sip debug SIP Debugging enabled -- Starting simple switch on 'Zap/3-1' <-- SIP read from 10.10.10.105:5060: NOTIFY sip:10.10.10.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK-ba15c279 From: Sara Hass ;tag=2512817a18f3124co0 To: Call-ID: d741e1ba-38712f0c@10.10.10.105 CSeq: 2319 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Sipura/SPA841-3.1.2(d) Content-Length: 0 --- (10 headers 0 lines)--- Transmitting (no NAT) to 10.10.10.105:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK-ba15c279;received=10.10.10.105 From: Sara Hass ;tag=2512817a18f3124co0 To: ;tag=as6bf18ef3 Call-ID: d741e1ba-38712f0c@10.10.10.105 CSeq: 2319 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call 'd741e1ba-38712f0c@10.10.10.105' -- Executing Macro("Zap/3-1", "stdexten|10|11010") -- Executing Dial("Zap/3-1", "SIP/11010|20|trWw") in new stack -- SIP Seeding peer from astdb: '11010' at 11010@10.10.10.101:5060 for 3600 We're at 10.10.10.10 port 13354 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 10.10.10.101:5060: INVITE sip:11010@10.10.10.101 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK7f34e2f5;rport From: "asterisk" ;tag=as185605f3 To: Contact: Call-ID: 475efa1d3aa20887064887886e6a278b@10.10.10.10 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 21 Jul 2006 07:30:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 238 v=0 o=root 16099 16099 IN IP4 10.10.10.10 s=session c=IN IP4 10.10.10.10 t=0 0 m=audio 13354 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 11010 <-- SIP read from 10.10.10.101:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK7f34e2f5;rport From: "asterisk" ;tag=as185605f3 To: ;tag=9AB14489-19E0390E CSeq: 102 INVITE Call-ID: 475efa1d3aa20887064887886e6a278b@10.10.10.10 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 --- (9 headers 0 lines)--- <-- SIP read from 10.10.10.101:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK7f34e2f5;rport From: "asterisk" ;tag=as185605f3 To: ;tag=9AB14489-19E0390E CSeq: 102 INVITE Call-ID: 475efa1d3aa20887064887886e6a278b@10.10.10.10 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Allow-Events: talk,hold,conference Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/11010-007ae680 is ringing -- SIP Seeding peer from astdb: '11010' at 11010@10.10.10.101:5060 for 3600 -- SIP Seeding peer from astdb: '11010' at 11010@10.10.10.101:5060 for 3600 <-- SIP read from 10.10.10.101:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK7f34e2f5;rport From: "asterisk" ;tag=as185605f3 To: ;tag=9AB14489-19E0390E CSeq: 102 INVITE Call-ID: 475efa1d3aa20887064887886e6a278b@10.10.10.10 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Type: application/sdp Content-Length: 185 v=0 o=- 978310424 978310424 IN IP4 10.10.10.101 s=Polycom IP Phone c=IN IP4 10.10.10.101 t=0 0 m=audio 2264 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 --- (11 headers 8 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.10.10.101:2264 Found description format PCMU Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), 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 10.10.10.101, port 5060 Transmitting (no NAT) to 10.10.10.101:5060: ACK sip:11010@10.10.10.101 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.10:5060;branch=z9hG4bK1f1edf5c;rport From: "asterisk" ;tag=as185605f3 To: ;tag=9AB14489-19E0390E Contact: Call-ID: 475efa1d3aa20887064887886e6a278b@10.10.10.10 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/11010-007ae680 answered Zap/3-1 -- SIP Seeding peer from astdb: '11010' at 11010@10.10.10.101:5060 for 3600 -- SIP Seeding peer from astdb: '11010' at 11010@10.10.10.101:5060 for 3600 <-- SIP read from 10.10.10.101:5060: BYE sip:asterisk@10.10.10.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.101;branch=z9hG4bK2b2fea1c9DA71C87 From: ;tag=9AB14489-19E0390E To: "asterisk" ;tag=as185605f3 CSeq: 1 BYE Call-ID: 475efa1d3aa20887064887886e6a278b@10.10.10.10 Contact: User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines)--- Sending to 10.10.10.101 : 5060 (non-NAT) Transmitting (no NAT) to 10.10.10.101:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.101;branch=z9hG4bK2b2fea1c9DA71C87;received=10.10.10.101 From: ;tag=9AB14489-19E0390E To: "asterisk" ;tag=as185605f3 Call-ID: 475efa1d3aa20887064887886e6a278b@10.10.10.10 CSeq: 1 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 (macro-stdexten, s, 1) exited non-zero on 'Zap/3-1' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'Zap/3-1' -- SIP Seeding peer from astdb: '11010' at 11010@10.10.10.101:5060 for 3600 -- SIP Seeding peer from astdb: '11010' at 11010@10.10.10.101:5060 for 3600 -- Hungup 'Zap/3-1' Jul 21 03:30:46 NOTICE[16111]: chan_sip.c:5336 sip_reregister: -- Re-registration for 7322761242@sip.sipmedia.com REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 69.1.236.33:5060: REGISTER sip:sip.sipmedia.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125:5060;branch=z9hG4bK462d159f;rport From: ;tag=as3062afb7 To: Call-ID: 7e1ee1647f15a5566ec2f4857f56660b@192.168.2.125 CSeq: 106 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="7322761242", realm="sip1.xchangetele.com", algorithm=MD5, uri="sip:sip.sipmedia.com", nonce="44c088066a22ad6ad167418dac1706be8251e333", response="9e7f28f4592e8d667e357448ff34da7d", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- Destroying call '475efa1d3aa20887064887886e6a278b@10.10.10.10' <-- SIP read from 69.1.236.33:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.125:5060;branch=z9hG4bK462d159f;rport=65415 From: ;tag=as3062afb7 To: ;tag=91791a7492f328483c7efb39681bf0a4.241c Call-ID: 7e1ee1647f15a5566ec2f4857f56660b@192.168.2.125 CSeq: 106 REGISTER WWW-Authenticate: Digest realm="sip1.xchangetele.com", nonce="44c0894155ef19a0f40546e7be36909fd1ef3df8", stale=true Server: Platform Experts BBT (Version 2.1a) Content-Length: 0 Warning: 392 69.1.236.33:5060 "Noisy feedback tells: pid=7913 req_src_ip=72.68.108.196 req_src_port=65415 in_uri=sip:sip.sipmedia.com out_uri=sip:sip.sipmedia.com via_cnt==1" --- (10 headers 0 lines)--- Responding to challenge, registration to domain/host name sip.sipmedia.com REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 69.1.236.33:5060: REGISTER sip:sip.sipmedia.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125:5060;branch=z9hG4bK79724411;rport From: ;tag=as7b899a4d To: Call-ID: 7e1ee1647f15a5566ec2f4857f56660b@192.168.2.125 CSeq: 107 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="7322761242", realm="sip1.xchangetele.com", algorithm=MD5, uri="sip:sip.sipmedia.com", nonce="44c0894155ef19a0f40546e7be36909fd1ef3df8", response="025eb7c28a8f59b95b65ae831e840ff9", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- <-- SIP read from 69.1.236.33:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.125:5060;branch=z9hG4bK79724411;rport=65415 From: ;tag=as7b899a4d To: ;tag=91791a7492f328483c7efb39681bf0a4.a64d Call-ID: 7e1ee1647f15a5566ec2f4857f56660b@192.168.2.125 CSeq: 107 REGISTER Contact: ;q=0.00;expires=120 Server: Platform Experts BBT (Version 2.1a) Content-Length: 0 Warning: 392 69.1.236.33:5060 "Noisy feedback tells: pid=7913 req_src_ip=72.68.108.196 req_src_port=65415 in_uri=sip:sip.sipmedia.com out_uri=sip:sip.sipmedia.com via_cnt==1" --- (10 headers 0 lines)--- Scheduling destruction of call '7e1ee1647f15a5566ec2f4857f56660b@192.168.2.125' in 32000 ms Jul 21 03:30:46 NOTICE[16111]: chan_sip.c:9828 handle_response_register: Outbound Registration: Expiry for sip.sipmedia.com is 120 sec (Scheduling reregistration in 105 s) <-- SIP read from 10.10.10.105:5060: NOTIFY sip:10.10.10.10 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK-5deabd7d From: Sara Hass ;tag=2512817a18f3124co0 To: Call-ID: d741e1ba-38712f0c@10.10.10.105 CSeq: 2320 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Sipura/SPA841-3.1.2(d) Content-Length: 0 --- (10 headers 0 lines)--- Transmitting (no NAT) to 10.10.10.105:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK-5deabd7d;received=10.10.10.105 From: Sara Hass ;tag=2512817a18f3124co0 To: ;tag=as362dc3f0 Call-ID: d741e1ba-38712f0c@10.10.10.105 CSeq: 2320 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call 'd741e1ba-38712f0c@10.10.10.105'