[asterisk-users] Voip realtime analysis tools

2018-03-22 Thread Marcus Kvarsell
3. Customizable by design 4. experienced support With kind regards, Marcus Kvarsell [www.fogwise.se]<http://www.fogwise.se/> marcus.kvars...@fogwise.se<mailto:marcus.kvars...@fogwise.se> | www.fogwise.se<http://www.fogwise.se> |

Re: [asterisk-users] Test

2018-03-22 Thread Marcus Kvarsell
Yes it is working  -Ursprungligt meddelande- Från: asterisk-users-boun...@lists.digium.com För Matt Fredrickson Skickat: den 21 mars 2018 03:46 Till: Asterisk Users Mailing List - Non-Commercial Discussion

[asterisk-users] Asterisk 15 autocreate peer

2018-02-28 Thread Marcus Kvarsell
Hi, Is it possible to use autocreate peer with webrtc in asterisk 15? With kind regards, Marcus Kvarsell [www.fogwise.se]<http://www.fogwise.se/> marcus.kvars...@fogwise.se<mailto:marcus.kvars...@fogwise.se> | www.fogwise.se<http://www.fogwise.se> |

Re: [asterisk-users] Asterisk crash on core show channel

2018-02-22 Thread Marcus Kvarsell
ething related? From the release notes, I couldn't find any direct change that could fix this Thanks, Kind regards, Patrick Wakano On 21 February 2018 at 20:29, Marcus Kvarsell <marcus.kvars...@fogwise.se<mailto:marcus.kvars...@fogwise.se>> wrote: Hello, i found upgrading to asteris

Re: [asterisk-users] Asterisk crash on core show channel

2018-02-22 Thread Marcus Kvarsell
Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Marcus Kvarsell Skickat: den 22 februari 2018 09:14 Till: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Ämne: Re: [asterisk-users] Asterisk

Re: [asterisk-users] Asterisk crash on core show channel

2018-02-22 Thread Marcus Kvarsell
, Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Marcus Kvarsell Skickat: den 22 februari 2018 08:29 Till: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Ämne: Re: [asterisk-users] Asterisk

Re: [asterisk-users] Asterisk crash on core show channel

2018-02-21 Thread Marcus Kvarsell
Hello, i found upgrading to asterisk 15 helped. Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Patrick Wakano Skickat: den 21 februari 2018 04:29 Till: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [asterisk-users] Sip cause and response codes in dialplan

2018-02-20 Thread Marcus Kvarsell
: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Ämne: Re: [asterisk-users] Sip cause and response codes in dialplan On Tuesday 20 February 2018 at 14:09:05, Marcus Kvarsell wrote: > Hi, > > I am experimenting with getting hold of the si

[asterisk-users] Sip cause and response codes in dialplan

2018-02-20 Thread Marcus Kvarsell
reason codes in the dialplan which i now am unable to do. Any help appreciated. [Beskrivning: Fogwise - logotype] Marcus Kvarsell phone: +46766350384 e-mail: mar...@fogwise.se url: http://www.fogwise.se Like us on facebook: https://www.facebook.com/WiseDialer<https://www.facebook.com/WiseDia

Re: [asterisk-users] # converts to %23

2018-02-20 Thread Marcus Kvarsell
Ämne: Re: [asterisk-users] # converts to %23 On Mon, Feb 19, 2018, at 9:56 AM, Marcus Kvarsell wrote: > It is in the To: Header. Encoding is supposed to be done in that case. This became the default in a later version, specifically the "pedantic" option in chan_sip was changed to de

Re: [asterisk-users] # converts to %23

2018-02-19 Thread Marcus Kvarsell
, Feb 19, 2018, at 4:24 AM, Marcus Kvarsell wrote: > Hello, > > I have a broblem in asterisk 15 where an ami originate suddenly > converts > 58#+46435345534 to 58%23+46435345534. This happenend when upgrading > asterisk 1.8 to 15. Could anyone help me with how to resolve this i

[asterisk-users] # converts to %23

2018-02-19 Thread Marcus Kvarsell
Hello, I have a broblem in asterisk 15 where an ami originate suddenly converts 58#+46435345534 to 58%23+46435345534. This happenend when upgrading asterisk 1.8 to 15. Could anyone help me with how to resolve this issue? Regards / Marcus [Beskrivning: Fogwise - logotype] Marcus Kvarsell phone

[asterisk-users] Segfault Asterisk 1.4.44 in wmvare ESXi 5.5

2014-09-09 Thread Marcus Kvarsell
Hi, Im getting daily segfaults when running 40-100 cuncurrent calls in G729 passthrough mode. Any thoughs on why this is happening is most appreciated. #0 0x003cd773356f in __strlen_sse42 () from /lib64/libc.so.6 #1 0x0043b352 in update_bridgepeer (c0=0x7faeac06a320,

Re: [asterisk-users] Enterprise VoIP Trunk

2014-03-06 Thread Marcus Kvarsell
Hi there, we have this possibility, we are a official telephone operator in Sweden. We have low price and premium quality. / Marcus From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N Sent: den 5 mars 2014 19:34 To:

Re: [asterisk-users] stream rtp from asterisk

2011-07-05 Thread Marcus Kvarsell
On 2011-07-04 15:07, Marcus Kvarsell wrote: Sending the rtp-data to external server. One example which I have not gotten to work is this below: http://oreka.sourceforge.net/ September 02, 2009: Asterisk interception via Xorcom Asterisk patch Added support for recording of Asterisk voice calls

[asterisk-users] stream rtp from asterisk

2011-07-04 Thread Marcus Kvarsell
Hi! Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Regards / Marcus -- _ -- Bandwidth and

Re: [asterisk-users] stream rtp from asterisk

2011-07-04 Thread Marcus Kvarsell
] För Alex Balashov Skickat: den 4 juli 2011 14:49 Till: asterisk-users@lists.digium.com Ämne: Re: [asterisk-users] stream rtp from asterisk On 07/04/2011 06:58 AM, Marcus Kvarsell wrote: Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp

[asterisk-users] xorcom asterisk patch for sending rtp stream to remote oreka server

2011-06-21 Thread Marcus Kvarsell
Hello! Im trying to setup the xorcom asterisk patch to be able of sending rtp setrema to an oreka voip recording server but I get error messages. [2011-06-20 15:43:07] VERBOSE[22529] logger.c: [2011-06-20 15:43:07] -- Reloading module 'res_monitor.so' (Call Monitoring Resource)

[asterisk-users] check if not human

2009-02-19 Thread Marcus Kvarsell
I am looking for someone that could share their code for this function: Outgoing call - macro that checks if line is (not human) or machine, fax, busy, subscriber problem and other fault tones - if human connect to agent else hangup and write status to cdr. Need help with this! Regards