similar to: PJSIP works on UDP but not TCP

Joshua Colp wrote: <snip> > Remove "transport=transport-tcp" from your endpoints. Joshua...I did that but now my endpoints won't register. Kind Regards, Chirag -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150304/ea40a0bd/attachment.html>

Hi all, I have Asterisk 13 running and I'm currently trying to get PJSIP working on TCP. My transport looks like this. My box is not behind NAT. [transport-tcp] type=transport protocol=tcp bind=0.0.0.0:5061 My endpoint looks like this: [user1] type=endpoint transport=transport-tcp context=local_out disallow=all allow=alaw allow=ulaw allow=g722 auth=user1 aors=user1 direct_media=no

> Joshua Colp wrote: > > There should be nothing different, except for how you configure things. > What is the full PJSIP configuration? What is the environment where > Asterisk is running? Is ICE actually in use on the other side? What is > the full SIP trace? > The full configuration is here: http://pastebin.com/XqZG1m5X I am connection over TLS / SRTP on port 5063. When

Chirag Desai wrote: >I've tried explicitly setting the IP in bind and leaving it as above. >Nothing seems to come into asterisk. Although, as mentioned I can see the >SIP messages when I ngrep 5061. I got it working, I can see the sip traffic in the CLI now. I was trying to match on the IP of kamailio, when really I should have been matching on the domain name in the sip message

Hi all, I was using 13.5 but upgraded today to 13.9 (13.10 came out a few hours after I upgraded). On both 13.5 and 13.9 asterisk seems to use 100% of the CPU. This usually happens a few hours after starting asterisk. A restart of asterisk gets the CPU back down, but only for a little while. There asterisk box has no call traffic flowing through it, just 15 or so registrations. I'm sure

From: Matthew Jordan <mjordan at digium.com> > > > >> If the INVITE request is not shown in the CLI with 'pjsip set logger > >> on', then Asterisk is not actually receiving the request. > >> > >> Does a pcap show the message being sent to the correct IP/port? If you > >> change the transports to bind to port 5060, does that change

OK, it stopped working. It turns out the transport and endpoints in PJSIP are ok. I can send an invite from my unregistered snom phone and I can see some activity in the CLI. However, when I dial from my snom to Kamailio and have it pass the message to asterisk, PJSIP seems to ignore the sip messages even though they are there. Is there something wrong in the invite that I'm missing? U

I'm dialling from the snom and every few calls asterisk sends media to the phones external IP and it works! And then now and again it sends the media to the phones internal IP and I hear nothing. I'm really at a loss. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Thanks Joshua, I assume by query asterisk you mean I'll need to query it via AMI? Is that information available via AMI? *Nick Olsen* Network Engineer Office: 321-408-5000 x103 Mobile: 321-794-0763 On Tue, May 26, 2020 at 2:57 PM Joshua C. Colp <jcolp at sangoma.com> wrote: > On Tue, May 26, 2020 at 10:48 AM Nick Olsen < > nick at floridavirtualsolutions.com> wrote: >

Joshua So once a transport is pulled from the transports table in realtime during asterisk startup it can't get any updates? Can a new transport be added to the table and the associated endpoints be updated to use the new transport, or are transport types only read at startup across the board? Thanks Bryant ---------------------------------------- From: "Joshua

Joshua We have completed more testing this morning and when we remove the realtime cache options from the sorcery file the endpoints complete registration, but we pjsip show/list does not offer any feed back at all, We also can't send any pjsip send notify commands as they say they don't have an endpoint there. Something has changed in the cache part of the system that is breaking

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp Sent: Sunday, October 04, 2015 12:44 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] pjsip realtime registrations not pulling from ODBC On 15-10-04 01:42 PM, Bryant Zimmerman wrote: >

Asterisk Project Security Advisory - AST-2014-003 Product Asterisk Summary Remote Crash Vulnerability in PJSIP channel driver Nature of Advisory Denial of Service Susceptibility Remote Unauthenticated Sessions Severity Moderate

Asterisk Project Security Advisory - AST-2014-003 Product Asterisk Summary Remote Crash Vulnerability in PJSIP channel driver Nature of Advisory Denial of Service Susceptibility Remote Unauthenticated Sessions Severity Moderate

On Mon, Jun 5, 2017, at 12:00 PM, Joshua Colp wrote: > On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote: > > On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > > > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: > > >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: > > >>> Just a guess (without knowing about your network), but are

Asterisk Project Security Advisory - AST-2014-016 Product Asterisk Summary Remote Crash Vulnerability in PJSIP channel driver Nature of Advisory Denial of Service Susceptibility Remote Unauthenticated Sessions Severity Critical

Asterisk Project Security Advisory - AST-2014-016 Product Asterisk Summary Remote Crash Vulnerability in PJSIP channel driver Nature of Advisory Denial of Service Susceptibility Remote Unauthenticated Sessions Severity Critical

Thanks for the info, Joshua. Does PJSIP handle database access the same way Chan_sip did? We had a number of boxes running chan_sip referencing the same mysql server without issue. We're going to attempt to get a backtrace on the next occurance. We're also going to run a local copy of the database on the same physical asterisk instance and have the system reference it. Just to

Nope, there are no contacts to show that pertain to these endpoints (only my SIP trunks show up). On Mon, Feb 15, 2016 at 5:31 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > >> Does this help: >> > > Yes, the transport parameter is in the Contact header so it's interesting > it didn't work. If you use pjsip show contacts what

This fixed the problem. Developer before me wrote some code to build up the dial string. Always thought that string appeared off, but it worked so I left it alone. Thanks Joshua and George for helping with this. Have a great day! Dan -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Cropp Sent: