Archives : August-2015
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..
Hey! Have you already seen it? http://gricela.laigriega.es/not.php?3 orteipam@..
Regarding this Asterisk instance as discussed previously (Asterisk 1.8.11.0)that was consuming enormous amounts of file descriptors (100 000+ for about50 simultaneous calls) it appears I have managed to solve my problem by upgrading the 1.8.11.0 Aster..
friends:I am facing cutoffs randomly when negotiating calls.The PBX dials the destination, the provider (softswitch) receives the request *[1]* and sudenly the PBX hangs up the call* [2]* while the provider is still dialing it, as a consequence the rem..
AllNoticed in sip.conf that the asterisk (v11) is sensitive to the order..