Archives : September-2015
I am using asterisk 13.5.0 and although my AMI-user has read=all and write=all permissions, I don´t get any QueueCallerJoin Events fired, when a new caller calls into a Queue… Strange enough, a QueueCallerAbandoned Event is fired, when the cal..
Everyone,I am trying to make use of asterisk-java live and had some questions for the mailing list however, it does not seem like its an active mailing list? Is the project dead?T..
i am trying to receive a call from freeswitch without transcoding , asterisk and freeswitch are installed on same machinein asterisk cliwith sip set debug onv=0 o=FreeSWITCH 1442495774 1442495775 IN IP4 127.0.0.1 s=FreeSWITCH c=IN IP4 127.0.0.1 t=..
All,All,I am trying to create an Inbound route destined to a Ring Group through a SIP trunk. I am able to call the extensions directly, but unable to call a Ring Group or an IVR through the Inbound Route config. I am really not sure, what i am missi..
folks, I have one server with multiple companies (multi-tenant).From AMI I get all events of all extensions so any one that connect can see other extensions, from different company (context).How can I limit specific user to get just specific conte..
list!Sorry for kinda dumb question, I guess, but I have too little time to research it by myself. I have a SIP packet, which looks like this:INVITE sip:XXXXXXX@10.186.35.98:5060;user=phone SIP/2.0Via: SIP/2.0/UDP 10.186.0.38:5060;branch=z9hG4bKh4utm43008vheqk093b0.1Call-..
I have asterisk 13.5 configured with a simple dial plan, 3 SIP clients two Laptops and smartphone with softphones installed. Now I am trying to store cdr into a database but not able to make a connection of ODBC drivers to MySQL is there an option..
all,im build and using a voip pbx system using OpenSIPS as a router (i need to serve thousand of users…) and an Asterisk server as media box, for IVR, queues and so on.Ive a PATTON PSTN GW (172.20.1.4), the VoIP OpenSIPS ROUTER(172.20.1.2) andnIn queu..
Greetings All, Regarding this archived post. http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html Did anyone ever find an solution to this? Ive got a new box running 13.3.0 with the exact same issue. For those that dont read ..
Id like to use queues in Asterisk and I have a few basic questions (Im a newbie) about Queues:1.- What are the differences between Agents and Members (if any)? 2.- I want to implemment a small call center and I think the best way it is by using dyna..