similar to: PJSIP question urgent

A few months ago I started using and had to abandon PJSIP because my dialplan could not read the inbound signalling IP address, which I can read now in Asterisk11 using CHANNEL(recvip). My app relies on this information. The question is, is it possible now access the signalling IP of an incoming SIP call using PJSIP? Philip

I am stuck in channel PJSIP trying to see the real flow of SIP messages, what in regular sip we used to type "sip set debug on" Also, is there an automated way to convert sip.conf options to pjsip.conf? Philip

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 >

?Asterisk 13.11 rc1 ./configure LDFLAGS="-z muldefs" --libdir=/usr/lib64 --with-unixodbc=$(odbc_config --include-prefix)/ --disable-dev-mode --with-pjproject-bundled ?checking for pjsip_dlg_create_uas_and_inc_lock in -lpjsip... no checking for pjsip_tsx_create_uac2 in -lpjsip... no checking if "pjmedia_mod_offer_flag flag = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE" compiles using

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,

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:

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

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

On Sun, Jun 11, 2017, at 01:31 PM, Michael Maier wrote: > On 06/11/2017 at 04:39 PM Joshua Colp wrote: > > On Sun, Jun 11, 2017, at 11:34 AM, Michael Maier wrote: > > > > <snip> > > > >>> > >>> PJSIP uses a dispatch model. The request is queued up, acted on, and > >>> then that's it. The act of acting on it removes it from

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?

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

Hi, In PJSIP configuration, I thought that "from_domain" parameter in a endpoint permits to have two SIP peers with the same usernames with a different domain. I've tested at the transport level, I see no changes. I've also tested with realm parameter in auth configuration, it seems to change only the digest auth value during registration. I'm pretty new with 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

>>>>> "JC" == Joshua Colp <jcolp at digium.com> writes: JC> This stems from PJSIP not being dynamic with transports (it JC> doesn't like its environment changed to that degree while JC> in use). I'm afraid if your IP changes you'd have to restart JC> Asterisk when you are using PJSIP. Wow. I say this having voted for pjsip over the listed

Hi all, Is there any native way to save endpoint statuses to database? I use asterisk 13 with pjsip and realtime, and didn't found proper way. I read that there is config parameter in sip.conf: rtupdate=yes. But how can I do that with pjsip? Or I should use sip.conf with pjsip simultaneously. Or is there any kind of hooks, which allows make custom action on endpoint status change. Thanks.

Le 27/11/2018 à 12:13, Joshua C. Colp a écrit : > On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...] >> >> [TOOTAiAudio] >> ; >> ; Call our gateway >> >> exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1}) >>  same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T) >>  same = n,Return >> >> exten = h,1,NoOp()

Hello, On a newly built Asterisk 13.13.1 system, I can't make HEP work with chan_sip (though I could make it work with PJSIP on another instance). Looking either at [1] or hep.conf, I can't see anything meaning HEP requires PJSIP. Before diging deeper, may I simply ask if HEP requires PJSIP or not ? What about a line mentioning the answer in above documents (to keep other from wondering

I am using AMI Originate to perform a new outbound call. The SIP Provider we send the call to wants us to pass the caller id of the person we are calling for in the Contact header. For the AMI Originate, I pass the caller id information data in the CallerID field. However, this is never being passed through the PJSIP INVITE header Action: Originate ActionID: S598 Channel: PJSIP/133 at 1002

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:

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