Tag : PJSIP
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I am currently trying to send faxes via T.38 using PJSIP (newest version 2.3) with Asterisk 13.0.2. After having configured PJSIP, I have seen several things the cause of which I would like to know.1) Ports and IP addresses which PJSIP bind toI h..
Im 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.Im currently running Aster..
Howdy,Im trying to re-write my voicemail check extension.I formerly used the SIPPEER function to get the mailbox for a peer with${SIPPEER(${peer},mailbox)}Is there a way to do this with PJSIP now that Ive converted over?I see a function PJSIP_ENDPO..
I just finished installing Asterisk 13 on our test server and I can now use PJSIP to register phones and make and receive calls. The only problem I am having is that when I register multiple phones to a single account only one of them rings.The AOR ..
all, We have recently upgraded some of our services to Asterisk 12 with PJSIP. We have 2 issues related to DTMF:1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMFsettings according to the incoming INVITE – RFC2833 or inba..
I need to respond with 380 Alternative Service. Is there a way to do this in PJSIP? Please note that I am not picking up the call. For instance, the Transfer app closes the call if you did not answer it first. There is a bug open about this. I want..
everyone, I am starting to work with PJSIP on release 12.1.0.rc3.I used to have Asterisk 1.8 with the regular sip channel. I was using the usereqphone settings in order to set user=phone on from and to URIs.Is there a similar config..
everyone, I am running asterisk with release 12.1.0.rc3 and PJSIP. I have a peer which sends OPTIONS method for session keep-alive, and the asterisk is not responding to it. That of course disconnects the call after a few minutes.Is there a setti..
I am trying to make PJSIP work with my Cisco SPA504G phone.I have no problems making it work with the chan_sip driver.When I configure my phone, it indicates the contact was added– Added contact sip:7001@192.168.9.142:5063 to AOR 7001 with expirat..
Asterisk Project Security Advisory – AST-2014-003ProductAsterisk SummaryRemote Crash Vulnerability in PJSIP channel driver Nature of Advisory Denial of ServiceSusceptibility Remote Unauthenticated Sessions SeverityModerate Exploits Known No Repor..