Archives : August-2015
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I have what I would think would be a common situation: I run asterisk at home simply as a land line. I started a new job working remotely and they gave me a SIP account with user name, domain, and proxy. Ive never had to deal with sip domains befo..
I am running Asterisk 13.5.0.I have the Transfer working when using the chan_sip support. However, when I try to perform a Transfer using pjsip, it is failing.The one difference I am seeing in the SIP trace is chan_sip automatically sends the Referred-By.PJ..
Greetings everyone, I am attempting to adjust the volume of a call using Set(VOLUME) in my extensions.conf file. I am finding that Set(VOLUME(TX)=x) and Set(VOLUME(RX)=y) have no discernable effect on my endpoints (Snom 300 IP phones). I have tried sett..
Everybody,in past times I used macros but since a while they are deprecated. So I replaced my macros with subroutines. In most cases this is really no problem.But in some rare cases I miss the macro channel variables (e.g. ${MACRO_CONTEXT}). https://wiki.asterisk.org/wiki/display/AST/Dialplan+Macros+Channel+Variable..
Can anyone advise on the status of SRV lookups in Asterisk 11?(specifically 11.17.1)Is there any difference given how the Dial is done, and how supported are weights and priorities?Thanks i..
all,i need to test how many calls can withstand an Asterisk server.Do you know any good tools to strain the server?At best, there are scripts that I can run on a Linux server.I thank you for your tipsSincerely Domini..
Id like to use a feature code for stopping recordings. Things are quite easy when the call is received from the outside or just dialed from inside to outside, but it can go really crazy when there are blind and attended transfer going on. It ends I d..
all,Im using Kamailio + Asterisk 13 (PJSIP), where Kamailio (using rtpengine)acts as the registrar and forwards all calls to Asterisk.This works fine when using udp / tcp and RTP. When switching to TLS/SRTP, the call is set up correctly, however, I ..
Im having what seems like a weird issue connecting Asterisk 13 (FreePBX 12) to a Cisco 2811 router via a chan_sip trunk.Whenever I try dialling out via this trunk, something appends @CUBE onto the end of the dialled number, as per the following examples;Aster..
If we have a shared RealTime database for sip registration of multiple Asterisk servers, is there a way to realize which Asterisk server registered sip phones ?RegardsM..