Dear In essence Caller ID ETSI and FSK US (Bellcore) is based on the same pattern as; _____________ ____________________________ _____________________________ |First Ring burst |_500ms_|Channel seizure 300 bits|__|Mark Signal|__| Caller ID Message|_..
I have a process that runs on a server and does a simple asterisk -rx sup show peers > /tmp/peers and then looks for any (Unspecified) items and reports them as having lost connection. My server is running 1.4.43 and the two boxes I am monitoring ..
We use AMI to originate calls. Sometimes, lately every morning, the AMI Originate process operates extremely slowly. I cannot see the calls in core show channels verbose, I dont know where they are, what state they are in, after 2-3 minutes the ca..
Ive upgraded my asterisk 1.4 to the version 1.8.11. After making some adjustments to the configuration files to port it to the new version, calls between registered phones in asterisk, work fine, but inbound calls coming from the SIP trunk I have w..
To put it simply, is the process where Asterisk tries to redirect the RTP media stream to go directly from the caller to the callee. Be careful that some devices do not support this (especially if one of them is behind a NAT). The default setting..
On 04/19/2012 05:59 PM, bilal ghayyad wrote: > Dears; > > I see this at the /var/log/asterisk/messages: > > [Apr 20 01:49:48] ERROR codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory If you arent using a DAHDI transcod..
After some customer complaints I find myself tcpdumping, gzipping and transferring large packagedumps over the network to be analyzed. While this manual process isnt a long-term solution, Im evaluating different options. Aside from the manual thin..
Is anybody running multiprocess of Asterisk on a server ? Does it work well? My configuration is too complicated.I know Asterisk on a virtual machine works well. but OS overhead is considerable. that is why I want to divide a process. Regards, Mak..
I am having problems with a deadlock in Asterisk 220.127.116.11. The system is a x86_64 machine that is being used as a callcenter. The agents log in via the AgentLogin application, and each Agent/XXXX channel is assigned to one or more queues. A custom separ..
On 01/28/2012 10:22 AM, Din Assegaf wrote: > All, > > Im trying to upgrade asterisk server to 1.8.x from my asterisk 1.6, > > But when making A Call from SIP Client, I got cli Warning … and no > call has been made. > > My Sip Client is using lib j..