Archives : October-2014
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I am new to mailing list ,please correct me if the way of posting is not correctRelpying to :Re: make asterisk do something when an outgoing call ispicked up (lee)For making asterisk do something on outgoing call Dial application is itself used L..
Open source projects survive on freedom of communication. Such projects are diminished when a community member can no longer participate, as the project no longer benefits from their opinions and insight. However, one of the few things worse than t..
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..
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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 ..
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..
I am very new to Asterisk world.This is what we are trying to achieve.We have Asterisk Server in our lab with Ver. 11.2.1 – FreePBX. We have Dialogic HMP application, which we want to get registered with Asterisk. After registration we would like..
Can we expect a SILK codec for 13 ? Or does the one for 12 work f..
Anybody care to share a script or snippet of what they use to remove leading and trailing silence from customer recorded files?Ive fiddled with sox to remove the leading; reverse the file; remove the now leading; and reverse the file again, but Im ..