similar to: PJSIP and Kamailio without registration

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

> 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, 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

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]

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 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

2017-02-16 14:27 GMT+01:00 Joshua Colp <jcolp at digium.com>: > On Thu, Feb 16, 2017, at 09:11 AM, Olivier wrote: > > Hello, > > > > I'm currently testing a so-called VQ RTCP-XR feature from a a SIP > > hardphone. > > > > When a phone has enabled this feature, it would send a SIP PUBLISH to its > > SIP Server letting this server dispatch to

Hello, I'm currently testing a so-called VQ RTCP-XR feature from a a SIP hardphone. When a phone has enabled this feature, it would send a SIP PUBLISH to its SIP Server letting this server dispatch to whatever is needs to. These messages are sent during calls but may also be sent when a call is over. At the moment, I'm using Asterisk to serve these SIP phones so my Asterisk box

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

Hi all, Sorry if this has been asked before. I searched a lot, but found conflicting answers, so hoping for some clarification. My question is does Asterisk 13 support OPUS? If so which version exactly? If asterisk 13 requires a patch, which is the correct one and where do I get it? Kind regards, C -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

Hi Guys Is it possible to register Kamallio directly to our SIP provider then load balance the RTP to 2 asterisk servers? We cant do the registration from the asterisk boxes as we want to do it directly from Kamallio. Is this possible? -------------- next part -------------- An HTML attachment was scrubbed... URL:

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, Does anyone have some good examples of a Kamalio or OpenSips configuration that integrates with Asterisk? Essentially I want to use the SIP router as the UA, but still run all the calls through Asterisk (for dialplan, etc..) I've looked for examples on the project web sites, but I haven't found anything decent yet. Thanks. -- James

Hi All, Do you'll have any recommendations on a Linux based Customer Management and Pre-paid Billing system for Asterisk, Freeswitch or Kamalio? The system should also allow customers to register, login, buy more credit, view call records, etc. Commercial or Open-source are ok as long as they run on Linux. Thanks, A. -------------- next part -------------- An HTML attachment was scrubbed...

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:

used Kamalio to "supplement" the features that Asterisk either doesn't provide or doesn't provide in as nice a form as the OP desired - can't really speak beyond this as I am not one of them. ------=_NextPart_000_010C_01CB6EAA.3AC2C610 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable <html

> 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

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