Archives : February-2015
In Version 1.8 asterisk introduced this parameter preferred_codec_only, when set to yes the 200 OK to the INVITE contains 1 codec only from the available ones in the user sip profile.But in version 13.1 (I think version 11.2 also) is not working l..
list,i have created a queue with and i have a question related to musiconholdf there is any way to set the musiconhold just for caller not for agent logged in the queuethanks an..
Can anyone recommend a good WebRTC phone to use with Asterisk? I do not mind if it is commercial or open source.Customers are starting to ask for web solutions and we need to start testing. — Telecomunicaciones Abiertas de México S.A. de C.V. Car..
listei have creat i trunk sip and inboun route for inbound calls the issue whe i use the DID in inboud route i have a error No DID or CID Match.but when i leave this DID field blank i can route the call without any issuehow can ido in order to use ..
I would like to do some tasks after the CDR has been closed, and the CDR(end), CDR(billsec) and CDR(duration) fields are available. I have tried to do that on the h extension, but it seems the CDR is not yet complete in the h extension.When is the ..
, I need your help ,I have an incoming call x the ivr and the operator takes the call. ext 101 , If a second call reenters and the operator is talking, I want to send to the extension 102 I use the Variable DIALSTATUS , but not workingcheck IVR[IVRINMA]ex..
Hay guys, got trouble with registration with cisco 7975Here is the debug :REGISTER sip:192.168.1.4 SIP/2.0Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK35076381From: ;tag=0c8525a68961001f44d503e2-d9359bd3To: Call-ID: 0c8525a6-89610004-b972d038-5864c98e@192.168.1.61Max-Forwar..
That would be a CNG frame (AST_FRAME_CNG). While a frame exists to convey CNG to capable channel drivers, CNG itself is not implemented or handled in chan_sip (or most of Asterisk).Theres a lot of intervening points between sip_write and whatever genera..
Hay guys, have question.When I do regular dial I use $this->AGI->get_fullvariable(${PJSIP_DIAL_CONTACTS(.$callObj.)},false,true);to get all contacts of current endpoint and so I dial to all phones at once, but if I exec QUEUE, I have just one phone rin..
Friends, I encountered a strange issue. I am running Asterisk 11.8.1 on Cent OS with no firewall running. It has 3 NIC interfaces.in my sip.conf I haveallowguest=yes bindaddr=0.0.0.0udpbindaddr = 0.0.0.0But my Asterisk instance does not pick the c..