237:5060;rport;branch=z9hG4bK3237775271;received=192.168.172.237 From: "Marcus" ;tag=2405749057 To: ;tag=as23f532a9 Call-ID: 285982932@192.168.172.237 CSeq: 1000 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="424e5cab" Content-Length: 0 --- Scheduling destruction of call '285982932@192.168.172.237' in 15000 ms asterisk*CLI> <-- SIP read from 192.168.172.237:5060: ACK sip:101@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.237:5060;rport;branch=z9hG4bK3237775271 From: "Marcus" ;tag=2405749057 To: ;tag=as23f532a9 Call-ID: 285982932@192.168.172.237 CSeq: 1000 ACK Content-Length: 0 --- (7 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.237:5060: INVITE sip:101@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.237:5060;rport;branch=z9hG4bK1258444218 From: "Marcus" ;tag=2405749057 To: Call-ID: 285982932@192.168.172.237 CSeq: 1001 INVITE Contact: Proxy-Authorization: Digest username="mobile", realm="asterisk", nonce="424e5cab", uri="sip:101@192.168.172.11", response="06703cee0fa3b484c014f2b8878d0458", algorithm=MD5 max-forwards: 70 user-agent: UTSTARCOM F1000/Device ID-0007ba25de69 Content-Type: application/sdp Content-Length: 307 v=0 o=mobile 123456 654774 IN IP4 192.168.172.237 s=none c=IN IP4 192.168.172.237 t=0 0 m=audio 10010 RTP/AVP 8 0 18 2 101 a=ptime:30 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729A/8000 a=fmtp:18 annexb=no a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (12 headers 14 lines)--- Using INVITE request as basis request - 285982932@192.168.172.237 Sending to 192.168.172.237 : 5060 (NAT) Found peer 'mobile' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 101 Peer audio RTP is at port 192.168.172.237:10010 Found description format PCMA Found description format PCMU Found description format G729A Found description format G726-32 Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x11c (ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 101 in default (domain 192.168.172.11) list_route: hop: Transmitting (no NAT) to 192.168.172.237:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.172.237:5060;rport;branch=z9hG4bK1258444218;received=192.168.172.237 From: "Marcus" ;tag=2405749057 To: Call-ID: 285982932@192.168.172.237 CSeq: 1001 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Reliably Transmitting (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.235:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK286fa731 From: ;tag=as0d3e5360 To: ;tag=1db389b1ed54bf83 Contact: Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 204 confirmed --- -- Executing Dial("SIP/mobile-8689", "SIP/marcus|20|r") in new stack We're at 192.168.172.11 port 13314 Adding codec 0x8 (alaw) to SDP 13 headers, 8 lines asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK286fa731 From: ;tag=as0d3e5360 To: ;tag=1db389b1ed54bf83 Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 103 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- Reliably Transmitting (no NAT) to 192.168.172.211:5060: INVITE sip:marcus@192.168.172.211 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK346786a0 From: "Marcus" ;tag=as0910e95d To: Contact: Call-ID: 6d7083fb641920222fab716d6c5dd9a3@192.168.172.11 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sat, 10 Jun 2006 12:22:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 164 v=0 o=root 10046 10046 IN IP4 192.168.172.11 s=session c=IN IP4 192.168.172.11 t=0 0 m=audio 13314 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - --- -- Called marcus Transmitting (no NAT) to 192.168.172.237:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.172.237:5060;rport;branch=z9hG4bK1258444218;received=192.168.172.237 From: "Marcus" ;tag=2405749057 To: ;tag=as37f5c549 Call-ID: 285982932@192.168.172.237 CSeq: 1001 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- asterisk*CLI> <-- SIP read from 192.168.172.211:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK346786a0 From: "Marcus" ;tag=as0910e95d To: Call-ID: 6d7083fb641920222fab716d6c5dd9a3@192.168.172.11 CSeq: 102 INVITE User-Agent: Grandstream BT110 1.0.8.16 Content-Length: 0 --- (8 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.211:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK346786a0 From: "Marcus" ;tag=as0910e95d To: ;tag=ed6bdfd0cc8b9601 Call-ID: 6d7083fb641920222fab716d6c5dd9a3@192.168.172.11 CSeq: 102 INVITE User-Agent: Grandstream BT110 1.0.8.16 Content-Length: 0 --- (8 headers 0 lines)--- Reliably Transmitting (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.235:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK4ba14774 From: ;tag=as441b2fe8 To: ;tag=0a646e4efb2d2f0e Contact: Call-ID: 2566e8c3f5ac1a4c@192.168.172.235 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 204 confirmed --- -- SIP/marcus-098b is ringing asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK4ba14774 From: ;tag=as441b2fe8 To: ;tag=0a646e4efb2d2f0e Call-ID: 2566e8c3f5ac1a4c@192.168.172.235 CSeq: 103 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.237:5060: CANCEL sip:101@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.237:5060;rport;branch=z9hG4bK1258444218 From: "Marcus" ;tag=2405749057 To: Call-ID: 285982932@192.168.172.237 CSeq: 1001 CANCEL Contact: Proxy-Authorization: Digest username="mobile", realm="asterisk", nonce="424e5cab", uri="sip:101@192.168.172.11", response="06703cee0fa3b484c014f2b8878d0458", algorithm=MD5 max-forwards: 70 user-agent: UTSTARCOM F1000/Device ID-0007ba25de69 Content-Length: 0 --- (11 headers 0 lines)--- Sending to 192.168.172.237 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.172.237:5060: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.172.237:5060;rport;branch=z9hG4bK1258444218;received=192.168.172.237 From: "Marcus" ;tag=2405749057 To: ;tag=as37f5c549 Call-ID: 285982932@192.168.172.237 CSeq: 1001 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Transmitting (NAT) to 192.168.172.237:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.237:5060;rport;branch=z9hG4bK1258444218;received=192.168.172.237 From: "Marcus" ;tag=2405749057 To: ;tag=as37f5c549 Call-ID: 285982932@192.168.172.237 CSeq: 1001 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Reliably Transmitting (no NAT) to 192.168.172.211:5060: CANCEL sip:marcus@192.168.172.211 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK346786a0 From: "Marcus" ;tag=as0910e95d To: Contact: Call-ID: 6d7083fb641920222fab716d6c5dd9a3@192.168.172.11 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '6d7083fb641920222fab716d6c5dd9a3@192.168.172.11' in 32000 ms == Spawn extension (default, 101, 1) exited non-zero on 'SIP/mobile-8689' Reliably Transmitting (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.235:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK1ae02d12 From: ;tag=as441b2fe8 To: ;tag=0a646e4efb2d2f0e Contact: Call-ID: 2566e8c3f5ac1a4c@192.168.172.235 CSeq: 104 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- Reliably Transmitting (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.235:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK6b1ef85f From: ;tag=as0d3e5360 To: ;tag=1db389b1ed54bf83 Contact: Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 104 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- asterisk*CLI> <-- SIP read from 192.168.172.211:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK346786a0 From: "Marcus" ;tag=as0910e95d To: ;tag=ed6bdfd0cc8b9601 Call-ID: 6d7083fb641920222fab716d6c5dd9a3@192.168.172.11 CSeq: 102 CANCEL User-Agent: Grandstream BT110 1.0.8.16 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 --- (11 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.211:5060: SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK346786a0 From: "Marcus" ;tag=as0910e95d To: ;tag=ed6bdfd0cc8b9601 Call-ID: 6d7083fb641920222fab716d6c5dd9a3@192.168.172.11 CSeq: 102 INVITE User-Agent: Grandstream BT110 1.0.8.16 Content-Length: 0 --- (8 headers 0 lines)--- Transmitting (no NAT) to 192.168.172.211:5060: ACK sip:marcus@192.168.172.211 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK346786a0 From: "Marcus" ;tag=as0910e95d To: ;tag=ed6bdfd0cc8b9601 Contact: Call-ID: 6d7083fb641920222fab716d6c5dd9a3@192.168.172.11 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Destroying call '6d7083fb641920222fab716d6c5dd9a3@192.168.172.11' asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK1ae02d12 From: ;tag=as441b2fe8 To: ;tag=0a646e4efb2d2f0e Call-ID: 2566e8c3f5ac1a4c@192.168.172.235 CSeq: 104 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- Retransmitting #1 (no NAT) to 192.168.172.235:5060: NOTIFY sip:1000@192.168.172.235:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK6b1ef85f From: ;tag=as0d3e5360 To: ;tag=1db389b1ed54bf83 Contact: Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 104 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 205 terminated --- asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK6b1ef85f From: ;tag=as0d3e5360 To: ;tag=1db389b1ed54bf83 Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 104 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.235:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.172.11:5060;branch=z9hG4bK6b1ef85f From: ;tag=as0d3e5360 To: ;tag=1db389b1ed54bf83 Call-ID: d69bef7e20c470dd@192.168.172.235 CSeq: 104 NOTIFY User-Agent: Grandstream GXP2000 1.1.0.13 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Supported: replaces, timer, 100rel Content-Length: 0 --- (11 headers 0 lines)--- asterisk*CLI> <-- SIP read from 192.168.172.237:5060: ACK sip:101@192.168.172.11 SIP/2.0 Via: SIP/2.0/UDP 192.168.172.237:5060;rport;branch=z9hG4bK1258444218 From: "Marcus" ;tag=2405749057 To: ;tag=as37f5c549 Call-ID: 285982932@192.168.172.237 CSeq: 1001 ACK Content-Length: 0 **** END ****