Archives : August-2015
is it possible simultaneously use chan_sip and chan_pjsip?if yes, can you recommend settingsim thinking about- chan_sip – for sip hardphones/softphones(sip udp 5060)- chan_pjsip – ..
HelloI was wondering of it is possible to have Queue Agents with the same priority (penalty) but with a certain order ?So I have 20 Agents.Agent 1 till Agent 10 has penalty 1.Agent 11 till Agent 15 has penalty 2.(only contacted if 1 -> 10 are busy)Ag..
HiI need to set the numberof incoming calls to one, but the outgoing calls should be unlimited. I think the busylevel parameter is for it(incoming calls), but not works. My config is:cat sip.conf[general][template](!)qualify=yes cc_agent_policy=gene..
MarkusThanks for the reply, I have set those files previously as well…It seems the problem for me is on my CentOS 7 box that the Asterisk binary does not know that these are the limits, and imposes a 1024 open file limit count for some reason.It se..
[Aug 11 21:57:14] WARNING[1992] translate.c: no samples for alawtolin[Aug 11 21:57:14] WARNING[2005] translate.c: no samples for alawtolin[Aug 11 21:57:15] WARNING[2027] translate.c: no samples for alawtolin to all,I have an elastix box running aster..
I have been banging my head against the wall for weeks now on this one.I have a switch running NetBSD and Asterisk 11.19.0 although Ihave had this problem on older versions as well.I, and my users, can call out, we can receive calls, quality is excell..
TonyThanks for replying.I suspected something like that, though repeatedly runninglsof | wc -lAlways stays quite low – 100 000 open files, which is still 8 times less than the system maximum as confirmed by running ulimit -nI also note that this num..
we have an issue where after a couple of days, a few random phones will lose registration. I dont notice any particular pattern. Out of 200, only about 5-10 will be suffering at any given time and we wont know until the user complains they are not receiv..
I am starting a new project to develop a predictive dialler system.- Agents can start receiving calls from the queue if agent pressAvailable button on the browser which will unpause the queue on Asterisk.- About 100-150 concurrents calls on a Aster..
Im facing strange problem:Aasterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked. Call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chro..