[asterisk-users] DTMF talkoff beep (still)

Hi all, 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

[asterisk-users] PJSIP realtime: lots of problems

Hello, 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

Re: [asterisk-users] PJSIP realtime: lots of problems

Hello Michael On Thu, Oct 08, 2015 at 01:32:07PM -0400, Michael Ulitskiy wrote: > > extconfig.conf: > [settings] > ps_endpoints => pgsql,users,pjsip_endpoints_v Does it change anything if you use odbc instead of pgsql? I did some testing with chan_sip/pgsql and had much less problems when pgsql

Re: [asterisk-users] PJSIP realtime: lots of problems

On 15-10-08 02:32 PM, Michael Ulitskiy wrote: Hello, I wonder if anybody is using PJSIP realtime in production environment? They are. 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

Re: [asterisk-users] PJSIP realtime: lots of problems

On Thursday, October 08, 2015 08:02:19 PM Stefan Tichy wrote: > Hello Michael > > On Thu, Oct 08, 2015 at 01:32:07PM -0400, Michael Ulitskiy wrote: > > > > extconfig.conf: > > [settings] > > ps_endpoints => pgsql,users,pjsip_endpoints_v > > Does it change anything if you use odbc instead of

Re: [asterisk-users] DTMF talkoff beep (still)

On 9/10/2015, at 5:16 AM, Jamie Rees wrote: > > > I understand this is DTMF talkoff > > > My question is how do people running SIP phone systems mitigate against this? My personal answer to this question has been to completely avoid the use of any ATAs at all. Since

Re: [asterisk-users] Polycom phone registering

It sounds like you have problems with your firewall. Your 401 replies don't reach the phones. On Thursday, October 08, 2015 02:50:24 PM Jerry Geis wrote: > Do polycom phones not LIKE using something other than port 5060 ??? > > I have five of them behind a firewall and my asterisk server is

[asterisk-users] Detect queue agent in wrapuptime period through AMI

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

[asterisk-users] Polycom phone registering

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