Archives : October-2015
Home » 2015 (paged 14)
Hello.Continue to move from chan_sip to res_pjsip. For the work of my algorithms is very important to know the IP address of all trunks and endpoints (phones).In the case of chan_sip, I used PeerStatus AMI event through which was received the fact..
Is there a way to use AMI to detect whether an agent that appears to be free is in its wrap-up-time period?I am using AMI to query the queue status and its members, in order to generate calls directed to the queue, and I do not want to originate ca..
Do polycom phones not LIKE using something other than port 5060 ???I have five of them behind a firewall and my asterisk server is remote. Other devices are registering just fine, just not my polycom phones.Today I notices that ONE registered, but..
I wonder if anybody is using PJSIP realtime in production environment?Ive started to play with it and encountered many problems. Heres my config:sorcery.conf:[res_pjsip]endpoint=realtime,ps_endpointsextconfig.conf:[settings]ps_endpoints => pgsql,users,pjsip_endpoints_vpjsip_endpoint..
——=_NextPart_001_005A_01D101ED.084A77B0Content-Type: text/plain; charset=us-asciiContent-Transfer-Encoding: 7bitall, I am still receiving reports from some users that calls they make or receive contain loud deafening beeps that can last a cou..
Guys Does anyone know of a way I can change the contact field in the sip invite to display sip:username:ip instead of sip:did:ipWe need to do this without changing the from field.I tried using fromuser=usernamebut that just modifies both the cont..
I have the following code that operates when a channel is hung-up:[record-hangupcause]exten => 1,n,Set(CDR(hangupcause)=${HANGUPCAUSE})exten => s,n,Return()Before the dial a hangup handler is registered:Set(CHANNEL(hangup_handler_push)=record-hangupcause,s,1)..
Ive started to play with PJSIP and got stuck at the following problem. I need to retrieve SIP Call-ID associated with PJSIP channel. For inbound channel I can use ${PJSIP_HEADER(read,Call-ID)}, but that doesnt work for outbound channel even in pre-d..
Hello. Do I understand correctly that the current implementation res_pjsip does not support ZRTP? http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html Nothing has changed since 2013? P.S. I greatly regret that moved from chan_..
I have 4 Polycom phones behind a firewall. I cannot get them to dial as is says URL Disabled so its not registered to my server.I have tried nat=no, nat=force_rport,comedia still no go.I have sip.conf set for bindaddr= my server IP.I have uncommen..