Archives : March-2015
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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..
I know there are people with much experience in asterisk, and Iwant to ask if anyone had experiance with this gw http://www.eurotech-communication.com/products/voip-gateways/VoIP-32-CHANNELS-2E1-PRI-1U/Im having trouble getting connect with asteriskany..
We have a FreePBX-12 / Asterisk-12 setup that supports about 24extensions, most internal Snom870s but six or so external (Jitsi-2.8). we use TLS and SRTP everywhere on our side of the fence.The server host is a dedicated atom(tm) box using the Free..
Im having an issue with CDR. Basically, I expect to have all legs of a call having the same linkedid and differing only by the sequence value. That does happen, but Im getting null dst values after doing an attended transfer.Im not sure if this i..
I found an issue with how PJSIP handles a typo in the Dial application. If the Channel is mistakenly typed with two slashes (i.e Dial(PJSIP//xxxx…), the Dial applications fails (obviously), but it also kills the server.I put some code in my pbx_con..