Archives : September-2015
I have many endpoints and each endpoint has some parameter in common so i wonder is there any way to config one for all endpoints? Like in my example I have two endpoints and I repeat the same thing,[100]type=endpointaors0auth0-authallow=ulaw,alaw,gsm,g726context=from-intern..
I think this may be a possible bug, but wanted to see if anyone else has run into something like this before. I am running Asterisk 11.6.0 currently. For the most part it works and works well. What happened yesterday though has me scratching my h..
I am using the asterisk 12 with pjsip, I wonder how could I config the instance meesseging for pjsqip in asterisk 12 ? What is the default message context for pjssip ? I use the default extension.conf from the installation and I successfully could m..
Dear fellows, how are you?I´m offering TDM routes for Brazil (landline and mobile destinations) with low prices, TDM ccts (no GSM), ASR and ACD great. Pre paid, by paypal. If you have interest, please just let me know.With Best Re..
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..