Re: [asterisk-users] asterisk 11.14 - voicemail incorrect duration

Hi Stefan, Dominique Haeber dominique.hae...@xig.ch schrieb am Die, 27. Jan 08:55: I have looked at the time and talked for at least 4 seconds. In CLI log are 5-6 seconds visible between open to writing and Hang up. Nevertheless, Asterisk writes about two seconds. The value for

Re: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit salah.elharit...@gmail.com wrote: hello list, i have asterisk 11.15.0 and i have some trunks sip from my provider we have some ip phone astra 6731i each Ip-phone is configured with trunk and we call no ihave configured another trunk

Re: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

** THIS IS NOT WHERE YOUR REPLY BELONGS ** On Wednesday 25 Mar 2015, Salaheddine Elharit wrote: tnaks for your response but the number dialed exist and i can call this number when i configure the trunk directly in x-lite and i call call also this number from my cell phone . any help thanks

[asterisk-users] Call Quality Measuring

Hi everyone. We regularly get customers complaining about call quality issues. Most of the time it turns out to be their own broadband. Very occasionally server load. Does anyone have any advice or links to advice on measuring call quality? I’ve been playing around with “sip show channelstats”

Re: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

tnaks for your response but the number dialed exist and i can call this number when i configure the trunk directly in x-lite and i call call also this number from my cell phone . any help thanks and regards 2015-03-25 12:59 GMT+00:00 Matthew Jordan mjor...@digium.com: On Wed, Mar 25, 2015 at

[asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

hello list, i have asterisk 11.15.0 and i have some trunks sip from my provider we have some ip phone astra 6731i each Ip-phone is configured with trunk and we call no ihave configured another trunk from the same provider in my asterisk i can call all numbers just the numbers are configured

Re: [asterisk-users] Call Quality Measuring

On Wed, Mar 25, 2015 at 2:21 PM, Patrick Beaumont p.beaum...@hatsoffsoftware.co.uk wrote: Hi everyone. We regularly get customers complaining about call quality issues. Most of the time it turns out to be their own broadband. Very occasionally server load. Does anyone have any advice or

Re: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

thank you for your response but i think that the issue is related to the RTP because i can call all numbers with the same format when i call any number except 0033149xx i get the same adress from provider only with this number cnfigurerd in ip-phone in our network i get this error best

Re: [asterisk-users] Call Quality Measuring (Laszlo)

Have you tried using tcpdump? Then analyze the pcap on wireshark? Marlon Araujo On Mar 25, 2015, at 13:00, asterisk-users-requ...@lists.digium.com wrote: 1. Re: Call Quality Measuring (Laszlo) -- _ -- Bandwidth and

[asterisk-users] PJSIP configuration for Asterisk 13.1.0/SIP trunk outbound calling

Hello, I have had numerous issues with PJSIP outbound calling in Asterisk 13.1.0 and SIP.US SIP trunk. My Asterisk server is on EC2 and I have opened up the appropriate ports. The SIP clients can be anywhere on the Internet, including behind NATs. I am able to get to my Asterisk server's

[asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]

Howdy, I'm looking at enabling autopause on one of my queues where my queue members are bad about leaving their desks without pausing. The problem I see is that when the queue pauses an Member it doesn't jump into the dialplan to do so which means my handy device state and asterisk database

Re: [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]

asterisk-users-boun...@lists.digium.com wrote on 03/25/2015 01:38:26 PM: I'm looking at enabling autopause on one of my queues where my queue members are bad about leaving their desks without pausing. The problem I see is that when the queue pauses an Member it doesn't jump into the dialplan

Re: [asterisk-users] Asterisk 13. Writing call quality parameters to CDR. How?

On Thu, 19 Mar 2015 10:12:22 +0100 Marek Cervenka cerv...@fpf.slu.cz wrote: because of problems you are facing i decided to go way with second table CREATE TABLE `cdr_extended` ( `id` int(11) unsigned NOT NULL AUTO_INCREMENT, `uniqueid` varchar(32) NOT NULL DEFAULT '', `callid`

Re: [asterisk-users] Call Quality Measuring

Hi Patrick, try voipmon, there it's free and you can even track MOS. Markus Am 25.03.2015 um 14:21 schrieb Patrick Beaumont: Hi everyone. We regularly get customers complaining about call quality issues. Most of the time it turns out to be their own broadband. Very occasionally server load.

Re: [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]

Thank you Kevin, I've looked at your solution and while I agree it's not ideal it does appear to be something that might work for me. I'll see if I can maybe backport the QUEUE_MEMBER stuff to 1.8 from 11. I'm also exploring an idea with a co-worker of using an AMI listener that will fire off

Re: [asterisk-users] Call Quality Measuring

Hi Markus, Sounds interesting to me too... However my google-fu is letting me down today - I found VOIPmonitor at Sourceforge http://sourceforge.net/projects/voipmonitor/ but this looks like you'll need a license. Any chance you have a link to voipmon? Cheers .. Brendan Ord OntheNet -