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...:1},20,r)) I tried a echo test, dialing in my case to 8500, but in spite of seeing traffic towards Internet, nothing is heard. Setting 'sip set debug', I get the following thing: <--- 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,C...

...sanitized sip trace from the snom phone for your perusal. Thanks for any help you can offer. Brian ### START SIP TRACE ### Sent to udp:192.168.0.129:5060 at 14/12/2004 18:21:29:500 (593 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-Agen...

...SIP messages to Softphone. 2. Softphone receives SIP messages and replys back. 3. Asterisk doesn't receive these replies and keeps on sending. Asterisk: Reliably Transmitting (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;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 A...

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+ /

I'm running Asterisk 10.7.0 with three sip trunks to my call termination provider. For the most part everything works great. However, at apparently random times and usually about 20 mins or so into the call, the outbound audio stream dies. The call stays connected and the inbound audio works fine. The thing is, it happens on such an irregular basis (once or twice per day) that I can't get

...y between 2 sip clients, no need to go through asterisk server here is my debug log: <--- SIP read from UDP://192.168.1.4:18341 ---> INVITE sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5> SIP/2.0 Via: SIP/2.0/UDP 192.168.1.4:18341 ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:test at 192.168.1.4:18341> To: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> >;tag=f543a140 Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2...

...ny help you can offer. >> >>Brian >> >>### START SIP TRACE ### >> >>Sent to udp:192.168.0.129:5060 at 14/12/2004 18:21:29:500 (593 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@192.1...

...e 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.. CSeq: 1 INVITE. Allow: INVITE, ACK,...

Dan Cropp wrote: > I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)? > > PJSIP is including the Contact for the ACK response to the OK. > Contact:<sip:1234 at xxx.xxx.xx.xxx:5060> > There is no configuration option to configure this behavior. What is the full SIP signaling? -- Joshua

...e it doesn't come in on port 5060, where it actually originated on the VM, but on a random port that the VM hosting providers' NAT router rewrote to, in the below case port 63664. And the remote SIP provider tries to send the reply back on that random port. Note MY.PUBLIC.IP.ADDRESS and rport below: IP my.provider.com.5060 > 192.168.1.2.63664: UDP, length 534 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK25a36d76;received=MY.PUBLIC.IP.ADDRESS;rport=63664 From: <sip:1234 at my.provider.com>;tag=as762d7322 To: <sip:1234 at my.provider.com>;tag=a...

...****************************************** <-- SIP read from 203.88.192.42:5160: INVITE sip:84104214@70.84.200.204 SIP/2.0 Record-Route: <sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on> Via: SIP/2.0/UDP 203.88.192.42:5160;branch=z9hG4bK5aeb.8f7e6027.0 Via: SIP/2.0/UDP 211.147.240.237:5060;rport=57786 From: <sip:0017911@211.147.240.237>;tag=1CA65AC-9C8 To: <sip:84104214@203.88.192.42> Date: Thu, 29 Sep 2005 20:14:40 GMT Call-ID: 805AF00B-305C11DA-814CFCF5-33432EF@211.147.240.237 Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 2153363387-811340250-2169109749-53752559 User-Agen...

...that, I will include the trace for the pjsip.conf which it seems Vitelity isn't accepting the ACK in response to the OK ---- SIP --- <--- Transmitting SIP request (1004 bytes) to UDP:64.2.142.93:5060 ---> INVITE sip:8005555555 at 64.2.142.93 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a From: "Dan" <sip:291 at XXX.XXX.XXX.XXX>;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8 To: <sip:8005555555 at 64.2.142.93> Contact: <sip:15062fef-986e-4fcf-a93e-06b28da02fff at XXX.XXX.XXX.XXX:5060> Call-ID: 309ec892-56a8-4...

...are the register auth exchange to the invite auth exchange and see if anything differs. > --------------------- > > Transmitting SIP request (885 bytes) to UDP:65.254.44.194:5060 ---> > INVITE sip:12025551212 at 65.254.44.194:5060 SIP/2.0 > Via: SIP/2.0/UDP > 18.18.19.123:5060;rport;branch=z9hG4bKPj70617ad5-d57a-4e5b-8043-086b0b8ebba2 > From: <sip:sonny at 192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8 > To: <sip:12025551212 at 65.254.44.194> > Contact: <sip:dea9e47d-3a06-4e6e-b88f-8bac70fb6e0b at 18.18.19.123:5060> > Call-ID: 012135e9-b...

Hi, I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42 (I can't upgrade to 1.8.x at the moment for various reasons). I've noticed for user agents that have a VIA header with a different port than the port the NOTIFY was sent from, the NOTIFY reply will always be sent back to that port, which is incorrect. (Sonicwalls and other routers love to do this, even

...:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:112 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <--- Transmitting SIP response (733 bytes) to UDP:217.10.79.9:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0 Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4b...

Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the

...lr;phase=terminating> Supported: replaces, timer Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Date: Wed, 23 Apr 2014 13:20:59 GMT Contact: <sip:3145152000 at 192.168.122.57:5060;gsid=d13ae820-caef-11e3-9b9c-6c3be5a59e68> Via: SIP/2.0/UDP 192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4 Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967 Via: SIP/2.0/UDP 192.168.175.130:15060;rport;ibmsid=local.1389145532068_1778704_1816625;branch=z9hG4bK605195140054947...

...77 sip debug peer DLS <-- SIP read from 147.202.nnn.nnn:5060: INVITE sip:6499777777@203.89.nnn.nnn SIP/2.0 Record-Route: <sip:147.202.nnn.nnn;ftag=as4917b107;lr=on> Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0 Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060 From: "6494444444" <sip:6494444444@202.180.nnn.nnn>;tag=as4917b107 To: <sip:6499777777@domain.co.nz> Contact: <sip:6494444444@202.180.nnn.nnn> Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 69 Date:...

...ot; back from 192.168.3.10 ubiphone*CLI> <-- SIP read from 192.168.3.64:5060: --- (0 headers 0 lines) Nat keepalive --- 12 headers, 0 lines Reliably Transmitting (NAT) to 192.168.3.64:5060: OPTIONS sip:6627 at 192.168.3.64:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4a6d0d34;rport From: "asterisk" <sip:asterisk at 192.168.3.6>;tag=as35c7a074 To: <sip:6627 at 192.168.3.64:5060> Contact: <sip:asterisk at 192.168.3.6> Call-ID: 42771eef7db8c7403af5def871cb477c at 192.168.3.6 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 10 Sep...

...ial("SIP/404-cbc9", "SIP/1088|60|tr") in new stack We're at XXX.XXX.XXX.XXX port 17506 Answering/Requesting with root capability 256 12 headers, 8 lines Reliably Transmitting: INVITE sip:1088@201.133.170.82 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK78f35612;rport From: "Weber Automundo" <sip:404@XXX.XXX.XXX.XXX>;tag=as4da46cda To: <sip:1088@201.133.170.82> Contact: <sip:404@XXX.XXX.XXX.XXX> Call-ID: 00b325641a0f0d680014aad165ce6d4c@XXX.XXX.XXX.XXX CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 16 Feb 2005 00:43:27 GMT Allow...