Joshua, issue has been filed. Thank you!
https://issues.asterisk.org/jira/browse/ASTERISK-26689
03.01.2017 20:58, Joshua Colp пишет:
On Mon, Dec 19, 2016, at 06:36 AM, Dmitriy Serov wrote:
Yes, this means the remote end was not sending any audio stream.
But it shouldn't.
According to [1]
Ok, thank you! Happy new year:)
3 янв. 2017 г. 11:13 PM пользователь "Joshua Colp"
написал:
> On Tue, Jan 3, 2017, at 02:12 PM, Anton Teyhrib wrote:
> > Hi Joshua!
> > Thank you for response.
> > I want to show user availability in external web application (ruby). I
> > also
>
On Tue, Jan 3, 2017, at 02:12 PM, Anton Teyhrib wrote:
> Hi Joshua!
> Thank you for response.
> I want to show user availability in external web application (ruby). I
> also
> found that ARI can do this. What is the best way to do that? ARI or AMI?
Either should be able to do what you need, but
Hi Joshua!
Thank you for response.
I want to show user availability in external web application (ruby). I also
found that ARI can do this. What is the best way to do that? ARI or AMI?
3 янв. 2017 г. 11:06 PM пользователь "Joshua Colp"
написал:
On Thu, Dec 29, 2016, at 03:43
On Fri, Dec 30, 2016, at 05:04 AM, Kevin Long wrote:
>
>
> Hello,
>
> I am using asterisk 14.2 and PJSIP, with TLS transport.
>
> I’m sure I’m doing something wrong here ..
>
>
> In 2 distinct softphone clients (Bria and Groundwire), I am able to
> register successfully, and place a SIP
On Thu, Dec 29, 2016, at 03:43 PM, Anton Teyhrib wrote:
> Hi all,
> Is there any native way to save endpoint statuses to database?
> I use asterisk 13 with pjsip and realtime, and didn't found proper way.
> I read that there is config parameter in sip.conf: rtupdate=yes. But how
> can I do that
On Mon, Dec 19, 2016, at 06:36 AM, Dmitriy Serov wrote:
> Yes, this means the remote end was not sending any audio stream.
> But it shouldn't.
> According to [1] before remote end send OK or ACK there is one way SDP,
> no any audio stream.
> PJSIP channel (option rtp_timeout) does not take this
On Tue, Dec 20, 2016, at 12:21 PM, Bryant Zimmerman wrote:
> I am working on moving from version 11 to version 13 for my fax
> applications.
> We are bumping into an issue where the bulk of the T38 faxes are
> failing.
>
> The sending test switch is reporting COMREC_ERR_TRANSMIT_PHASE
>
>
On Tue, Jan 3, 2017, at 12:32 PM, Motty Cruz wrote:
> Thank you Carlos, you’re right I am using PJSIP. Should I not use it?
PJSIP allows the following two formats:
PJSIP/@
PJSIP//
Where in the second case the SIP URI can contain the number. Provided
you use either of those (the first would be
On Thu, Dec 29, 2016, at 02:33 PM, Gabriel Ortiz Lour wrote:
> Hi all,
> Anyone with a clever ideia to find out (programmaticaly) on what
> context,
> within the dialplan structure, a exten is just telling the base context
> (with many includes inside) ?
> Att.
Can you provide more detail
Thank you Carlos, you’re right I am using PJSIP. Should I not use it?
Thanks,
Motty
From: Carlos Chavez [mailto:cur...@telecomab.mx]
Sent: Saturday, December 31, 2016 5:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Motty Cruz
Subject: Re: [asterisk-users] how to
Hello,
I'm not totally sure but HEP permit SIP signaling and RTCP data capture
only on PJSIP channels. For chan_sip you have to use captagent.
Regards
El 03/01/2017 a las 10:04, Olivier escribió:
Hello,
On a newly built Asterisk 13.13.1 system, I can't make HEP work with
chan_sip (though
On Tue, Jan 3, 2017, at 11:04 AM, Olivier wrote:
> Hello,
>
> On a newly built Asterisk 13.13.1 system, I can't make HEP work with
> chan_sip (though I could make it work with PJSIP on another instance).
>
> Looking either at [1] or hep.conf, I can't see anything meaning HEP
> requires PJSIP.
>
Hello,
On a newly built Asterisk 13.13.1 system, I can't make HEP work with
chan_sip (though I could make it work with PJSIP on another instance).
Looking either at [1] or hep.conf, I can't see anything meaning HEP
requires PJSIP.
Before diging deeper, may I simply ask if HEP requires PJSIP or
14 matches
Mail list logo