search for: set_destination

Am having a unique problem, calls received on my SPA942 seem to end after 15 seconds, but calls made from this device do not have this problem. For this device (when receiving calls) I get periodic "chan_sip.c set_destination: can't find address for host" I have set the "canreinvite=no" in the sip.conf. Does anyone have a sample entry from sip.conf for the Lynksys SPA 942 to share with me. Allan. ____________________________________________________________________________________ Yahoo! Music Unl...

....5>;tag=as6556b0d9 To: <sip:723@216.52.153.207>;tag=26845C24-FDA Date: Tue, 08 Jul 2003 22:22:57 GMT Call-ID: 14bce0f47fb42b734f7904ca351a4220@217.168.168.5 Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 CSeq: 117 INFO Contact: <sip:723@216.52.153.207:5060> 10 headers, 0 lines set_destination: Parsing <sip:723@216.52.153.207:5060> for address/port to send to set_destination: set destination to 216.52.153.207, port 5060 Reliably Transmitting: INFO sip:723@216.52.153.207 SIP/2.0 Via: SIP/2.0/UDP 217.168.168.5:5060;branch=z9hG4bK5a5cde5e From: "21382890" <sip:21382890@21...

...tag=as7b9f4bfb' is not a valid SIP contact (missing sip:) trying to use anyway [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:8097 set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : '"72.16.1.20>' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '"72.16.1.20' [Jul 19 10:46:41] WARNING[2466]: chan_sip.c:5839 set_destination: Can't find address for host '"72.16.1.20' or [Jul 19 10:45:03] NOTICE[2466]: chan_sip.c:8068 set_address_from_contact: '"172.16.1.20>;tag=a...

...Found audio format UNKN Found audio format UNKN Found video format UNKN Found video format UNKN Found video format UNKN Capabilities: us - 786436, them - 303/851968, combined - 786436 Non-codec capabilities: us - 1, them - 0, combined - 0 list_route: hop: <sip:192.168.1.38:5060;transport=udp> set_destination: Parsing <sip:192.168.1.38:5060;transport=udp> for address/port to send to set_destination: set destination to 192.168.1.38, port 5060 Transmitting: ACK sip:192.168.1.38:5060 SIP/2.0 Via: SIP/2.0/UDP <asterisk ip>:5060;branch=z9hG4bK0ee1ee6b From: "3002" <sip:3002@<aster...

...-8a62-5bacd517e1ae To: <sip:1361@asterisk;user=phone>;tag=as05865310 Call-ID: 860136be-1ae5-44db-b86d-90b5d31f0c08@192.168.3.54 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:1361@192.168.3.6> Content-Length: 0 to 192.168.3.54:14250 set_destination: Parsing <sip:123@192.168.3.52:5060;user=phone;transport=udp> for address/port to send to set_destination: set destination to 192.168.3.52, port 5060 We're at 192.168.3.6 port 29312 Answering with preferred capability 4 Answering with non-codec capability 1 11 headers, 10 lines Reliably T...

...ontent-Type: application/sdp Content-Length: 240 v=0 o=root 1431 1433 IN IP4 192.168.96.16 s=session c=IN IP4 192.168.96.16 t=0 0 m=audio 27002 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> set_destination: Parsing <sip:302 at 192.168.96.16:5060;user=phone;transport=udp> for address/port to send to set_destination: set destination to 192.168.96.16, port 5060 Audio is at 192.168.96.5 port 16816 Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.96.16:5060: IN...

...b10eea268512db62e41181dc3d@61.95.68.84 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:2001@61.95.68.84> Content-Length: 0 to 213.137.73.146:5060 == Spawn extension (mpa-phones, 861892142222, 2) exited non-zero on 'SIP/2001-40ae' set_destination: Parsing <sip:2001@211.28.211.217:5060> for address/port to send to set_destination: set destination to 211.28.211.217, port 5060 Reliably Transmitting: BYE sip:2001@211.28.211.217:5060 SIP/2.0 Via: SIP/2.0/UDP 61.95.68.84:5060;branch=z9hG4bK097c0033 From: <sip:861892142222@61.95.68.84>...

...45.199:5260 ? [Khfemsrv*CLI> [Apr 19 14:26:03] WARNING[18442]: chan_sip.c:7724 set_address_from_contact: ?Invalid host name in Contact: (can't resolve in DNS) : '7113' ?list_route: hop: <sip:7113;phone-context=cdp.udp@qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone> ?set_destination: Parsing <sip:7113;phone-context=cdp.udp@qg.com:5060;maddr=192.168.45.129;transport=udp;user=phone> for address/port to send to ?set_destination: set destination to 192.168.45.129, port 5060 ?Transmitting (no NAT) to 192.168.45.129:5060: ACK sip:7113;phone-context=cdp.udp@qg.com:5060;maddr=19...

...d AVP profile in audio answer but AVPF is enabled: audio 7200 RTP/AVP 8 101 [Aug 20 09:19:57] WARNING[7080][C-0000011f]: chan_sip.c:10819 process_sdp: Failing due to no acceptable offer found I enabled debug on the IP of the dect-phone (full log attached), but it does not make me any wiser... set_destination: Parsing <sip:dect at 192.168.0.27:5060> for address/port to send to set_destination: set destination to 192.168.0.27:5060 Reliably Transmitting (no NAT) to 192.168.0.27:5060: BYE sip:dect at 192.168.0.27:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK239cc5d8 Max-Forwards: 70...

...p:0 PCMU/8000 10 headers, 8 lines Found audio format ALAW Found audio format UNKN Found description format PCMA Found description format PCMU Capabilities: us - 12, them - 12/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 list_route: hop: <sip:304@192.168.3.211:5060> set_destination: Parsing <sip:304@192.168.3.211:5060> for address/port to send to set_destination: set destination to 192.168.3.211, port 5060 Transmitting: ACK sip:304@192.168.3.211:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4e82ae48 From: "0" <sip:0@192.168.3.6>;tag=as1dbb...

...er) : /[Jul 31 11:56:58] NOTICE[32273]: chan_sip.c:26194 sip_poke_noanswer: Peer 'testacc77000' is now UNREACHABLE! Last qualify: 49// //[Jul 31 11:56:58] Really destroying SIP dialog '78b0d1701d3694b1494a0c4b55344d57 at ip-sip-server:5060' Method: OPTIONS// //[Jul 31 11:56:58] set_destination: Parsing <sip:testacc77003 at 192.168.1.109:1024> for address/port to send to// //[Jul 31 11:56:58] set_destination: set destination to 192.168.1.109:1024// //[Jul 31 11:56:58] Reliably Transmitting (NAT) to my-public-ip:1024:// //NOTIFY sip:testacc77003 at 192.168.1.109:1024 SIP/2.0// //Via...

...er - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.12:55246 list_route: hop: <sip:adf at 116.18.35.235:28614> [Jul 21 11:10:27] DEBUG[9707]: chan_sip.c:5695 reqprep: Strict routing enforced for session 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x set_destination: Parsing <sip:adf at 116.18.35.235:28614> for address/port to send to set_destination: set destination to 116.18.35.235, port 28614 Transmitting (NAT) to 116.18.35.235:28614: ACK sip:adf at 116.18.35.235:28614 SIP/2.0 Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK07eb06b5;rport From: "pe...

....5>;tag=as26b5df58 Call-ID: 67d8e3801b04410659f8ea1b635b6db6 at 201.73.67.5 CSeq: 15651 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:3130296800 at 201.73.67.5> Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- set_destination: Parsing <sip:201.73.67.7:5080> for address/port to send to set_destination: set destination to 201.73.67.7, port 5080 Reliably Transmitting (no NAT) to 201.73.67.7:5080: NOTIFY sip:201.73.67.7:5080 SIP/2.0 Via: SIP/2.0/UDP 201.73.67.5:5060;branch=z9hG4bK26db8c59;rport From: "3130296800&...

...rmat GSM Found description format PCMU Found description format telephone-event Capabilities: us - 6, them - 6/0, combined - 6 Non-codec capabilities: us - 1, them - 1, combined - 1 list_route: hop: <sip:612@192.246.69.223;ftag=as1f0e4544;lr=on> list_route: hop: <sip:30342@65.121.72.14> set_destination: Parsing <sip:612@192.246.69.223;ftag=as1f0e4544;lr=on> for addr ess/port to send to set_destination: set destination to 192.246.69.223, port 5060 Transmitting: ACK sip:30342@65.121.72.14 SIP/2.0 Via: SIP/2.0/UDP 82.38.193.149:5060;branch=z9hG4bK42c80372 Route: <sip:30342@65.121.72.14>...

...51:5060;branch=z9hG4bK41bcdf63;rport From: "Steffen" <sip:116 at 192.168.150.151>;tag=as085e9ac6 To: <sip:119 at 192.168.150.11:5060> Call-ID: 17eb890c14a50dfa541efbb517835bb4 at 192.168.150.151 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing <sip:116 at 192.168.150.51:5060;transport=udp> for address/port to send to set_destination: set destination to 192.168.150.51, port 5060 We're at 192.168.150.151 port 16216 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-...

...-- Called 1021 -- Started music on hold, class 'default', on Zap/1-1 -- SIP/1021-eaad is ringing -- SIP/1011-4c98 answered Zap/1-1 ;another phone picked up pressing 888 -- Stopped music on hold on Zap/1-1 pbx*CLI> sip debug ; i enabled debug here SIP Debugging Enabled set_destination: Parsing <sip:1011@192.168.123.60:5060> for address/port to send to set_destination: set destination to 192.168.123.60, port 5060 Reliably Transmitting: BYE sip:1011@192.168.123.60:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.123.50:5060;branch=z9hG4bK01879857 From: <sip:888@192.168.123.50>;ta...

...N IP4 192.168.4.204 t=0 0 m=audio 2236 RTP/AVP 0 a=rtpmap:0 PCMU/8000 11 headers, 7 lines Found audio format UNKN Found description format PCMU Capabilities: us - 4, them - 4/0, combined - 4 Non-codec capabilities: us - 1, them - 0, combined - 0 list_route: hop: <sip:3004@192.168.4.204:5060> set_destination: Parsing <sip:3004@192.168.4.204:5060> for address/port to send to set_destination: set destination to 192.168.4.204, port 5060 Transmitting: ACK sip:3004@192.168.4.204:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf From: "3124048745" <sip:3124048745@192.1...

...To: "CCUK" <sip:01618313800@194.24.251.3>;tag=as0eb5392e Call-ID: 27584e3a339e535209ea89102043184e@194.24.251.3 CSeq: 1 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:01618313800@194.24.251.3> Content-Length: 0 to 10.0.0.82:5060 set_destination: Parsing <sip:1301@10.0.0.82:5060> for address/port to send to set_destination: set destination to 10.0.0.82, port 5060 Reliably Transmitting: NOTIFY sip:1301@10.0.0.82:5060 SIP/2.0 Via: SIP/2.0/UDP 194.24.251.3:5060;branch=z9hG4bK08ef2ac1 From: "CCUK" <sip:01618313800@194.24.251...

IVR Number :17275691533 When I try it from xlite configuring my provider directly, it works perfectly. When I try to dial out from dialer , it doesnt work. [sip8] type=peer username=user fromuser=user authuser=user secret=password host=8.14.146.111 nat=no canreinvite=yes insecure=very disallow=all allow=g729 allow=ulaw context=default dtmfmode=rfc2833 What cld be the reason ? --------------

...8.242:5060;branch=z9hG4bK2142c11da4177 From: <sip:5555832351@sipproxy.magenta.cl>;tag=2142c11da4 To: <sip:005622408196@sipproxy.magenta.cl>;tag=as1be17fe7 Call-ID: 21fb7142-05e9-c19e-821d-0002a400f1e9@xxx.xxx.148.242 CSeq: 177 ACK Content-Length: 0 Max-Forwards: 69 10 headers, 0 lines set_destination: Parsing <sip:005622408196@xxx.xxx.148.246;ftag=2142c11da4;lr=on> for address/port to send to set_destination: set destination to xxx.xxx.148.246, port 5060 We're at xxx.xxx.148.232 port 19848 Answering with capability 0x4(ULAW) Answering with non-codec capability 0x1(G723) 12 headers, 10...