Hi John,
There is also a newer (but not newest) version of the tutorial:
https://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8
As per the setup description (in the tutorial), Asterisk is driven via
the OpenSIPS database, so you do not have to do any user provisioning
Someone plese?
Opensips 2.4.3 and control panel 8.2.4
has been installed
Cannot find how to integrate Asterisk
On Friday, November 9, 2018, John Tuxies wrote:
> Hi. i am new to the area of opensips and i would like some help. I am
> doing my baby steps, but i have some experience with Aster
Hi. i am new to the area of opensips and i would like some help. I am doing
my baby steps, but i have some experience with Asterisk.
In a Debian 9 64bit system i did install opensips as described in here:
https://www.powerpbx.org/content/opensips-v24-debian-v8-mariadb-apache-v1
I would like to inte
so, take a look at this lines in your cfg script.
and tls is not installed/compiled.
br
miha
Dne 1/3/2014 1:49 PM, pis(e Vishnu Vardhan:
Hi,
Thanks for quick response when i add the script for asterisk
integration to the opensips.cfg then i got problem, at that time
opensips server not runn
Hi,
Thanks for quick response when i add the script for asterisk integration to
the opensips.cfg then i got problem, at that time opensips server not
running when i removed that it is connecting normally and the log messages
are like follow.
Jan 3 06:40:00 osipsdev opensips: CRITICAL:core:yyerro
In tutorial there is written that on bouth side you have to create db.
You are getting error. What kind of errors. Look at logs, there you will
see which error's are u getting and what needs to be done.
br
miha
Dne 1/3/2014 1:19 PM, pis(e Vishnu Vardhan:
Hi,
I am using opensips 1.9 and ast
Hi,
I am using opensips 1.9 and asterisk 11.0.3 with centos 6.5 two servers.
How Can I integrate these two. I am following these this URL for
integrating this
http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8
.
But i am getting issue after placing the script in the
On Thu, Aug 25, 2011 at 11:54 AM, Gopal krishnan <
gopalakrishnan...@gmail.com> wrote:
> In OpenSER can't I see any log as like Asterisk console?
>
>
No, you'll need a trace.. Or alternatively you could modify your
configuration to print some meaningful debug. There isn't really an
interactive int
I am trying to register a VoIP trunk with OpenSER. I am not able
to register, the status is showed as "Request Sent" while
checking the sip debug, the register message is keep on sending
to openSER server but there is no response from the server.
If domains are
In OpenSER can't I see any log as like Asterisk console?
Regards
On Thu, Aug 25, 2011 at 10:01 PM, Brett Nemeroff wrote:
> You really need to perform a SIP trace to see what's going on. Tshark is
> really good for this. tshark is part of the wireshark platform.
>
> -Brett
>
>
> On Thu, Aug 25,
You really need to perform a SIP trace to see what's going on. Tshark is
really good for this. tshark is part of the wireshark platform.
-Brett
On Thu, Aug 25, 2011 at 11:12 AM, Gopal krishnan <
gopalakrishnan...@gmail.com> wrote:
> Hi,
>
> I am trying to register a VoIP trunk with OpenSER. I a
Hi,
I am trying to register a VoIP trunk with OpenSER. I am not able to
register, the status is showed as "Request Sent" while checking the sip
debug, the register message is keep on sending to openSER server but there
is no response from the server.
But the same I am able to register with Eyebea
Thanks for the suggestion Mike.
--
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Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
On 9/11/10 3:55 AM, osiris123d wrote:
> I can't use {SIPDOMAIN} because the {SIPDOMAIN} variable is actually the IP
> address of callers phone as it appears in the location table.
>
> On a side note I was able to not use P-Asserted-Identity. because of a
> different issue I learned about the uac_r
I can't use {SIPDOMAIN} because the {SIPDOMAIN} variable is actually the IP
address of callers phone as it appears in the location table.
On a side note I was able to not use P-Asserted-Identity. because of a
different issue I learned about the uac_replace_to() function. I was able
to place the
Hi,
Have you tried :
exten => _VMR_.,n,Voicemail(${EXTEN:4...@{sipdomain},u)
**
Regards,
Bogdan
osiris123d wrote:
> I have set up Asterisk to work with OpenSIPS so that instead of the context
> for all OpenSIPS Subscribers being "default" it is their actual domain.
> Following the OpenSIPS Tut
I have set up Asterisk to work with OpenSIPS so that instead of the context
for all OpenSIPS Subscribers being "default" it is their actual domain.
Following the OpenSIPS Tutorial to integrate with Asterisk worked well with
this until it came time for a x...@domaina to call a a...@domainb.
x...@
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