Re: [asterisk-users] Detect Low Quality Calls - Realtime
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 measuring the delay and the jitter. The delay you can somewhat estimate prior to the call (with qualify for example). The jitter / packetloss you can only figure out when the call is already up for a while. (e.g. you might have no issues the first minute, but maybe packet loss will come in bursts after a minute). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect Low Quality Calls - Realtime
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 / packetloss in real time? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Moving User Agent To Remote Location
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 performing the AUTH and REGISTER. This is fine for peers with static host definition, but not for the dynamic ones. Is it possible to have the fullcontact realtime info in sip_buddies populated upon initial INVITE? This is my first problem right now. After that will come RTP, and Codec issues... PS I have seen fullcontact info get populated with the correctly in the past, just can't get it to do it every time Thanks for your help!!! Nick. On 1/4/13, isr...@gmail.com isr...@gmail.com wrote: Did you set externip and localnet in your sip conf ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon SIP trunking Field Trial
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 Message - From: Matthew J. Roth mr...@imminc.com At least Verizon maintains a consistent customer experience. ; ) Overall, we've found the service to be reliable and stable, but when there are problems or changes needed you're dealing with Verizon and the w...h...e...e...l...s..t...u...r...n..s...l...o...w...l...y. Haha... that is funny... it is sooo true. Well, you are right. Once it is working, it is usually pretty stable. Just a pain in the butt when things are not working. Hopefully we can get through the Field Trial and that is all I have to worry about for a while. Thanks Matthew for all the encouragement as I go down this temporary (I hope) unpleasant path. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Logan Logan Bibby, CEO Ke*o*bi Communications Tuscaloosa, Alabama -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verizon SIP trunking Field Trial
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 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 Message - From: Matthew J. Roth mr...@imminc.com At least Verizon maintains a consistent customer experience. ; ) Overall, we've found the service to be reliable and stable, but when there are problems or changes needed you're dealing with Verizon and the w...h...e...e...l...s..t...u...r...n..s...l...o...w...l...y. Haha... that is funny... it is sooo true. Well, you are right. Once it is working, it is usually pretty stable. Just a pain in the butt when things are not working. Hopefully we can get through the Field Trial and that is all I have to worry about for a while. Thanks Matthew for all the encouragement as I go down this temporary (I hope) unpleasant path. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Logan Logan Bibby, CEO Keobi Communications Tuscaloosa, Alabama -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Limit registration concurrency per friend
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: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit registration concurrency per friend
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 two or more peers attempt to register at the same time, the last one is the only one working. Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MaxCallBR Peer Setting
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 video calls (default 384 kb/s) ; Videosupport and maxcallbitrate is settable ; for peers and users as well Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Limit registration concurrency per friend
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@lists.digium.com Subject: [asterisk-users] Limit registration concurrency per friend 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: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Get CONNECTEDLINE info from other Asterisk system via IAX2
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 Asterisk system. I seem to be able to 'send' all the information I want to the system I am calling but cannot 'return' or lookup any information. I can use CALLERID and IAXVAR to 'send' information just fine. Is this as expected or does anyone have any ideas? I am using Digium D40s and D70s and Asterisk 1.8.11-cert10. Your help is appreciated. Thank you. Chet Stevens -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get CONNECTEDLINE info from other Asterisk system via IAX2
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 PMSubject[asterisk-users] Get CONNECTEDLINE info from other Asterisk system via IAX2 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 Asterisk system. I seem to be able to 'send' all the information I want to the system I am calling but cannot 'return' or lookup any information. I can use CALLERID and IAXVAR to 'send' information just fine. Is this as expected or does anyone have any ideas? I am using Digium D40s and D70s and Asterisk 1.8.11-cert10. Your help is appreciated. Thank you. Chet Stevens -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detect Low Quality Calls - Realtime
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 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 / packetloss in real time? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
