Re: [asterisk-users] Detect Low Quality Calls - Realtime

2013-01-05 Thread joachim
On 5.1.2013 г. 03:37 ч., XBrian wrote: I can only detect calls as they hit our server, do the magic and based on latency, bandwidth and MOS (Meaning Opinion Score) - decide whether the call should be let through. I will accept all MOS values of 4.0 You are pretty much limited to

Re: [asterisk-users] Detect Low Quality Calls - Realtime

2013-01-05 Thread XBrian
Joachim, thanks for the reply - delay you can somewhat estimate prior to the call (with qualify for example) Pls be explicit. How do I use qualify to measure delay - The jitter / packetloss you can only figure out when the call is already up for a while. what would you use to measure jitter

Re: [asterisk-users] Moving User Agent To Remote Location

2013-01-05 Thread Nick Khamis
Hello Ishfaq, and Isrlgb, The canreinvite value for UA friend entries are set to no, and for the OpenSIPS peer entry it's set to yes. I do have esternip and localnet cid set in sip.conf. I did not want to start a new email, but part of my problem right now is that OpenSIPS is in charge of

Re: [asterisk-users] Verizon SIP trunking Field Trial

2013-01-05 Thread Logan Bibby
Does anyone have a good contact for their sales? I've attempted calling their Enterprise sales a few times and was just spun around in circles. Having a sales rep I can just call would be awesome. - Logan On Fri, Jan 4, 2013 at 1:36 PM, Michael L. Young myo...@acsacc.com wrote: - Original

Re: [asterisk-users] Verizon SIP trunking Field Trial

2013-01-05 Thread Robert-GMAIL
Good luck! Finding the right person at VZ has always been a beef of mine Sent from my iPhone 5 On Jan 5, 2013, at 11:12 AM, Logan Bibby lo...@keobi.com wrote: Does anyone have a good contact for their sales? I've attempted calling their Enterprise sales a few times and was just spun around

[asterisk-users] Limit registration concurrency per friend

2013-01-05 Thread XBrian
Can I restrictthe number of concurrent registrations per friend? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] Limit registration concurrency per friend

2013-01-05 Thread Leandro Dardini
2013/1/5 XBrian bobo...@yahoo.co.uk Can I restrictthe number of concurrent registrations per friend? Your question has no meaning. The registration is the way a peer says to asterisk which is the IP address and port to use to contact him. There can be just one registration active at time. If

Re: [asterisk-users] MaxCallBR Peer Setting

2013-01-05 Thread Larry Moore
On 4/01/2013 9:39 PM, XBrian wrote: Hi sip show peer 21342 gives me peer 21342's parameters. I am interested in the MaxCallBR line i.e. MaxCallBR: 384 kbps What exactly does this mean? Extracted from sample sip.conf file; ;maxcallbitrate=384 ; Maximum bitrate for

Re: [asterisk-users] Limit registration concurrency per friend

2013-01-05 Thread Eric Wieling
No. You may only have one registration per peer or friend. It cannot be changed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of XBrian Sent: Saturday, January 05, 2013 5:17 PM To:

[asterisk-users] Get CONNECTEDLINE info from other Asterisk system via IAX2

2013-01-05 Thread Chet W. Stevens
I have been racking my brain attempting to get the remote callerid information for calls made to extensions on another Asterisk system connected via IAX2 but nothing has worked. To clarify, I would like to display the number AND name on the calling phone when calling extensions on another

Re: [asterisk-users] Get CONNECTEDLINE info from other Asterisk system via IAX2

2013-01-05 Thread Kevin Larsen
Do you have sendrpid and trustrpid set to yes for those IAX2 connections?Sent from Lotus TravelerChet W. Stevens --- [asterisk-users] Get CONNECTEDLINE info from other Asterisk system via IAX2 --- From:Chet W. StevensToasterisk-users@lists.digium.comDate:Sat, Jan 5, 2013 7:55

Re: [asterisk-users] Detect Low Quality Calls - Realtime

2013-01-05 Thread Robert-GMAIL
Asterisk sip show peers lists the qualify value in ms (milliseconds). Please read up on this and the setting for it in sip.conf config file Sent from my iPhone 5 On Jan 5, 2013, at 5:30 AM, XBrian bobo...@yahoo.co.uk wrote: Joachim, thanks for the reply - delay you can somewhat estimate