Tag : sip
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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..
masters,I’m not an expert on this my friends, but I’m trying to understand which the expected behaviour is from Asterisk side when you deal with the following scenario:Caller —> GSM Gateway with SIM card A —> Asterisk queue —> extension 1000..
,I have the following situation:Local T.38 endpointASTERISKSIP provider (with T.38 support)I am trying to send a fax from my local T.38 endpoint to arbitrary external fax numbers (which I am not in control of, so I dont know if the other end suppo..
Asterisk Project Security Advisory – AST-2014-015 ProductAsterisk SummaryRemote Crash Vulnerability in PJSIP channel driverNature of AdvisoryDenial of Service SusceptibilityRemote Unauthenticated Sessions Severity ModerateExploits KnownNo Reported O..
Using realtime for SIP. Using standard DB schema. Tried mailbox as varchar(50) and bigint(10)sip show peer XXX shows Mailbox: (empty)So MWI isnt working This happened before when we tried to go up to 1.8, so we stayed at 1.4Were forced to go to 13 now.Obviou..
Right now we I am using Asterisk boxes as a gateway between our Level 3 SIP trunks and our customer PBXs. I love and understand Asterisk but the company I am working for is looking for a more Commercial type solution where we can go to a vendor for supp..
,probably this is a FAQ but I cant seem to find it. How to find the RTP IP address of an ongoing SIP call?sip show channels seems to list the RTP address under the very left column called Peer. And it also lists the associated Call ID which I could associ..
All, I am using asterisk-11.12.0 version and I am trying to setup secure call(TLS + SRTP) between two extensions and while making a call, I got following error*CLI> == Using SIP RTP CoS mark 5– Executing [6004@from-office:1] Dial(SIP/6003-00000000,SIP/6004,..
Before I go down a rabbit hole, does the mwi publish/subscription work for non SIP phones?For instance, I have a single voicemail server, connected to multiple asterisk boxes via SIP.On each of those servers, there are a mix of SIPand SCCP phones attached.Current..
I read on the wiki :Asterisk 1.8 will allow to read SIP response codes in the dialplan via *${HASH(SIP_CAUSE,)}*. Additionally make sure youre using the destination channel, not the source channel.But when I use this in my dialplan, this variable..