search for: cseq

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Here are the details: Successful call: INVITE cseq 1 From NexTone 100 Trying cseq 1 From Asterisk 100 Trying cseq 1 From Asterisk 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone INVITE (G711U) cseq 2 From NexTone 100 Trying cseq 2 From Asterisk 200 OK cseq 2 From Asterisk ACK cseq 2 From NexTone 200 OK (711U) cseq 1 From Ast...

...rio and digging deep into it. Nothing > immediately springs to mind. > After enabling pjsip specific debug log in asterisk, I can see, the following behavior: Incoming packages from network are always signaled like this (e.g.): sip_endpoint.c Processing incoming message: Request msg INVITE/cseq=10 (rdata0x7f5f1801a758) <--- Received SIP request (918 bytes) from UDP:195.185.37.60:5060 ... The path of the critical not existing package is like this and can't be seen elsewhere (there wasn't any corresponding incoming message entry before): sip_endpoint.c Distributing rdata to...

...help is appreciated. The specific error I see on the CLI is: Connected to Asterisk 13.6.0 currently running on ... (pid = 30046) *CLI> pjsip set logger on PJSIP Logging enabled [Feb 15 18:28:12] WARNING[30075]: pjsip:0 <?>: tsx0x7f14b0003 .Temporary failure in sending Request msg OPTIONS/cseq=52002 (tdta0x7f14b0001470), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT) [Feb 15 18:28:12] WARNING[30075]: pjsip:0 <?>: tsx0x7f14b0003 .Temporary failure in sending Request msg OPTIONS/cseq=52002 (tdta0x7f14b0001470), will try next server: Unsupported transport (PJSIP_E...

...P/99952880474@200.61.32.142") in new stack -- Called 99952880474@200.61.32.142 -- SIP/200.61.32.142-029e is ringing -- SIP/200.61.32.142-029e answered SIP/-081221b0 -- Attempting native bridge of SIP/-081221b0 and SIP/200.61.32.142-029e -- Got SIP response 481 "Invalid CSeq Number" back from 200.61.32.142 == Spawn extension (default, 2003, 1) exited non-zero on 'SIP/-081221b0' -- Got SIP response 481 "Invalid CSeq Number" back from 200.61.32.142 -- Got SIP response 481 "Invalid CSeq Number" back from 200.61.32.142 ==========...

...far.. Maybe one of you has an idea ? vectra*CLI> Sip read: > INVITE sip:0534280105@217.114.97.249;user=phone SIP/2.0 Via: SIP/2.0/UDP 130.89.224.240:5060 From: sip:ata1-1@217.114.97.249;tag=2733832243 To: <sip:0534280105@217.114.97.249;user=phone> Call-ID: 855024110@130.89.224.240 CSeq: 1 INVITE Contact: <sip:ata1-1@130.89.224.0:5060;transport=udp> User-Agent: Cisco ATA v2.15 ata18x (020927a) Expires: 300 Content-Length: 253 Content-Type: application/sdp v=0 o=ata1-1 33968 33968 IN IP4 130.89.224.240 s=ATA186 Call c=IN IP4 130.89.224.0 t=0 0 m=audio 16384 RTP/AVP 0 4 8 1...

...satisfying, but that is currently not my concern. The problem is, that most SIP phones or software (e.g. SJPhone) do resend the invite request, after approx 500 msec (measured by ethereal). chan_sip from asterisk seems to have a special handling for duplicate messages (i.e. messages with the same CSeq number) in order to ignore those messages in certain circumstances. Unfortunately asterisk doesn't ignore that second invite message by the client and sends an error message. (Though in the log files those "Too late messages" are mentioned.) Does anyone know a solution for the below...

On 06/14/2017 at 05:53 PM Joshua Colp wrote: > On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote: > > <snip> > >> >> I added this patch to see, if really all packages are are freed after >> they have been processed: >> >> --- b/res/res_pjsip/pjsip_distributor.c 2017-05-30 19:44:16.000000000 >> +0200 >> +++

...dpoints . switched from sipml5 to jssip with upgrade to 13.15.0(from 13.9.0) few days ago today i have problems with stopping/crashing asterisk /var/log/asterisk/messages dont show any clues [May 9 12:10:52] WARNING[25762] pjproject: tsx0x7fbb28024088 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7fbb080d29e8): Unknown Error (PJ_EUNKNOWN) [May 9 12:10:54] WARNING[6458] pjproject: tsx0x7fbb28024088 ..Error sending Response msg 200/REGISTER/cseq=4 (tdta0x7fbb382be318): Unknown Error (PJ_EUNKNOWN) [May 9 12:10:54] WARNING[20014] pjproject: tsx0x7fbb2c4a93e8 ..Error sending Respo...

...3 bytes): REGISTER sip:192.168.0.129 SIP/2.0 Via: SIP/2.0/UDP 192.168.102.70:5060;branch=z9hG4bK-wg4ok3zkt573;rport From: "snom_01" <sip:snom_01@192.168.0.129>;tag=i7u8p4i1vi To: "snom_01" <sip:snom_01@192.168.0.129> Call-ID: 3c267319f1b3-igmsa9072v8z@192-168-102-70 CSeq: 45683 REGISTER Max-Forwards: 70 Contact: <sip:snom_01@192.168.102.70:5060;line=v8ppcao5>;q=1.0 User-Agent: snom190-3.56i P-NAT-Refresh: 15 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.102.70 WWW-Contact: <http://192.168.102.70:80> WWW-Contact: <https://192.168.102.70:443&...

...- SIP read from 10.1.0.65:5060 ---> INVITE sip:8500 at 10.1.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKjxcxrrks Max-Forwards: 70 To: <sip:8500 at 10.1.0.10> From: "Hector" <sip:201 at 10.1.0.10>;tag=uucwz Call-ID: mrsyiysrdkwmkeg at defiant.freesoftware.org CSeq: 183 INVITE Contact: <sip:201 at 10.1.0.65> Content-Type: application/sdp Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE Supported: replaces,norefersub,100rel User-Agent: Twinkle/1.2 Content-Length: 247 v=0 o=twinkle 2122879389 441437466 IN IP4 10.1.0.65 s=- c...

...REGISTER sip:voip.livewirenet.com SIP/2.0 Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682215623 Max-Forwards: 70 From: <sip:3036284311@voip.livewirenet.com;user=phone>;tag=1c1682209279 To: <sip:3036284311@voip.livewirenet.com;user=phone> Call-ID: 1494991476221200001530@66.185.98.152 CSeq: 11 REGISTER Contact: <sip:3036284311@66.185.98.152;user=phone>;expires=86400 Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 86400 User-Agent: Audiocodes-Sip-Gateway-MP-108 FXS/v.4.40.227.419 Content-Length: 0...

...R sip:10.2.7.2 SIP/2.0 Via: SIP/2.0/UDP 10.2.7.254:5060;branch=z9hG4bK1029a1dd From: <sip:6096@10.2.7.2>;tag=0017e0134094000d045a4ac2-6d217bab To: <sip:6096@10.2.7.2> Call-ID: 0017e013-40940006-1e1b45e9-2aa6b3d9@10.2.7.254 Max-Forwards: 70 CSeq: 104 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: <sip:6096@10.2.7.254:5060;user=phone;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0017e0134094>";+u.sip!model.ccm.cisco.com="7" Content-Length: 0 Expires: 120...

...oy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. -------------- next part -------------- SIP Debugging Enabled Sip read: REGISTER sip:pbx SIP/2.0 Via: SIP/2.0/UDP 192.168.144.247 CSeq: 299 REGISTER To: "Roy Sigurd Karlsbakk" <sip:roy@pbx> Expires: 900 From: "Roy Sigurd Karlsbakk" <sip:roy@pbx> Call-ID: 1793030308@192.168.144.247 Content-Length: 0 User-Agent: KPhone/3.1 Event: registration Allow-Events: presence Contact: "Roy Sigurd Karlsbakk&...

...39; We're at 192.168.0.203 port 17456 ------------------------------------------------ Sip read: INVITE sip:3218888@192.168.0.203 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.153;branch=z9hG4bK2038176231 From:<sip:2000@192.168.0.203>; To: <sip:3218888@192.168.0.203> Call-ID: 52@192.168.0.153 CSeq: 21 INVITE Contact: <sip:2000@192.168.0.153> Max-Forwards: 5 User-Agent:SKYPHONE/1.03 Subject: hello Expires: 120 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER,SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length:180 Proxy-Authorization: Digest username="2000",r...

Hi all, Asterisk is great but I'm having issues with setting up realtime for our call center, which is needed for login integration with the rest of our applications (telephonists' web interface, etc.). I have reviewed a large number of previous posts to the mailing list and the voip-info wiki to no avail. Setup is as follows: Linux 2.6.23 (gentoo) / AMD Athlon(tm) 64 Processor 3000+ /

...sip:8719@192.168.1.15;user=ip SIP/2.0 Via: SIP/2.0/UDP 192.168.1.84:5060 From: "5285" <sip:5285@192.168.1.15>;tag=000d287e269a000f5181f06d-45b64a55 To: <sip:8719@192.168.1.15;user=ip> Call-ID: 000d287e-269a0014-009bca0e-0f8ef73a@192.168.1.84 Date: Thu, 23 Oct 2003 21:23:19 GMT CSeq: 101 INVITE User-Agent: CSCO/5 Contact: <sip:5285@192.168.1.84:5060> Expires: 180 Content-Type: application/sdp Content-Length: 246 Accept: application/sdp Remote-Party-ID: "5285" <sip:5285@192.168.1.84>;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA...

....202:5060;branch=z9hG4bK-0-35c-47a0 Max-Forwards: 70 Supported: replaces User-Agent: FXS_GW (4asipfxs.107a) Contact: <sip:1234@192.168.0.202:5060>;expires=60 From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-0-35c-48a3 To: <sip:1234@192.168.0.200> Call-ID: c0a800ca-13c4-0-334-1c34 CSeq: 1 REGISTER Content-Length:0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.202 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-0-35c-47a0 From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-0-35c-48...

...060;branch=z9hG4bK5a5cde5e From: "21382890" <sip:21382890@217.168.168.5>;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...

...: OPTIONS sip:192.168.1.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK17299c89;rport From: "asterisk" <sip:asterisk@192.168.1.10>;tag=as1ab4b0c6 To: <sip:192.168.1.4> Contact: <sip:asterisk@192.168.1.10> Call-ID: 2285e5551ca492cf6d3f6a8f52949df7@192.168.1.10 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 17 Feb 2006 07:13:32 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Retransmitting #1 (no NAT) to 192.168.1.4:5060: OPTIONS sip:192.168.1.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.10:5060;br...

...0.227.109.232 SIP/2.0 Via: SIP/2.0/UDP 211.129.117.89:5060;branch=z9hG4bK638c731a From: "050xxxxxxx" <sip:anonymous@localhost>;tag=as0d659f34 To: <sip:090xxxxxxx3@210.227.109.232> Contact: <sip:anonymous@211.129.117.89> Call-ID: 3cea8d85269ea4d0269c629a2720af7c@ocn.ne.jp CSeq: 102 INVITE User-Agent: Fletsphone/2.3 (VOIP_AD 3.00; NTTEAST/NTTWEST) Max-Forwards: 70 Proxy-Require: privacy Remote-Party-ID: "050xxxxxxx" <sip:050xxxxxxx@ocn.ne.jp>;privacy=full;screen=pass Date: Tue, 14 Nov 2006 07:12:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSC...