Yes, as far as I remember, in your dial string, simply use a
Dial(DAHDI/X/1234567) where X is the dahdi device channel number.
Hope that helps.
Matthew Fredrickson
On Wed, Jul 13, 2016 at 5:22 AM, Mehdi Shirazi wrote:
> Hi
>
> How is it possible to use Dial application
On Thu, Jul 14, 2016 at 6:45 AM, A J Stiles
wrote:
> On Thursday 14 Jul 2016, Joshua Colp wrote:
> > Carlos Chavez wrote:
> > > Until Asterisk 11 I could use sip.conf to set defaults for all phones
> > > (language, dtmf, vmexten, etc) and just leave many fields in
hi,
i'm trying replace CDR with CEL
reasons:
- minimize Stasis listeners (CDR)
- CEL, CDR produces "similar" data
- own logic of CDR meaning like "calldate,src,dst,direction,.." dst is
always first connected point in PBX - real user or IVR/queue etc.,
numbers are only attributes of object
Hi List
I have two questions:
1- Mailbox on the Asterisk Voicemail Server are created automatically?
2- Is there any support on the code to put the voice records on a
Cassandra NoSQL database?
BR
Joaquin
This email is confidential and may be subject to privilege. If you are
The Asterisk Development Team has announced the release of Certified Asterisk
13.8-cert1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/certified-asterisk
The release of Certified Asterisk 13.8-cert1 resolves several issues reported
by the
Hello.
Anybody in the list is using this IP phone?
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
--
No problems with authentication during invite after reboot?
I'm using insecure=no in SIP configuration.
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 07/14/2016 02:14 PM, Marcelo Terres wrote:
Hello.
Anybody in the list is using this IP phone?
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 Thursday 14 Jul 2016, Joshua Colp wrote:
> Carlos Chavez wrote:
> > Until Asterisk 11 I could use sip.conf to set defaults for all phones
> > (language, dtmf, vmexten, etc) and just leave many fields in the
> > database as NULL. What would be the proper way to do this for Asterisk
> > 13 and
Sporadically we get 1 way audio when one party is outside our firewall.
The caller is on NAT, and it works fine most of the time. Caller can hear
the called party, same thing going the other direction. Caller can hear
called party.
Asterisk 13.9 on Debian
chan_sip with two identical Grandstream
Many people are reporting the same issue, so it is not my imagination.
Asterisk 13 above 13.1 is useless for anybody who relies on res_odbc.so.
As you know, after that version, the dropped the complexity of Pooling onto
unix_odbc itself. Not so simple, it seems. I noticed that after a few hours
Saint Michael wrote:
Many people are reporting the same issue, so it is not my imagination.
Asterisk 13 above 13.1 is useless for anybody who relies on
res_odbc.so. As you know, after that version, the dropped the complexity
of Pooling onto unix_odbc itself. Not so simple, it seems. I noticed
Carlos Chavez wrote:
Until Asterisk 11 I could use sip.conf to set defaults for all phones
(language, dtmf, vmexten, etc) and just leave many fields in the
database as NULL. What would be the proper way to do this for Asterisk
13 and PJSIP?
Kia ora,
PJSIP doesn't have the ability in it to
On Wed, Jul 13, 2016 at 3:44 PM, Carlos Chavez
wrote:
> On 7/12/16 9:27 PM, George Joseph wrote:
>
>
>
> On Tue, Jul 12, 2016 at 11:55 AM, Carlos Chavez
> wrote:
>
>> I am still a little confused about how to activate MWI with PJSIP on
>>
with templates.
Regards
El 13/07/2016 a las 23:49, Carlos Chavez escribió:
Until Asterisk 11 I could use sip.conf to set defaults for all
phones (language, dtmf, vmexten, etc) and just leave many fields in
the database as NULL. What would be the proper way to do this for
Asterisk 13
On 2016-07-13 17:09, Ernie Dunbar wrote:
Hi everyone.
I'm trying to compile Asterisk with the smsq utility on Ubuntu 16.04
LTS, and while most things are compiling fine, smsq fails with the
following output:
root@test25:/usr/src/asterisk-certified-13.1-cert7/utils# make smsq
[CC] smsq.c ->
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