Archives : November-2016
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Related to http://lists.digium.com/pipermail/asterisk-users/2016-November/290384.html, at the moment Im passing one variable via DIAL.Now Id like to pass a whole bunch, and my idea was to rather than having a great string ofb(synctest3b^setVar^1(something)^2(m..
I am hoping someone else has seen this and can offer a solution or at least a direction to investigate.I am running 11.23.Most of my clients are fine but one has a strange behaviour.He has a Grandstream HT701 like most of my clients who use an ATA..
The Asterisk Development Team has announced the release of Asterisk 14.2.0-rc2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 14.2.0-rc2 resolves an issue reported by ..
The Asterisk Development Team has announced the release of Asterisk 13.13.0-rc2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 13.13.0-rc2 resolves an issue reported..
In my setup, which is FreeBSD, using pjsip 2.5.5 as sip backend I am observing a regression when testing the latest Release Candidate.Any calls get refused and the following error shown on console:[Nov 22 10:49:26] WARNING[101105]: res_rtp_asterisk.c:2..
Is there a way to force a SIP client to re-register using a SIP command (or an AMI command)? If not, is there some other standard way to do so – or would I have to post/get to a web GUI of the phone (unique to each model) to force a reset, ..
From [1], I read the following example:[applicationmap]retrieveinfo => #8,peer,Set(ARRAY(CDR(mark),CDR(name))=${ODBC_FOO(${CALLERID(num)})})Then I wrote is my own simplified example:[applicationmap]setuserfield => 11,peer,Set(CDR(userfield)=FOO)W..
I have a working telephone project that uses SIP.js 0.7.5 with Asterisk on the server side. Currently it handles both audio and video correctly.The SIP.js webpage has instructions for setting up a datachannel through a SIP call. The online demo u..
Hellowhen using Asterisk version 13.12.2 I notice that it takes up to 30 seconds (sometimes even longer) for a call queue to call its members.Example 1 :[Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15] Queue(SIP/incoming-00000246, myqueue1,,,,300,..
In the musiconhold.conf example, it says:announcement=queue-thankyou;If this option is set for a class, then when callers get put on hold, the specified sound will be be played to them.Im using the m option in Dial and was hoping to make use of t..