hi,
do you have someone example of
http://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/
in node.js asterisk-ari ?
thanks
Marek
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Hello,
Yes. When I today understood to set rtcp_mux=yes, at least Chrome (60.0
beta) worked (quickly tested) as expected.
I'm sure that some day dtls_rekey can be set to the other value than 0
as well with Chrome.
Best regards,
Teijo
10.4.2017, 16.57, Matt Fredrickson kirjoitti:
On Sat,
ts.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Friday, April 21, 2017 10:18 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)
Hello
you mean while placin
Hi all,
I'm working on migrating all of my servers to store voicemail in a mysql
database via
odbc.
I've got a development server that I can reconfigure and test at will. When
it's
configured to store vm on the file system, it seems to be rock solid.
However, when I ONLY change it to store
On Tue, Jun 20, 2017, at 09:50 AM, Jason TOMLINSON wrote:
> Hi, I've put the sip output here : https://pastebin.com/W7M4zxHA
> Thanks
It certainly shows it happening but nothing stands out as to why. Looks
like a bug, go ahead and file one on the issue tracker[1].
[1] https://issues.asterisk.org/
digium.com
Objet : Re: [asterisk-users] Asterisk 13 attended transfer alcatel
On Fri, Jun 9, 2017, at 04:59 AM, Jason TOMLINSON wrote:
> Hello,
>
> Since upgrading from asterisk 11 to asterisk 13 (I have tested up to
> the latest 13.16.0 release), we have a problem with attended transfer
On 06/18/2017 at 12:11 PM, Joshua Colp wrote:
> On Sun, Jun 18, 2017, at 06:00 AM, Michael Maier wrote:
>> Hello!
>>
>> unchanged asterisk crashes during udptl / t.38 negotiation with telekom
>> - they do not support t.38 / udptl.
>
> All Asterisk issues need to go through the issue tracker[1]. In
On Sun, Jun 18, 2017, at 06:00 AM, Michael Maier wrote:
> Hello!
>
> unchanged asterisk crashes during udptl / t.38 negotiation with telekom
> - they do not support t.38 / udptl.
All Asterisk issues need to go through the issue tracker[1]. In this
case we'd also need to see the full SIP debug so
Hello!
unchanged asterisk crashes during udptl / t.38 negotiation with telekom
- they do not support t.38 / udptl.
In detail:
fax client -> asterisk -> telekom -> easybell -> asterisk -> fax server
Fax server sends t.38 reinvite via asterisk to easybell.
Session Description Protocol Versio
On 06/17/2017 at 02:18 PM, Michael Maier wrote:
> On 06/16/2017 at 04:00 PM, Joshua Colp wrote:
>> On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote:
>>
>>
>>
>>>
>>> t38modem and asterisk are using
>>>
>>> m=image 35622 udptl t38
>>>^
>>>
>>> Provider uses
>>>
>>> m=image
On Sat, Jun 17, 2017, at 09:18 AM, Michael Maier wrote:
>
> I can proof, that UDPTL vs. udptl is the problem. After "fixing"
> asterisk and opal both using UDPTL, the negotiation works flawlessly.
> See attached patches.
>
> Sending t38 faxes internally works fine. Externally via provider gets
On 06/16/2017 at 04:00 PM, Joshua Colp wrote:
> On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote:
>
>
>
>>
>> t38modem and asterisk are using
>>
>> m=image 35622 udptl t38
>>^
>>
>> Provider uses
>>
>> m=image 35622 UDPTL t38
>>^
>>
>> Could this be
On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote:
>
> t38modem and asterisk are using
>
> m=image 35622 udptl t38
>^
>
> Provider uses
>
> m=image 35622 UDPTL t38
>^
>
> Could this be a problem? If I'm sending internal only, it's always
> lower
Am 16.06.2017 um 11:12 schrieb Joshua Colp:
On Fri, Jun 16, 2017, at 02:13 AM, Michael Maier wrote:
Has anybody any idea why asterisk drops the media stream in the 200 OK?
The channel has been T38_ENABLED before! Or is it necessary to add more
debug code? Who does the negotiating?
Only asterisk
On Fri, Jun 16, 2017, at 02:13 AM, Michael Maier wrote:
> Has anybody any idea why asterisk drops the media stream in the 200 OK?
> The channel has been T38_ENABLED before! Or is it necessary to add more
> debug code? Who does the negotiating?
> Only asterisk or is pjsip doing some parts, too?
On 06/15/2017 at 08:15 AM Michael Maier wrote:
> On 06/14/2017 at 10:17 PM, Joshua Colp wrote:
>> On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote:
>>
>>
>>
>>>
>>> I can now say, that asterisk / pjsip seams to work *mostly* as expected.
>>> Just one exception - and that's the package in que
On Thu, Jun 15, 2017 at 12:11:36PM +0200, Benoit Panizzon wrote:
> Or does anyone have an idea over what the asterisk is stumbling?
What if you set the maxdata in asterisk to a value lower than the other
side? e.g. sip.conf:
t38pt_udptl = yes,fec,maxdatagram=400
--
_
Hi all
I know, a fairly old asterisk installation.
Is there any way to debug T.38 handshaking issues?
We have a C3 Voice Switch with link to the Asterisk server.
I see this Dialogue:
C3 => Asterisk
=> Invite g711
<= 200OK
C3 detects Fax and send re-invite
=> Invite T.38
Version:0
RateManagem
On 06/14/2017 at 10:17 PM, Joshua Colp wrote:
> On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote:
>
>
>
>>
>> I can now say, that asterisk / pjsip seams to work *mostly* as expected.
>> Just one exception - and that's the package in question, which can't be
>> seen in tcpdump.
>>
>> I exte
On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote:
>
> I can now say, that asterisk / pjsip seams to work *mostly* as expected.
> Just one exception - and that's the package in question, which can't be
> seen in tcpdump.
>
> I extended the above patch by adding the info at the last output
On 06/14/2017 at 05:53 PM Joshua Colp wrote:
> On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote:
>
>
>
>>
>> I added this patch to see, if really all packages are are freed after
>> they have been processed:
>>
>> --- b/res/res_pjsip/pjsip_distributor.c 2017-05-30 19:44:16.0
>> +02
On Wed, Jun 14, 2017, at 12:47 PM, Michael Maier wrote:
>
> I added this patch to see, if really all packages are are freed after
> they have been processed:
>
> --- b/res/res_pjsip/pjsip_distributor.c 2017-05-30 19:44:16.0
> +0200
> +++ a/res/res_pjsip/pjsip_distributor.c 2017-06-13 2
On 06/11/2017 at 06:51 PM Joshua Colp wrote:
> On Sun, Jun 11, 2017, at 01:47 PM, Joshua Colp wrote:
>> The distributor is in res/res_pjsip/pjsip_distributor.c, the distributor
>> function being the entry point. That function returning PJ_TRUE
>> indicates to PJSIP that it has been handled and no s
On 06/11/2017 at 11:35 PM Daniel Tryba wrote:
> On Sun, Jun 11, 2017 at 01:16:10PM +0200, Michael Maier wrote:
>> Let's go into details:
>> I'm starting at the point where asterisk / fax client receives the T.38
>> reininvite from the server from the provider 195.185.37.60:5060 for the
>> fax clien
On 06/11/2017 at 04:34 PM Michael Maier wrote:
> On 06/11/2017 at 01:29 PM Joshua Colp wrote:
>> On Sun, Jun 11, 2017, at 08:16 AM, Michael Maier wrote:
>>
>>
>>
>>> I did some further investigations and fixed a local problem. Now,
>>> asterisk is able to successfully activate T.38 - unfortunate
On Sun, Jun 11, 2017 at 01:16:10PM +0200, Michael Maier wrote:
> Let's go into details:
> I'm starting at the point where asterisk / fax client receives the T.38
> reininvite from the server from the provider 195.185.37.60:5060 for the
> fax client (extension 91):
I'm running Asterisk 11 on my fax
On Sun, Jun 11, 2017, at 01:47 PM, Joshua Colp wrote:
> On Sun, Jun 11, 2017, at 01:31 PM, Michael Maier wrote:
> > On 06/11/2017 at 04:39 PM Joshua Colp wrote:
> > > On Sun, Jun 11, 2017, at 11:34 AM, Michael Maier wrote:
> > >
> > >
> > >
> > >>>
> > >>> PJSIP uses a dispatch model. The reques
On Sun, Jun 11, 2017, at 01:31 PM, Michael Maier wrote:
> On 06/11/2017 at 04:39 PM Joshua Colp wrote:
> > On Sun, Jun 11, 2017, at 11:34 AM, Michael Maier wrote:
> >
> >
> >
> >>>
> >>> PJSIP uses a dispatch model. The request is queued up, acted on, and
> >>> then that's it. The act of acting
On 06/11/2017 at 04:39 PM Joshua Colp wrote:
> On Sun, Jun 11, 2017, at 11:34 AM, Michael Maier wrote:
>
>
>
>>>
>>> PJSIP uses a dispatch model. The request is queued up, acted on, and
>>> then that's it. The act of acting on it removes it from the queue.
>>
>> That's the *expected* behavior ..
On Sun, Jun 11, 2017, at 11:34 AM, Michael Maier wrote:
> >
> > PJSIP uses a dispatch model. The request is queued up, acted on, and
> > then that's it. The act of acting on it removes it from the queue.
>
> That's the *expected* behavior ... . I rechecked again and again. All
> existing tcpdu
On 06/11/2017 at 01:29 PM Joshua Colp wrote:
> On Sun, Jun 11, 2017, at 08:16 AM, Michael Maier wrote:
>
>
>
>> I did some further investigations and fixed a local problem. Now,
>> asterisk is able to successfully activate T.38 - unfortunately just for
>> very shot time because of a phantom pac
On Sun, Jun 11, 2017, at 08:16 AM, Michael Maier wrote:
> I did some further investigations and fixed a local problem. Now,
> asterisk is able to successfully activate T.38 - unfortunately just for
> very shot time because of a phantom package it receives!
What was the local problem?
> Let's
On 06/05/2017 at 09:32 PM Joshua Colp wrote:
> On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote:
>> On 06/05/2017 at 06:29 PM, Joshua Colp wrote:
>>> On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote:
Do you have any idea where to start to look at? Adding additional output
i
How are both machines connected to each other ?
Through a SIP trunk ? A TDM one ?
2017-06-09 9:59 GMT+02:00 Jason TOMLINSON :
> Hello,
>
>
>
> Since upgrading from asterisk 11 to asterisk 13 (I have tested up to the
> latest 13.16.0 release), we have a problem with attended transfers to an
> alca
On Fri, Jun 9, 2017, at 04:59 AM, Jason TOMLINSON wrote:
> Hello,
>
> Since upgrading from asterisk 11 to asterisk 13 (I have tested up to the
> latest 13.16.0 release), we have a problem with attended transfers to an
> alcatel pbx in which the call being transferred still has music on hold
> even
Hello,
Since upgrading from asterisk 11 to asterisk 13 (I have tested up to the latest
13.16.0 release), we have a problem with attended transfers to an alcatel pbx
in which the call being transferred still has music on hold even after the
transfer has completed.
Is this a known issue? Any new
And it is worst (and that could be the reason of the error).
127.0.0.1 is configured in 2 interfaces (lo and venet0), just with
different network masks.
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.c
Well, based on the config that you sent, your server just have the
localhost IP (127.0.0.1)
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On 6 June 2017 at 18:54, andre castro wr
I am using version: 14.5.0
No, Im not using Dundi.
Can you a bit more informative when you say I "need to configure the IPs
in your server"?
thanks!
a
On 06/06/2017 07:47 PM, Marcelo Terres wrote:
> I think you need to configure the IPs in your server. You just have
> localhost...
> Marcelo H. Ter
I think you need to configure the IPs in your server. You just have localhost...
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On 6 June 2017 at 16:27, andre castro wrote:
> Than
Looks like it comes com pbx_dundi.c.
Why are you using dundi?
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On 6 June 2017 at 18:43, Marcelo Terres wrote:
> Which Aste
Which Asterisk version are you using?
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On 6 June 2017 at 18:32, andre castro wrote:
> Any ideas.
> After configuring port forwardin
Any ideas.
After configuring port forwarding on the server (machine making nat) to
forward connections originated from external clients to the machine
running asterisk, as explained in
https://www.voip-info.org/wiki/view/port+forwarding
My peers were unable to register.
And When running Asterisk
Try to use the echo app. If you can listen your echo, so it is
something in the network.
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On 6 June 2017 at 14:18, andre cas
Thanks Anthony.
I did it on the server, according to
https://www.voip-info.org/wiki/view/port+forwarding
However after doing it, when running Asterisk I get the following message
sudo asterisk -vvr
No ethernet interface found for seeding global EID. You will have to set
it manually.
Unable to
On Tuesday 06 June 2017 16:57:07 andre castro wrote:
> On 06/06/2017 04:36 PM, Antony Stone wrote:
> >
> > Tell us about your networking arrangement - are both phones and the
> > Asterisk machine on the same network?
>
> Nop. They are in 2 different networks. The phones in one and the
> Asterisk
Thank you Daniel for pointing out the errors and debug option. Both
fixed and on.
It made no difference. There are no errors printed and still no sound on
ppers
Now to Antony questions:
On 06/06/2017 04:36 PM, Antony Stone wrote:
> On Tuesday 06 June 2017 15:18:32 andre castro wrote:
>
>> I just
Le 06/06/2017 à 16:25, Daniel Tryba a écrit :
On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote:
extensions.conf:
[home]
exten = 102,1,Answer()
same = n,Wait(1)
If this is copy and paste, then your dialplan is broken (= should be =>)
Well, no. = or => are the same.
--
Daniel
--
On Tuesday 06 June 2017 15:18:32 andre castro wrote:
> I just installed asterisk in a debian server.
> All seems to be running fine, but the audio sent by the server.
> But I hear nothing at the peer's end.
>
> When one peer calls another, sound comes through just fine.
Tell us about your netwo
On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote:
> extensions.conf:
> [home]
> exten = 102,1,Answer()
> same = n,Wait(1)
If this is copy and paste, then your dialplan is broken (= should be =>)
But to debug, enable logging (core set verbose 5), when needed debugging
(core set debug
hello folks,
this might be a simple question...
I just installed asterisk in a debian server.
All seems to be running fine, but the audio sent by the server.
If I have one of my registered peers call and extension (102) that plays
back audio (extension.conf and sip.conf coffee-pasted below), Aster
On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote:
> On 06/05/2017 at 06:29 PM, Joshua Colp wrote:
> > On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote:
> >>
> >> Do you have any idea where to start to look at? Adding additional output
> >> in the source code? Which functions could be inte
On 06/05/2017 at 06:29 PM, Joshua Colp wrote:
> On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote:
>>
>> Do you have any idea where to start to look at? Adding additional output
>> in the source code? Which functions could be interesting? I may add own
>> debug code to see why things are happen
On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote:
>
> Do you have any idea where to start to look at? Adding additional output
> in the source code? Which functions could be interesting? I may add own
> debug code to see why things are happening as they happen here.
The logic for T.38 negoti
On 06/05/2017 at 06:10 PM, Joshua Colp wrote:
> On Mon, Jun 5, 2017, at 12:00 PM, Joshua Colp wrote:
>> On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote:
>>> On 06/05/2017 at 11:30 AM, Joshua Colp wrote:
On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote:
> On 06/04/2017 at 01:41 P
On Mon, Jun 5, 2017, at 12:00 PM, Joshua Colp wrote:
> On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote:
> > On 06/05/2017 at 11:30 AM, Joshua Colp wrote:
> > > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote:
> > >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote:
> > >>> Just
On 06/05/2017 at 05:00 PM, Joshua Colp wrote:
> On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote:
>> On 06/05/2017 at 11:30 AM, Joshua Colp wrote:
>>> On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote:
On 06/04/2017 at 01:41 PM Telium Technical Support wrote:
> Just a guess (witho
On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote:
> On 06/05/2017 at 11:30 AM, Joshua Colp wrote:
> > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote:
> >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote:
> >>> Just a guess (without knowing about your network), but are the two e
On 06/05/2017 at 11:30 AM, Joshua Colp wrote:
> On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote:
>> On 06/04/2017 at 01:41 PM Telium Technical Support wrote:
>>> Just a guess (without knowing about your network), but are the two ends
>>> points on public networks and visible to one another?
On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote:
> On 06/04/2017 at 01:41 PM Telium Technical Support wrote:
> > Just a guess (without knowing about your network), but are the two ends
> > points on public networks and visible to one another? If not the reinvite
> > may be passing an interna
On 06/04/2017 at 01:41 PM Telium Technical Support wrote:
> Just a guess (without knowing about your network), but are the two ends
> points on public networks and visible to one another? If not the reinvite
> may be passing an internal (nat'ed) address to the other and the connection
> will fail.
users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Maier
Sent: Sunday, June 4, 2017 3:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207
t38_automatic_r
Hello!
I'm still trying to get a working t.38 configuration w/ pjsip.
I'm now able to send t.38 faxes to my own extension:
hylafax -> t38modem -> extension -> extension -> t38modem -> hylafax.
The fax is sent by t38modem. The receiving part of t38modem accepts the
call, sends ReInvite for t.3
The Asterisk Development Team would like to announce the release of Asterisk
14.5.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 14.5.0 resolves several issues reported by the
community and would have not been po
The Asterisk Development Team would like to announce the release of Asterisk
13.16.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.16.0 resolves several issues reported by the
community and would have not been
Will do, thanks for the confirmation!
Chris.
On 2017-05-30 18:08, Joshua Colp wrote:
> On Tue, May 30, 2017, at 01:55 PM, Christopher van de Sande wrote:
>
>> Cool, I've attached 2 sip trace examples, 14.3.1 and 14.4.1. Both are
>> using identical configurations.
>>
>> The biggest differenc
On Tue, May 30, 2017, at 01:55 PM, Christopher van de Sande wrote:
> Cool, I've attached 2 sip trace examples, 14.3.1 and 14.4.1. Both are
> using identical configurations.
>
> The biggest difference I can see happens right after the UPDATE message
> from MicroSIP.
>
> The SDP headers on 14.4.1
On Tue, May 30, 2017, at 01:07 PM, Christopher van de Sande wrote:
> Hi first post, so hope I'm not violating protocol.
>
> Been using Asterisk as home phone and hobby use for nearly 10 years. I
> love this project.
>
> Anyway, would someone mind verifying my pjsip.conf ? This seems to work
>
Hi first post, so hope I'm not violating protocol.
Been using Asterisk as home phone and hobby use for nearly 10 years. I
love this project.
Anyway, would someone mind verifying my pjsip.conf ? This seems to work
well for 14.3.1 but I get no rtp into my natted Linphone when I upgrade
to 14.4.1
On Tue, May 16, 2017 at 3:00 PM, Tiago Ferreira
wrote:
> Anyone?
> I tried converting the file to g722 with ffmpeg and got the same result.
>
> regards
> Tiago
> On 12-05-2017 12:10, Tiago Ferreira wrote:
>
> Hello everyone,
>
> I am using the Asterisk REST API in order to establish a call to an e
The Asterisk Development Team has announced security releases for
Certified Asterisk 13.13 and Asterisk 13 and 14. The available
security releases are released as versions 13.13-cert4, 13.15.1, and
14.4.1.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/te
...@lists.digium.com] On Behalf Of Héctor Royo
Sent: May 17, 2017 08:55
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13 queue and DND phones
Hi. I will try to give you an idea:
You can remap de 'Do not disturb' key to do some actions.
(Bes
Hi. I will try to give you an idea:
You can remap de 'Do not disturb' key to do some actions.
(Best examples I've found so far:
http://community.polycom.com/t5/VoIP/FAQ-Using-Enhanced-Feature-Keys-EFK-macros-to-change-key/td-p/5705
)
I once tried to remap de DND on a Polycom IP 550 to something
Hi,
I've noticed that when I set a phone on DND (phone-side DND, meaning it
rejects calls with a busy status, SIP 486 response code I believe) the
queue keeps on trying the phone over and over again.
This creates issues in terms of CDR entries - in a scenario where there is
only one phone o
Anyone?
I tried converting the file to g722 with ffmpeg and got the same result.
regards
Tiago
On 12-05-2017 12:10, Tiago Ferreira wrote:
Hello everyone,
I am using the Asterisk REST API in order to establish a call to an
endpoint and to send over a remote file (HTTP).
The issue is that I am
Hello everyone,
I am using the Asterisk REST API in order to establish a call to an
endpoint and to send over a remote file (HTTP).
The issue is that I am experiencing an audio quality issue.
I have tried encoding the file differently, but everytime Asterisk is
cutting the audio frequencies above
On Tue, May 9, 2017, at 11:06 AM, marek cervenka wrote:
> i can upgrade asterisk to DONT_OPTIMIZE version at night
>
> before that, do you see something strange?
>
> is it known issue?
The only issue that looks like it could be related is ASTERISK-26969[1].
Once you have an unoptimized backtra
i can upgrade asterisk to DONT_OPTIMIZE version at night
before that, do you see something strange?
is it known issue?
[Thread debugging using libthread_db enabled]
Using host libthread_db library "/lib64/libthread_db.so.1".
Core was generated by `/usr/sbin/asterisk -f -C
/etc/asterisk/asteri
when run from console without systemd i found its segfaulting
turned core dump on because it was off
Dne 09/05/2017 v 13:52 marek cervenka napsal(a):
hi,
i have strange problem with asterisk 13.15.0+pjsip bundled/centos
7/systemd start script
we are using chan_pjsip only for webrtc endpoint
hi,
i have strange problem with asterisk 13.15.0+pjsip bundled/centos
7/systemd start script
we are using chan_pjsip only for webrtc endpoints . switched from sipml5
to jssip with upgrade to 13.15.0(from 13.9.0) few days ago
today i have problems with stopping/crashing asterisk
/var/log/as
http://linoxide.com/tools/install-setup-asterisk-13-pbx-centos-7/
2017-04-27 2:15 GMT+08:00 Jerry Geis :
> > yum install jansson*
>
> This works for CentOS 7 but not CentOS 6.
>
> Thanks,
>
> Jerry
>
>
>
> --
> _
> -- Bandwidth a
> yum install jansson*
This works for CentOS 7 but not CentOS 6.
Thanks,
Jerry
--
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yum install jansson*
Jerry Geis 于2017年4月26日 周三下午8:32写道:
> >It can't be disabled. jansson is a required dependency for Asterisk 13
> >as JSON is used internally for things.
>
>
> Ok thanks - that is a little confusing since there are entries in the
> configure script that lead one to think it can
>It can't be disabled. jansson is a required dependency for Asterisk 13
>as JSON is used internally for things.
Ok thanks - that is a little confusing since there are entries in the
configure script that lead one to think it can be a configure time
switch.
I'll go the other route and install the
On Wed, Apr 26, 2017, at 09:24 AM, Jerry Geis wrote:
> Trying to install asterisk 13 on CentOS 6.
>
> The ./configure tells me:
> configure: error: *** JSON support not found (this typically means the
> libjansson development package is missing)
>
> I don't really need JSON so I thought I would j
Trying to install asterisk 13 on CentOS 6.
The ./configure tells me:
configure: error: *** JSON support not found (this typically means the
libjansson development package is missing)
I don't really need JSON so I thought I would just disable it.
./configure --with-jansson=no does not work
./conf
Greetings,
This is your friendly 6 month warning that Asterisk 11 will be
reaching an official end of life state on October 25, 2017. As many
of you know, for the past 6 months Asterisk 11 has been in security
fix only mode. This means it currently does not receive bug fixes,
but it does receive
I will try to reinstall everything according to your instructions and i
will come back. I might need a couple of weeks due to a business trip though
Στις 24 Απρ 2017 6:54 μ.μ., ο χρήστης "John Kiniston" <
johnkinis...@gmail.com> έγραψε:
> Well, My suggestion was to use FUNC_ODBC but instead you w
-users
Subject: Re: [asterisk-users] Asterisk download stats
I have tried to find these in the past, best I came up with was using Shodan.io
search
Looking for Asterisk I get:
TOTAL RESULTS
42,036
TOP COUNTRIES
United States 12,914
Russian Federation 3,173
> Subject: [asterisk-users] Asterisk download stats
> Message-ID:
> gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi,
>
> Are there any stats on where Asterisk is downloaded from based on the
>
--
Well, My suggestion was to use FUNC_ODBC but instead you went with
APP_MYSQL which has been depricated.
Did you compile APP_MYSQL? It's not enabled by default.
On Sat, Apr 22, 2017 at 1:25 PM, Atux Atux wrote:
> Thanks a lot for the reply.
> I did follow that already, but i do have a problem. H
In article ,
Atux Atux wrote:
>
> Thanks a lot for the reply.
> I did follow that already, but i do have a problem. Here is my
> extensions.conf part for that particular number
> exten => 6912345678,1,Answer()
> exten => 6912345678,n,MYSQL(Connect connid 127.0.0.1 root mypasswd asterisk)
> exten
Re: Hack attempt sequential config file read looking for
> valid files. (Tim S)
>
>
> --
>
> Message: 1
> Date: Sat, 22 Apr 2017 23:25:52 +0300
> From: Atux Atux
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
Hi,
Are there any stats on where Asterisk is downloaded from based on the
country?
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On Saturday 22 April 2017 at 22:25:52, Atux Atux wrote:
> Thanks a lot for the reply.
> I did follow that already, but i do have a problem. Here is my
> extensions.conf part for that particular number
> exten => 6912345678,1,Answer()
> exten => 6912345678,n,MYSQL(Connect connid 127.0.0.1 root mypa
Thanks a lot for the reply.
I did follow that already, but i do have a problem. Here is my
extensions.conf part for that particular number
exten => 6912345678,1,Answer()
exten => 6912345678,n,MYSQL(Connect connid 127.0.0.1 root mypasswd asterisk)
exten => 6912345678,n,MYSQL(Query resultid ${connid}
You can use func_odbc to do this.
https://wiki.asterisk.org/wiki/display/AST/Getting+Asterisk+Connected+to+MySQL+via+ODBC2
There is a good chapter in the Asterisk book about using ODBC for
hotdesking that may help you understand ODBC as well.
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-b
hi. currently i am running the phonebook in astdb with
*database put cidname 0123456789 "name_surname"*
and i retrive it with
*exten =>9876543210,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})*
Now, my system has mysql and i got all my contacts in there in a database
is called *asterisk
the output of ls -l is
root@pbx: ~ $ ls -l /var/run/asterisk/asterisk.ctl
srwxr-xr-x 1 asterisk asterisk 0 Apr 20 19:47 /var/run/asterisk/asterisk.ctl
root@pbx: ~ $
On Thu, Apr 20, 2017 at 7:46 PM, Antony Stone <
antony.st...@asterisk.open.source.it> wrote:
> On Thursday 20 April 2017 at 18:31:0
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, April 21, 2017 10:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264)
Hello
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