I am running asterisk 188.8.131.52 and have compiled in the srtp module All but Snom phones are working. I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they worked for a few hours. This morning all snoms are reporting this when try..
All, Along with my asterisks server, all incoming calls to my D-linkDPH-80 ip phones are are working fine while calling from soft phones with good voice clarity. But not able to make outgoing calls from the same D-link DPH-80 ip phones to either s..
TE410P card down. I have three (3) TE410P in one machine running asterisk with SS7. My problems started last week when one of my cards started switching to E1 every time after reboot.I set the following in dahdi.conf and that solve the problem. /etc/modprobe..
Running build 184.108.40.206 (compiled from source) I seem to be having an issue with codec negotiation. I have a Grandstream HT503 FXO port connected to a pstn line, a Polycom SP501, and a SIP trunk with callwithus. What Im essentially looking to accompl..
I´ve compiled asterisk-220.127.116.11 on my Debian based distro (Pinguy)
I also compiled iksemel (v1.4) with the option 2./c..
I would like to try the ILBC codec on one of our systems. The system is currently running Asterisk 18.104.22.168 installed from the Asterisk provided repositories for Centos 5. Is there a process for installing the ILBC codec under this environment, or w..
Hi Im using asterisk 22.214.171.124 (with a couple of patches) I have the following scenario… SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call..
We have Asterisk 126.96.36.199.2 connected to a sangoma E1 card. The problem we are having is that we have a calling card type application and when people enter the number to be dialled we call the Dial application. It gets back an indication that the num..
Can you please let me know if the asterisk has speech to text and text
to speech facilities?