search for: d87543

...sip dialog xxxx". What to do for this problem??? I had enabled the sip debug at the asterisk. I have pasted the messages, I got below. Please help me in solving the problem. Thanks in advance, Treesa REGISTER sip:192.168.12.160 SIP/2.0 Via: SIP/2.0/UDP 192.168.25.116:52166;branch=z9hG4bK-d87543-b65c86230550ff4f- 1-- d87543-;rport Max-Forwards: 70 Contact: <sip:1002 at 192.168.25.116:52166;rinstance=1a12ef13351e0ee1> To: "1002"<sip:1002 at 192.168.12.160> From: "1002"<sip:1002 at 192.168.12.160>;tag=5f799517 Call-ID: Yjk5ZTFjZjI1Mzc5N2RiMWFhYWRi...

...re is no "asterisk" in the from header except the option message. I wonder why "asterisk" will be shown in the receiver end's screen. ango U 10.0.0.25:2750 -> 10.201.0.224:5060 INVITE sip:85236418505@10.201.0.224 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.25:2750;branch=z9hG4bK-d87543-5d65ca22ac139c29-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:9000220002@10.0.0.25:2750>. To: "85236418505"<sip:85236418505@10.201.0.224>. From: "angry boy"<sip:9000220002@10.201.0.224>;tag=b842555d. Call-ID: MWE0YWUzYzBmOTQ2NGUyYTE2OTAxYTRlYTk4ODAxOTA.. C...

...see the ReInvite Event in the cli , some thing like the attached logs, is there any available patch to note down the "On Hold" and "Off Hold" event in log file or database? CLI Screen Logs: INVITE sip:2@192.168.0.20 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.199:24608;branch=z9hG4bK-d87543-2e63d915643a940b-1--d87543-;rport Max-Forwards: 70 Contact: <sip:1015@192.168.0.199:24608> To: "2"<sip:2@192.168.0.20>;tag=as2377f10b From: "1015"<sip:1015@192.168.0.20>;tag=bf1d1102 Call-ID: MTBkODdlYjE4NWE1NTlkOTc3MzViZjY5MWU5MGZhNTA. CSeq: 3 INVITE Allow: IN...

...he Contact header not being present although no message is coming up on the console to that effect. Regards, Greyman. INVITE sip:1234 at server SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKbfe545ae83ef7d0ec9fe44ea063d72c67f4bc926 Via: SIP/2.0/UDP 192.168.1.102:4110;rport=10260;branch=z9hG4bK-d87543-302f8c6313727f46-1--d87543- To: <sip:1234 at server> From: <sip:user at server>;tag=6e2df459 Call-ID: ZTFmYjU1OWVhZDFlOTMxN2NlM2NhNzdlYzBmNjZiNWI. CSeq: 2 INVITE Contact: <sip:user at 192.168.1.102> Max-Forwards: 69 Record-Route: <sip:10.0.0.1;lr> User-Agent: Bria release 2....

...------------------------------------- ------------ pbx*CLI> sip debug SIP Debugging Enabled pbx*CLI> Sip read: INVITE sip:232@pbx.ocg.ca SIP/2.0 To: <sip:232@pbx.ocg.ca> From: pbx.ocg.ca<sip:233@pbx.ocg.ca>;tag=620dc660 Via: SIP/2.0/UDP 172.31.254.106:9330;branch=z9hG4bK-d87543-22694588-1--d87543-;rport Call-ID: 113d5508a72b5176 CSeq: 1 INVITE Contact: <sip:233@172.31.254.106:9330> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3004t stamp 16741 Content-Leng...

...ort=udp> SIP/2.0 To: <sip:4415 at x.x.x.x5060;transport=udp<sip:4415 at s144701.trixbox.fonality.com:5060;transport=udp> > From: <sip:000000003992 at x.x.x.x:5060<http://sip:000000003992 at 10.247.1.126:5060> >;tag=4f5cb549 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK-d87543-62412f606c62824d-1--d87543-;rport Call-ID: 350da2493d160e6f at ZHQtZGVsbDN2ZHIwZjEuQU0uSURUQ09SUC5ORVQ. CSeq: 1 INVITE Contact: <sip:000000003992 at x.x.x.x:5060<http://sip:000000003992 at 10.247.1.126:5060> > Max-Forwards: 70 Session-Expires: 1800 Min-SE: 90 Accept-Language: en...

...that i switch over to using the domain name, X-Lite can't log in. =========With Domain Name (doesnt work)============ Transmitting (NAT) to 85.250.206.46:6007 <http://85.250.206.46:6007>: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 85.250.179.93 <http://85.250.179.93> ;branch=z9hG4bK-d87543-895382184-1--d87543-;received=85.250.206.46<http://85.250.206.46>;rport=6007 From: Dan<sip:110010@sip1.sippal.com>;tag=42134245 To: Dan<sip:110010@sip1.sippal.com>;tag=as67377eb5 Call-ID: 184e0832a904b418 CSeq: 1 REGISTER User-Agent: SipPal.com PBX Allow: INVITE, ACK, CANCEL, OPT...

...rname=001234 secret=Xxxx host=host.net insecure=very canreinvite=no ASTERISK CONSOLE LOG 09:49:44.2 REGISTER sip:host.net SIP/2.0 To: new test<sip:001234@host.net> From: new test<sip:001234@host.net>;tag=6b1a582b Via: SIP/2.0/UDP 192.168.2.2:8324;branch=z9hG4bK-d87543-741421617-1--d87543-;rport Call-ID: 29692c204a75e445 CSeq: 6 REGISTER Contact: <sip:001234@192.168.2.2:8324> Expires: 3600 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE User-Agent: eyeBeam release 3003x stamp 16296 (sn:0ea18597a34ba5ef8f48...

...status> <contact priority="1.00">sip:1001 at example.com</contact> <note xml:lang="en">Available</note> </tuple> </presence> ## ASTERISK -> PHONE ## SIP/2.0 489 Bad Event To: "1001"<sip:1001 at example.com >;tag=z9hG4bK-d87543-ff0f99061907dd6eaa3b-1--d87543- From: "1001"<sip:1001 at example.com>;tag=98slbhbn16 Via: SIP/2.0/UDP 172.31.19.250:2048;branch=z9hG4bK-w2orn21sre9u Call-ID: 54ddf28f87c7-ak5cx3jtjc8c CSeq: 12480 PUBLISH Server: Asterisk PBX 13.2.0 Content-Length: 0 And the debug logging is report...

...25162fa29794", response="98e4d21ca0f75497d7fb12a8a4914bcb", opaque="5adc3dd2" Expires: 3600 Contact: <sip:{registration_user}@{asterisk_ip}> Content-Length: 0 Expected registration: REGISTER sip:broadsmart.net SIP/2.0 Via: SIP/2.0/UDP 208.73.25.70:5060;branch=z9hG4bK-d87543-826b1b62b62ac91d-1--d87543-;rport Record-Route: <sip:2135997816 at 208.73.25.70;lr> From: "2135997816" <sip:2135997816 at broadsmart.net>;tag=e944c233 To: "2135997816" <sip:2135997816 at broadsmart.net> Call-ID: e0576109f9699a4d at dGFjd3MxLmludC5uYXRlbGNvbW0uY...

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 ? --------------

...G-----16-----ANSWER-----48-----45) completed, returning 0 vicidialnow*CLI> sip debug shows below lines: *Quote:* --- (12 headers 0 lines) --- Sending to 192.168.0.50 : 12714 (NAT) Transmitting (NAT) to 192.168.0.50:12714: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.50:12714 ;branch=z9hG4bK-d87543-930550325154e53d-1--d87543-;received=192.168.0.50;rport=12714 From: "cc106"<sip:cc106 at 192.168.0.2 <sip%3Acc106 at 192.168.0.2>>;tag=7f1cff22 To: "817275691533"<sip:817275691533 at 192.168.0.2<sip%3A817275691533 at 192.168.0.2> >;tag=as02559696 Call-ID:...