Hi Everyone,
I am trying to setup an Audio Call from firefox WebRTC to Asterisk. The
Flow is:
PC -> SIPoWS -> KAMAILIO -> SIPoUDP -> ASTERISK
Regular call (no srtp) works fine. However when I setup SRTP the asterisk
replies with 488 Not Acceptable Here.
I followed the Secure Calling Tutorial, but
One way to do it is to use Transfer. This will cause the callee to send 302
redirect to the caller. The caller then will jump to the extension
specified in the contact. You will have to dial again to the callee in the
new extension.
This solution will increase the traffic on your asterisk and you
Thanks Carlos,
I have created the table and changed the extconfig to :
musiconhold = mysql,asterisk,musiconhold
It works fine.
Yaron
On Mon, Feb 23, 2015 at 6:57 PM, Carlos Chavez cur...@telecomabmex.com
wrote:
On 2/23/15 3:03 AM, Yaron Nachum wrote:
Hello everyone,
I am trying to activate
Hello everyone,
I am trying to activate Music On Hold using DB on Asterisk 13.
It works fine but in order to use new Music On hold definitions I have to
reload the moh module.
- The following is my configuration in extconfig.conf - I added the
following line:
musiconhold.conf =
Hello Asterisk Users,
I have been looking for similar auto dtmf mode implementation on pjsip, but
didn't see it coming, so I decided to give it a try.
My basic plan was to do it as simple as possible with minimum changes
because I am not familiar with all Asterisk code. My idea is to use rfc4733
Hello Binni,
It is hard to say anything without more information.
You need to understand what happens in those dropped calls.
Logs would help. Traces might help also. Try mirror the traffic to another
server and capture it using tcpdump, or even run tcpdump on the server
itself.
On Tue, Dec
Hello Asterisk users and developers,
The last few weeks we had several crashes on live asterisks running
versions 12.2.0rc1 / 12.6.1 with PJPROJECT versions 2.1.0 / 2.2.1. We
opened a ticket - ASTERISK-24471.
After investigating the issue I can say that the scenario is a CANCEL being
received
...@digium.com wrote:
Yaron Nachum wrote:
Hello Asterisk users and developers,
The last few weeks we had several crashes on live asterisks running
versions 12.2.0rc1 / 12.6.1 with PJPROJECT versions 2.1.0 / 2.2.1. We
opened a ticket - ASTERISK-24471.
After investigating the issue I can say
Hello Mathew and everyone,
We had another crash on the 12.6.1 machine. This time it was Sigmentation
Fault. I opened another issue - ASTERISK-24493
https://issues.asterisk.org/jira/browse/ASTERISK-24493.
Yaron.
On Tue, Nov 4, 2014 at 6:16 PM, Yaron Nachum nachum.ya...@gmail.com wrote:
Mathew
if the message is not understood.
Yaron
On Tue, Nov 4, 2014 at 3:59 PM, Matthew Jordan mjor...@digium.com wrote:
On Tue, Nov 4, 2014 at 12:59 AM, Yaron Nachum nachum.ya...@gmail.com
wrote:
Hello Asterisk users developers,
I have opened an issue few days ago regarding the crash and the zombie
we still have crashes every day or two, and we
can't reproduce the issue in our lab with a simulator.
Thank you,
Yaron.
On Thu, Oct 30, 2014 at 9:01 AM, Yaron Nachum nachum.ya...@gmail.com
wrote:
Hello everyone,
I have opened a ticket number - ASTERISK-24471
https://issues.asterisk.org/jira
No,
I went over all my scripts.
Thanks for the help.
Yaron
On Wed, Oct 29, 2014 at 6:11 PM, Paul Belanger paul.belan...@polybeacon.com
wrote:
On Tue, Oct 28, 2014 at 11:10 AM, Yaron Nachum nachum.ya...@gmail.com
wrote:
Mathew,
When I run 'ps -ef|grep asterisk' the following processes
.
On Thu, Oct 30, 2014 at 8:43 AM, Yaron Nachum nachum.ya...@gmail.com
wrote:
No,
I went over all my scripts.
Thanks for the help.
Yaron
On Wed, Oct 29, 2014 at 6:11 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Tue, Oct 28, 2014 at 11:10 AM, Yaron Nachum nachum.ya...@gmail.com
?
Thanks,
Yaron.
On Tue, Oct 28, 2014 at 5:10 PM, Yaron Nachum nachum.ya...@gmail.com
wrote:
Mathew,
When I run 'ps -ef|grep asterisk' the following processes are displayed:
root 6861 1 0 Aug27 ?00:00:00 /bin/sh
/ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C
27, 2014 at 1:20 AM, Yaron Nachum nachum.ya...@gmail.com
wrote:
Hello Mathew,
Thank you for the reply.
I will open an issue and send debug information.
Can you explain more about the workaround? A reference to the
documentation would be fine.
Sure - really, what you are running
Hello Asterisk users,
We noticed that on Asterisk 12 zombie processes are being generated - They
are released after a while, but we have around 10-20 zombie processes
running.
We are not sure if this is a normal behavior or an issue.
We saw in the documentation that the bridging module creates
28, 2014 at 4:58 AM, Yaron Nachum nachum.ya...@gmail.com
wrote:
Hello Asterisk users,
We noticed that on Asterisk 12 zombie processes are being generated -
They are released after a while, but we have around 10-20 zombie processes
running.
We are not sure if this is a normal behavior
of them
are related to the PID's of those zombie processes.
Do you have any idea how to find out what are these processes?
Yaron.
On Tue, Oct 28, 2014 at 4:53 PM, Matthew Jordan mjor...@digium.com wrote:
On Tue, Oct 28, 2014 at 9:44 AM, Yaron Nachum nachum.ya...@gmail.com
wrote:
Hello Mathew
, 2014 at 3:22 AM, Yaron Nachum nachum.ya...@gmail.com
wrote:
Hello all,
We have recently upgraded some of our services to Asterisk 12 with PJSIP.
We have 2 issues related to DTMF:
1. in the regular SIP channel we had DTMF auto mode, which adapted the
DTMF settings according to the incoming
Hello all,
We have recently upgraded some of our services to Asterisk 12 with PJSIP.
We have 2 issues related to DTMF:
1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMF
settings according to the incoming INVITE - RFC2833 or inband. The is no
such settings in PJSIP. Do you
Hello,
I have an issue with Asterisk 12 PJSIP. When receving an INVITE with FROM
URI that has a port number, the Asterisk removes the port from URI on
consecutive Responses / Requests. This causes an issue with one of our SIP
servers (it doesn't recognize the response / request).
Below you can see
Thanks a lot Joshua. That's a great idea :-)
I will try it and get back to you.
On Thu, Apr 10, 2014 at 5:50 PM, Joshua Colp jc...@digium.com wrote:
Yaron Nachum wrote:
Hi everyone,
Kia ora,
I am starting to work with PJSIP on release 12.1.0.rc3.
I used to have Asterisk 1.8
Hi,
Anyone has a workaround?
On Tue, Apr 8, 2014 at 9:17 PM, Yaron Nachum nachum.ya...@gmail.com wrote:
Hi everyone,
I am running asterisk with release 12.1.0.rc3 and PJSIP.
I have a peer which sends OPTIONS method for session keep-alive, and the
asterisk is not responding
Hi everyone,
I am starting to work with PJSIP on release 12.1.0.rc3.
I used to have Asterisk 1.8 with the regular sip channel. I was using the
usereqphone settings in order to set user=phone on from and to URIs.
Is there a similar config in PJSIP?
--
Hi everyone,
I am running asterisk with release 12.1.0.rc3 and PJSIP.
I have a peer which sends OPTIONS method for session keep-alive, and the
asterisk is not responding to it. That of course disconnects the call after
a few minutes.
Is there a settings in the PJSIP.conf to respond to in dialog
Hello everyone,
I am upgrading from release 1.8 and I have a strange behavior with CDR
generation.
We are using a Redirect server for Number portability, and I see that once
the call is going through the Redirect Server additional CDR records are
generated - we have 3 additional records.
This
Hello,
I have installed the latest version 12 that has been released (12.1.0.rc3).
I have setup default dtmf mode (rfc47..) but when I am calling to a
endpoint that doesn't support it (no telephony event in the rtpmap) the
asterisk responds OK in the signalling but DTMF is not working.
Is it a
Hello everyone,
I have started testing the PJSIP stack.
I saw that it is possible to setup statically multiple AOR contacts, setup
qualify_timeout and attach it to an endpoint, and then dial using this
endpoint.
When I setup the configuration I used the cli in order to see the status of
the
Thanks Joshua,
I tried it already. That would generate a call to both AORs which is not
what I was looking for.
Isn't there a way to retrieve the AOR status from the dialplan?
On Tue, Mar 11, 2014 at 3:27 PM, Joshua Colp jc...@digium.com wrote:
Yaron Nachum wrote:
Hello everyone,
I have
Thanks for the response anyway.
I think that it would be great if someone would make it happen. It seems to
me trivial that once you enable to setup multiple AORs you would use them
:-)
Yaron.
On Tue, Mar 11, 2014 at 3:38 PM, Joshua Colp jc...@digium.com wrote:
Yaron Nachum wrote:
Thanks
Hi Mathew,
The regular sip stack has 'auto' dtmfmode which behaved as I said - if the
remote replied with telephony event it used RFC2833 otherwise it used
inband.
On Tue, Mar 11, 2014 at 5:43 PM, Matthew Jordan mjor...@digium.com wrote:
On Tue, Mar 11, 2014 at 8:23 AM, Yaron Nachum
Mathew,
Thanks Mathew. It's good to know the limitations :-)
Is there any plan to add it?
On Tue, Mar 11, 2014 at 6:38 PM, Matthew Jordan mjor...@digium.com wrote:
On Tue, Mar 11, 2014 at 11:23 AM, Yaron Nachum nachum.ya...@gmail.com
wrote:
Hi Mathew,
The regular sip stack has 'auto
Thanks Mathew,
That would be great - just to validate the status of the AOR before you
send the INVITE.
Great mailing list.
On Tue, Mar 11, 2014 at 5:38 PM, Matthew Jordan mjor...@digium.com wrote:
On Tue, Mar 11, 2014 at 8:45 AM, Yaron Nachum nachum.ya...@gmail.com
wrote:
Thanks
33 matches
Mail list logo
