Archives : March-2015
everybody, Ive a matter with the queue annoucement with the thereare, because if I put just one member in my configuration (member => SIP/2098), the ivr gave me that I was the firt or second in the next at the queue. But the problem is, if I add ..
I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. Im getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection.Howev..
I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. Im getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection.Howev..
HelloI use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom Germany. We have sometimes problems with incoming and outgoing calls. I hope I can explain it understandable.For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de..
I am trying to debug a SIP issue, between an Asterisk 1.2.32 system that is behind a network device to which I dont have ready access, which is performing NAT with possibly some kind of SIP ALG, and an Asterisk 11system on a public IP.My question..
AllHow to have access to the IAX2 call statistics inside the dialplan (not CLI)?I have no IAX2 clients (yet) to test, but do RTPAUDIOQOS.* variables do the job?Are they available to IAX2 calls as they are for SIP?Stats like total packets sent and receiv..
The HASH function is really useful when you have to deal with values loaded using func_odbc, but how do you use with the LOCAL function? Is it possible to define a HASH as LOC..
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What is your experience: if you plan to make a test SIP call to check voice quality overa connection what would be the best call duration? The point is that we should have a call long enough to be able to catch/hear impairments that the connection ..
All;I have a prob..