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
, 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
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
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
: Asterisk Users Mailing List - Non-Commercial Discussion
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 sip cause and sip response
> from o
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
Ämne: [asterisk-users] Asterisk cra
,
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
Ämne: Re: [asterisk-users] Asterisk crash on core show channel
Den här
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
Ämne: Re: [asterisk-users] Asterisk crash on core show channel
Den här
couldn't find any direct change that could fix
this
Thanks,
Kind regards,
Patrick Wakano
On 21 February 2018 at 20:29, Marcus Kvarsell
mailto:marcus.kvars...@fogwise.se>> wrote:
Hello, i found upgrading to asterisk 15 helped.
Från:
asterisk-users-boun...@lists.digium.com<m
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> |
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
Ämne: [asterisk-users] Test
Testing, 1, 2, 3.
--
Matthew Fredrickson
Digium,
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> |
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: Asteri
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,
c1=0x7faea47
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!
Regard
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)
[2011-0
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
] 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 s
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 rec
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