Archives : November-2015
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AllAfter a Dial() I get:WARNING[7964][C-000075a8]: app_dial.c:2437 dial_exec_full: Unable to create channel of type SIP (cause 20 – Subscriber absent)if the subscriber is not registered.Is there a way from dialplan to know, *before* Dial(), if a destinat..
I have a somewhat confusing use case.We use a mobile voip app and our users connect to our PBX via a public IP of our firewall which port forwards to asterisk (TLS and SRTP ports). Works fine. Sometimes however, our users are also connected to our ..
Good day Asterisk users, If this is the wrong place to post this, my apologies. However, Im trying to see where I can get a patch for the res_musiconhold.so module. I have an issue where if someone is placed on hold, or is placed in a queue, after ..
Good day Asterisk users, If this is the wrong place to post this, my apologies. However, Im trying to see where I can get a patch for the res_musiconhold.so module. I have an issue where if someone is placed on hold, or is placed in a queue, after ..
HiWho to split a very long line in extension.conf?Something like: same => n,very long sequence \ of a binary expression inside a \ gotoif test..
everyone.Weve got a fairly large base of customers who use our Asterisk server for phone service in a virtual PBX kind of way, where the server is security hardened and exposed to the internet for them to connect to remotely with SIP and IAX. Its certai..
I have a problem where SIP calls from some providers are dropping at 15 minutes.The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server.Below,Client is the IP address of the clients host (running FPBX-2.8.1(1.8.20.0)OpenS..
I was wonder is there any way to custom the message on the call busy or no answer I actually get the error code from asterisk server on busy or no answer. Can I custom the text message or custom the message to sound ?Anyone have any idea could u ple..
Do you know if there are any e.164 country codes (CC) that provide a gateway from PSTN reachable numbers to URI (SIP) for their whole range?The only example I knowis iNum (+883 5100 country code) Each iNum number can be converted into an URI in the f..
I am looking for documentation support for enabling instant messaging between endpoints using Asterisk 13.1.0 and vanilla VoIP clients such as Zoiper. Where do I enable this support on the server side and does it need anything on the client side? I ..