*CLI> sip debug SIP Debugging Enabled -- Starting simple switch on 'Zap/1-1' -- Executing Dial("Zap/1-1", "Zap/g2/") in new stack -- Called g2/ -- Zap/3-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/3-1 -- Starting simple switch on 'Zap/4-1' -- Executing Wait("Zap/4-1", "1") in new stack -- Executing GotoIf("Zap/4-1", "0?4") in new stack -- Executing SetCIDNum("Zap/4-1", "916503588567") in new stack -- Executing Dial("Zap/4-1", "Zap/1&SIP/wilson-polycom|20|tT") in new stack -- Called 1 We're at 10.1.1.48 port 17184 Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 12 lines Reliably Transmitting (no NAT) to 10.1.1.200:5060: INVITE sip:wilson-polycom@10.1.1.200:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.48:5060;branch=z9hG4bK45cc38d1 From: "CHAN WILSON " ;tag=as3369d29d To: Contact: Call-ID: 08d228fb3456d15f055695c021087f3f@10.1.1.48 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Sat, 19 Mar 2005 17:50:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 257 v=0 o=root 17417 17417 IN IP4 10.1.1.48 s=session c=IN IP4 10.1.1.48 t=0 0 m=audio 17184 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 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 wilson-polycom -- Zap/1-2 is ringing <-- SIP read from 10.1.1.200:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.1.48:5060;branch=z9hG4bK45cc38d1 From: "CHAN WILSON" ;tag=as3369d29d To: ;tag=F5E24555-AC33A322 CSeq: 102 INVITE Call-ID: 08d228fb3456d15f055695c021087f3f@10.1.1.48 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Content-Length: 0 --- (9 headers 0 lines)--- <-- SIP read from 10.1.1.200:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.1.1.48:5060;branch=z9hG4bK45cc38d1 From: "CHAN WILSON" ;tag=as3369d29d To: ;tag=F5E24555-AC33A322 CSeq: 102 INVITE Call-ID: 08d228fb3456d15f055695c021087f3f@10.1.1.48 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Allow-Events: talk,hold,conference Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/wilson-polycom-a2d8 is ringing -- CPE supports Call Waiting Caller*ID. Sending 'CHAN WILSON /916503588567' <-- SIP read from 10.1.1.200:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.48:5060;branch=z9hG4bK45cc38d1 From: "CHAN WILSON" ;tag=as3369d29d To: ;tag=F5E24555-AC33A322 CSeq: 102 INVITE Call-ID: 08d228fb3456d15f055695c021087f3f@10.1.1.48 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Content-Type: application/sdp Content-Length: 183 v=0 o=- 1111254617 1111254617 IN IP4 10.1.1.200 s=Polycom IP Phone c=IN IP4 10.1.1.200 t=0 0 m=audio 2242 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 10.1.1.200:2242 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) 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 10.1.1.200, port 5060 Transmitting (no NAT) to 10.1.1.200:5060: ACK sip:wilson-polycom@10.1.1.200:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.48:5060;branch=z9hG4bK1a1f5dd6 From: "CHAN WILSON " ;tag=as3369d29d To: ;tag=F5E24555-AC33A322 Contact: Call-ID: 08d228fb3456d15f055695c021087f3f@10.1.1.48 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/wilson-polycom-a2d8 answered Zap/4-1 -- Hungup 'Zap/1-2' <-- SIP read from 10.1.1.200:5060: BYE sip:916503588567@10.1.1.48 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.200:5060;branch=z9hG4bK62ac983839973ECB From: ;tag=F5E24555-AC33A322 To: "CHAN WILSON" ;tag=as3369d29d CSeq: 1 BYE Call-ID: 08d228fb3456d15f055695c021087f3f@10.1.1.48 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines)--- <-- SIP read from 10.1.1.200:5060: BYE sip:916503588567@10.1.1.48 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.200:5060;branch=z9hG4bK62ac983839973ECB From: ;tag=F5E24555-AC33A322 To: "CHAN WILSON" ;tag=as3369d29d CSeq: 1 BYE Call-ID: 08d228fb3456d15f055695c021087f3f@10.1.1.48 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines)--- <-- SIP read from 10.1.1.200:5060: BYE sip:916503588567@10.1.1.48 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.200:5060;branch=z9hG4bK62ac983839973ECB From: ;tag=F5E24555-AC33A322 To: "CHAN WILSON" ;tag=as3369d29d CSeq: 1 BYE Call-ID: 08d228fb3456d15f055695c021087f3f@10.1.1.48 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines)--- <-- SIP read from 10.1.1.200:5060: BYE sip:916503588567@10.1.1.48 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.200:5060;branch=z9hG4bK62ac983839973ECB From: ;tag=F5E24555-AC33A322 To: "CHAN WILSON" ;tag=as3369d29d CSeq: 1 BYE Call-ID: 08d228fb3456d15f055695c021087f3f@10.1.1.48 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines)--- <-- SIP read from 10.1.1.200:5060: BYE sip:916503588567@10.1.1.48 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.200:5060;branch=z9hG4bK62ac983839973ECB From: ;tag=F5E24555-AC33A322 To: "CHAN WILSON" ;tag=as3369d29d CSeq: 1 BYE Call-ID: 08d228fb3456d15f055695c021087f3f@10.1.1.48 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines)--- -- Hungup 'Zap/3-1' == Spawn extension (default, 9, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' set_destination: Parsing for address/port to send to set_destination: set destination to 10.1.1.200, port 5060 Reliably Transmitting (no NAT) to 10.1.1.200:5060: BYE sip:wilson-polycom@10.1.1.200:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.48:5060;branch=z9hG4bK17ae43df From: "CHAN WILSON " ;tag=as3369d29d To: ;tag=F5E24555-AC33A322 Contact: Call-ID: 08d228fb3456d15f055695c021087f3f@10.1.1.48 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 --- == Spawn extension (home-incoming, s, 4) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' <-- SIP read from 10.1.1.200:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 10.1.1.48:5060;branch=z9hG4bK17ae43df From: "CHAN WILSON" ;tag=as3369d29d To: ;tag=F5E24555-AC33A322 CSeq: 103 BYE Call-ID: 08d228fb3456d15f055695c021087f3f@10.1.1.48 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Content-Length: 0 --- (9 headers 0 lines)--- -- Got SIP response 500 "Internal Server Error" back from 10.1.1.200 Destroying call '08d228fb3456d15f055695c021087f3f@10.1.1.48' <-- SIP read from 10.1.1.200:5060: BYE sip:916503588567@10.1.1.48 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.200:5060;branch=z9hG4bK62ac983839973ECB From: ;tag=F5E24555-AC33A322 To: "CHAN WILSON" ;tag=as3369d29d CSeq: 1 BYE Call-ID: 08d228fb3456d15f055695c021087f3f@10.1.1.48 Contact: User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.4.1 Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines)--- Sending to 10.1.1.200 : 5060 (non-NAT) Transmitting (no NAT) to 10.1.1.200:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.200:5060;branch=z9hG4bK62ac983839973ECB From: ;tag=F5E24555-AC33A322 To: "CHAN WILSON" ;tag=as3369d29d Call-ID: 08d228fb3456d15f055695c021087f3f@10.1.1.48 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0