similar to: PJSIP and Kamailio without registration

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

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

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]

> 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? >

Hi all, I saw Matt Jordan's recent Kamailio world talk and was interested in the idea he proposed of stripping out authentication and registration from asterisk and letting Kamailio handle it. All of the tutorials I've seen (e.g. on asipto) show Kamailio forwarding registrations to asterisk. In order to do what Matt suggested would I be correct in assuming I would have to use the

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

Joshua Colp wrote: > Chirag Desai wrote: > >* Joshua Colp wrote: *> >> >* <snip> *> >> >* > Remove "transport=transport-tcp" from your endpoints. *> >> >> >* Joshua...I did that but now my endpoints won't register. * > That should have no impact on things. Can you clarify what you mean by > it doesn't register?

Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x

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

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

Le 15/01/2020 à 19:24, Administrator a écrit : > Hi all, > > we face a strange behavior while connecting an Asterisk16 instance > with PJSIP to 2 providers: we receive error 401 Unauthorized, both of > them having Kamailio as front-end. With other providers -we don't know > if they run kamailio- registration is just fine. > > One of the provider took a pcap and told

On Sat, Jan 18, 2020 at 1:14 PM Administrator <admin at tootai.net> wrote: > > Le 17/01/2020 à 11:54, Administrator a écrit : > > > > Le 15/01/2020 à 19:24, Administrator a écrit : > >> Hi all, > >> > >> we face a strange behavior while connecting an Asterisk16 instance > >> with PJSIP to 2 providers: we receive error 401 Unauthorized,

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

Hi all, we face a strange behavior while connecting an Asterisk16 instance with PJSIP to 2 providers: we receive error 401 Unauthorized, both of them having Kamailio as front-end. With other providers -we don't know if they run kamailio- registration is just fine. One of the provider took a pcap and told us that expiration was set to 0 that's why they don't accept the

Hello, I put together a new tutorial about asterisk realtime integration with kamailio (openser). This time the database used is the one of asterisk, also call routing logic is controlled by asterisk, here is the link: http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb Practically is an easier way to scale starting from existing asterisk installations. The other

Hi I have an installation based on Daniel-Constantin Mierla's excellent Kamailio 3.3 / Asterisk 10 Realtime document ( http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb) but have come across an issue which is a potential problem. In this installation all SIP clients register with Kamailio, and the registrations are forwarded to Asterisk. This means that all

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>

Le 17/01/2020 à 11:54, Administrator a écrit : > > Le 15/01/2020 à 19:24, Administrator a écrit : >> Hi all, >> >> we face a strange behavior while connecting an Asterisk16 instance >> with PJSIP to 2 providers: we receive error 401 Unauthorized, both of >> them having Kamailio as front-end. With other providers -we don't >> know if they run