similar to: Destruction of SIP dialog for OPTIONS requests

Hello, I got situation which is unclear for me, hope somebody could explain this. A calls to B INVITE sent from A to B B responds with 100 Trying B responds with 183 Progress After 10 seconds: Asterisk CLI: Scheduling destruction of SIP dialog '..' in 32000 ms (Method: INVITE) Asterisk sends CANCEL _instantly_ B responds with 200 OK and 487 Request Terminated Asterisk confirms 102 ACK

Hi Friends, need to help. *I have problem about sip : SIP/2.0 401 Unauthorized* Is it require to nathelper module in kamailio ? *what can i write kamailio.cfg file when kamailio and Asterisk on same network?* Scenario is like as : ----------------------------- 1) kamailio server on 172.18.100.74 kamailio.cfg ( nathelpler module ) ----------------- loadmodule "nathelper.so"

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

Hi, I want to have Kamailio in front of one or more Asterisk boxes. I don't think it is necessary for Kamailio and Asterisk to register with one another. I'd like for PJSIP to recognise Kamailio by its IP address. I have two boxes, both have public IP addresses, they also have private IP addresses and can communicate with each other. I have a Snom phone accessing Kamailio via its

On Mon, Apr 6, 2020 at 2:06 PM Administrator <admin at tootai.net> wrote: > Hello, > > We have a provider which is using Kamailio as front end. Our asterisk > 13/chan_sip server has no problem to register and pass/receive calls > form this provider. > > Now we want to move to asterisk 16/pjsip and face problem. Registration > is OK but when we pass a call our INVITE

Hello, We have a provider which is using Kamailio as front end. Our asterisk 13/chan_sip server has no problem to register and pass/receive calls form this provider. Now we want to move to asterisk 16/pjsip and face problem. Registration is OK but when we pass a call our INVITE never receive answer from the provider. We opened a ticket to their support but in the mean time we want to know

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

Joshua Colp wrote: > Have you configured any transports? PJSIP does not create any by > default, you have to explicitly configure them. Without them no traffic > can come in or go out. You can also remove the explicit transport from > the endpoint. Yes I have two transports [transport-udp] type=transport protocol=udp ;udp,tcp,tls,ws,wss bind=0.0.0.0:5061 [transport-tcp-kamailio]

Hey John, In one installation I have, we use several monitoring tools (nagios based and custom scripts based) and we have the following: ; Reply OK to SIP:OPTIONS [public] exten => s,1,Wait(1) same => n,Hangup : For Nagios exten => nagios,1,Wait(1) same => n,Hangup NOTES: 1- We have context=public in sip.conf, if you have anything else, you must update the dialplan above

> 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 anything? >

I've got this setup in a test context. [test] exten => s,hint,SIP/7124 exten => s,1,NoOP(Options to $EXTEN) same => n,Hangup() exten => _x.,hint,SIP/7124 exten => _X.,1,NoOP(Options to $EXTEN) same => n,Hangup() exten => Anonymous,hint,SIP/7124 exten => Anonymous,1,NoOP(Options to $EXTEN) same => n,Hangup() I added hints to see if that would make a difference

Wow, commenting those two lines worked out fine for me, thanks! On 9 May 2013 09:34, Giacomo Tagliabue <giacomo.tag at gmail.com> wrote: > Thanks, > Also, every method inherited by LoopBase causes the same error, while Loop > methods go smooth. > > > On 9 May 2013 01:05, Andrew Trick <atrick at apple.com> wrote: > >> >> On May 8, 2013, at 7:43 PM,

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

On Wed, Mar 4, 2015 at 12:47 AM, Jarrod Cuzens <jarrod at mogl.com> wrote: > For those that were interested I have attached the kamailio.cfg which we > have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the > following yum packages: > > kamailio.x86_64 4.2.1-4.1 > @home_kamailio_v4.2.x-rpms > kamailio-auth-ephemeral.x86_64

For those that were interested I have attached the kamailio.cfg which we have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the following yum packages: kamailio.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-auth-ephemeral.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms kamailio-bdb.x86_64 4.2.1-4.1 @home_kamailio_v4.2.x-rpms

On May 8, 2013, at 7:43 PM, Giacomo Tagliabue <giacomo.tag at gmail.com> wrote: > Hello, > I am building a loop pass following these instructions: http://llvm.org/docs/WritingAnLLVMPass.html > Everything works fine, I did it many times for Function Passes, but in the runOnLoopmethod, whenever I call a method of the loop L passed as argument, for example L->begin(), I get the

Thanks, Also, every method inherited by LoopBase causes the same error, while Loop methods go smooth. On 9 May 2013 01:05, Andrew Trick <atrick at apple.com> wrote: > > On May 8, 2013, at 7:43 PM, Giacomo Tagliabue <giacomo.tag at gmail.com> > wrote: > > Hello, > I am building a loop pass following these instructions: >

On Thu, Jan 29, 2015 at 2:43 AM, Kirill Marchuk <62mkv at mail.ru> wrote: > Hi all > > Have recently watched Matt Jordan's session on Kamailio World 2014 > > On slides 26-29 of his presentation > (http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf) > he speaks about a (completely new, for me at least) approach to build

Op 9-2-2019 om 19:27 schreef Giacomo via dovecot: > I got a core file this morning. > > opening it with gdb I get this: > > (gdb) core imap.core > Core was generated by `dovecot/imap'. > Program terminated with signal 6, Aborted. > #0? 0x0000000011c1347a in ?? () > (gdb) bt > #0? 0x0000000011c1347a in ?? () > #1? 0x0000000011c13444 in ?? () > #2?

Works for me.... Thanks, Hristo Benev -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alexandru Oniciuc Sent: Monday, May 17, 2010 6:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] R: new way of asterisk and kamailio(openser) realtime integration