Archives : September-2015
Home » 2015 (paged 18)
All,I have a question about the Queues.Im using Asterisk 11.13.0 , and I want to configure the following setup :When there is an incoming call to the queue all agents should ring even those that are already in call, they should receive a second call..
Hi!After upgrading kernel from 2.6.32-openvz-042stab108.8-amd64 to 2.6.32-openvz-042stab111.11-amd64 I fail to load the DAHDI kernel modules.Its the package dahdi-linux-complete-2.10.2+2.10.2 and it looks like it compiles and installs just fine (a ..
Im having trouble configuring Asterisk to respond to an incoming out of call SIP MESSAGE. The transport protocol is TLS and the Asterisk version is 10 (its old, but Im kind of stuck with it at the moment). Currently I have roughly the following configurat..
sir ,How to enable SIP text messaging wi..
Im using Asterisk 13.4.0 and DAHDI 2.10.2.Ive got a FXO line that I use for in and outgoing PSTN calls.Unfortunately Im getting a lot of spam calls on the number.I had the extension configured to forward incoming calls to 2 SIP extensions or go to voicemail…
Dear, sirI have installed the Freepbx 12 on amazon could and it and asterisk run successfully. I could registered the sips but I wonder why when we make the call between those sip, each sip cannot hear the sound talking from each side ? could you ple..
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..