Archives : May-2015
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We love our Digium phones and DPMA – but we really need it to work on our Realtime Platform. Otherwise we lose all the cool features and they are just standard SIP phones.Anyone working on a solution for this? Or anyone from Digium see this on th..
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I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP phones, which appear to be working correctly. I have a few external phones (Yealink SIP-T32G..
2015-04-30 17:45 GMT+02:00 Richard Mudgett :Simply adding a CONNECTEDLINE(num) statement made it !Thank you very much for..
I recently gave CONNECTEDLINE function a try on an Asterisk 11 setup with a couple of SIP phones.When a SIP phone dials an other one, with a CONNECTEDLINE statement in its dialplan, I noticed that Asterisk update callers information using a Remote-Party..
asterisk-users,Weve been having intermittent issues with chan_sip – it stops responding to cli requests, trying to reload chan_sip from cli doesnt seem to have any effect, initiated calls carry on for a short period, but no new SIPrequests are proces..
Using Asterisk 13.3.2 on CentOS7 and pjproject 2.4, I cant make asterisk try to send a register.I have configured my pjsip.conf similar to https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples#res_pjsipConfigurationExamples-ASIPtrunktoyourserviceprovider,includingoutboundregistratio..
guys i was trying to get working sessions-timer over PJSIP channel i was trying to see what is supported or not about this features on the new pjsip channel since chan_sip was kind of flexible on this , at the moment since wiki says pjsip support 4 mo..