similar to: ​ PJSIP and Non Media Proxy

Dear Asterisk-Users, By means of Asterisk 11 and sip.conf, I got success implementing early media. That is, all information that come from callee (SIP 183 message/ SDP) is passed to the caller without any modification in the SDP body. However, in Asterisk 13 and using pjsip.conf I'm still failing to do the same thing. See: Softphojne1

Is there a guide on how to use PJSIP and never have the media travel inside Asterisk? No matter what I do, I cannot make this work. Philip Orleans -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200213/c80f007e/attachment.html>

Hello, I've started to play with PJSIP and got stuck at the following problem. I need to retrieve SIP Call-ID associated with PJSIP channel. For inbound channel I can use ${PJSIP_HEADER(read,Call-ID)}, but that doesn't work for outbound channel even in pre-dial or hangup handler. Whatever I do PJSIP_HEADER seem to be unable to read headers for outbound channel. Here's what I do:

Hello List I am in the progress of migrating from chan_sip to pjsip. I fear I have missed something on how hints need to be specified for pjsip. For chan_sip I have configured sip.conf subscribecontext = localuser and in the dialplan I set: [localuser] exten => 11,hint,SIP/11 Now if a phone subscribes to '11' this works. Now I try to get the same working for pjsip. I understood

We are having an issue where on the latest version of asterisk when configuration pjsip via realtime. we do a pjsip list endpoints it shows our endpoints but lists them as invalid. When we do the pjsip list endpoints again it shows no objects. This applies to pjsip list aors as well. We did not have this issue on our older asterisk 13 installs. My guess is something has changed

On 2/15/16 12:50 PM, Joshua Colp wrote: > Carlos Chavez wrote: >> Is it possible to use serveral protocols for a single transport section >> in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you >> cound use webrtc along with your phones but if I try: >> >> [transport-udp] >> type=transport >> protocol=udp,ws,wss >> bind=0.0.0.0 >

Hi All, I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code on CentOS7. -- https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject The installation is OK. But the connected SIP cilents (both Linphone on Windows7) cannot communicate. I hope your comment such as the testing for resolving the problem. My status is the following(1 and 2). Why 'Everyone

Hi Joshua, thanks for answer. In this particular test my extension is on a simple network. There is no NAT, just an asterisk running on a virtual machine on a 7- 64bit CentOs. I am simulating an environment to be able to use PJSIP on my client. And even in this small environment, my extension does not call. My problem with NAT was with SIP "one way audio" on a client. All of this

Hi guys, I'm building a PoC Asterisk 12 cluster based on a number of guides I've found on the net. The basic concept is using ARA in conjunction with DUNDi. I have set up ARA with pjsip according to this excellent guide here: https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime This is working nicely, so now I am turning my attention to DUNDi, as per this guide here:

Has anyone used Telynx as a SIP trunk provider?? It works with chan_sip but it I seem to be having problems trying to set up a PJSIP trunk.? I always get a 401 Unauthorized when they send me a call.? I know my username and password are correct since I can register and PJSIP uses the same information for inbound as for the registration.? Unfortunately their support department said "PJSIP

Hey all I am trying to configure all my pjsip transports form a database table. The issue I am running into is that pjsip is auto binding to 0.0.0.0:5060 before it reads my list of transports from the database. This means that my entries for port 5060 are already bound and the settings in the database are not loaded. When loading the transport form the .conf file it works as expected

Hello, Is it possible to mix PJSIP realtime and flat file configuration in pjsip,conf? What we want is to set up endpoints in the ps_endpoints table with some columns set but most being NULL, and then allow end-customers to optionally add configuration by adding a pjsip.conf section. For example, in ps_endpoinds might be an endpoint with id "asterisk-1" with the transport, aors, auth,

Hi all, (sending this again from the correct address) I'm running Asterisk 13.8.0 (I need to check if that happens with 13.9.1 too when I have the time to build it) with PJSIP realtime config. I've defined several aors in the table ps_aors, like this (real url replaced by myurl): *CLI> pjsip show aor pbx-node-1 Aor: <Aor..............................................>

Thank you for the speedy reply. My originate string is something like the following where xxxxx is really the sip provider's supplied IP address 1234567890 is really the phone number I am dialing PJSIP/outbound.vitelity.net/1234567890 In the chan_sip based solution, it's... SIP/outbound.vitelity.net/1234567890 Have a great day! Dan -----Original Message----- From:

I'm working with a SIP provider to try and transition our sip connection with them to PJSIP. I thought I had transitioned the settings correctly, but whenever I attempt an Originate it never even tries to send any PJSIP messages. I'm currently running Asterisk 13.0.0. Anyone have any suggestions as to what I am doing wrong? The SIP provider says the latest version of Asterisk they have

Hello. I am having a lot of problems with SIP through NAT. So, I decided to adopt PJSIP. However, I am not able to make the extensions ring when receiving a call from the queue. I'm using telnet to include the extension and on the asterisk console, it even shows Called PJSIP/6001, but the extension doesn't ring. If I call from extension to extension, it works normally. telenet:

Hello, How should I configure Asterisk (13.7.0) to get persistently PJSIP SIP messages in a log file and not in console ? I would expect adding "debug=yes" in pjsip.conf to produce the same output as "pjsip set logger on". Am I understanding correctly ? Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Yes, we're implementing the dialplan in realtime too. Here the contents of sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints aor=realtime,ps_aors contact=realtime,ps_contacts [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips Cheers, Francisco. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf

I found that PJSIP allows only one asterisk per box. I tried to start several asterisks with the parameter -C and PJSIP only worked on the first process. In the other processes, the command "pjsip reload" was absent. Each pjsip transport in the second and subsequent processes was bound to a different IP in a multihomed box, something I routinely do with regular SIP. Am I wrong?

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