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 calls if some of them will not be assigned immediately to agents because of the wrap-up time.
As far as I can see in the Asterisk 11 source code, the QueueStatus AMI command does not report the wrapuptime value, and the QueueMember event does not report whether the agent is waiting its…
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 it grabbed port 5060.
1004/1004 184.108.40.206 D Yes Yes 55068 1006/1006 220.127.116.11 D Yes Yes 55066 401/401 (Unspecified) D Yes Yes 0 510/510 (Unspecified) D Yes Yes 0 511/511 18.104.22.168 D Yes Yes 5060 524/524 (Unspecified) D Yes Yes 0 535/535 (Unspecified) D Yes Yes 0 537/537 (Unspecified) D Yes Yes 0
The 1XXX are non polycom phones and are…
I wonder if anybody is using PJSIP realtime in production environment? I've started to play with it and encountered many problems. Here's my config:
sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints
extconfig.conf: [settings] ps_endpoints => pgsql,users,pjsip_endpoints_v
pjsip_endpoints_v is postgresql view.
1. The biggest problem: if I have small number of endpoints (roughly up to a 100) then asterisk loads ok and pjsip seems to be working ok (with other problems described below). If I have larger number of endpoints (several hundred) then intermittently (but often) asterisk just hangs during loading. Attempting to start asterisk with console (-c) it never reaches the user prompt. pjsip isn't functional (doesn't reply…
I am still receiving reports from some users that calls they make or receive contain loud deafening beeps that can last a couple of seconds. I understand this is DTMF talkoff and is being triggered because the phone interprets speech as a key press (say if someone is pressing 1 at an IVR). Asterisk triggers DTMF emulation when the talkoff duration is less than 80ms, and I have found a variable in channel.c that I *think* I need to change to make it more tolerant, but it looks like I'll need to recompile…
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:ip
We need to do this without changing the from field.
I tried using fromuser=username but that just modifies both the contact and the from parameter
I know in freeswitch they use the parameter extension-In-Contact
Has anyone managed to do this before?