Jeremy McNamara wrote:
> Iñaki Baz Castillo wrote:
>> El Lunes, 18 de Agosto de 2008, Ali Jawad escribió:
>>
>>> Hi All
>>>
>>>
>>>
>>> Was anyone able to get the accounting part of asterisk2billing to work
>>> with Openser ? This looks like a pretty mature solution to me.
>>>
>>
>> Humm...
Hello Ali,
Please use b2bua.org. I used and i have success with this both Pre and Post
Paid solution.
without using Asterisk.
Thanks &Regards
Ravi Prakash Sunkara
VoIP Development Tech Lead
91-882776
Joe E. Lewis - "I distrust camels, and anyone else who can go a week
without a drink."
200
Daniel,
Could you explain what does these messages mean? Lack or resources?
I have configured children=20. What are the calculation for number of
children?
Thanks,
Toly
--
View this message in context:
http://www.nabble.com/openser-1.2.1-errors-in-the-log-tp19007938p19042450.html
Sent from the
I just remembered that long time ago I got tired about this question
and I built a simple sample config.
Here it is: http://voipembedded.com/resources/openser_cr.cfg
Regards,
Ovidiu Sas
On Mon, Aug 18, 2008 at 3:09 PM, Jeremy McNamara <[EMAIL PROTECTED]> wrote:
> Ovidiu Sas wrote:
>> append_brach
Hi Daniel,
Please find below the capture of a call from INVITE to BYE using the default
SIPp UAC and UAS scenarios that I modified to copy the Contact: header
content to the R-URI of ACK and BYE and the Record-Route: header content to
Route: (though I'm not sure whether it is the desired behaviou
Ovidiu Sas wrote:
> append_brach() is needed on failure_route(). This was discussed
> several times on the mailing list.
>
Smells like a high visiablity blog entry needs to be made on this topic
then :)
___
Users mailing list
Users@lists.kamaili
append_brach() is needed on failure_route(). This was discussed
several times on the mailing list.
Regards,
Ovidiu Sas
On Mon, Aug 18, 2008 at 2:54 PM, Elena-Ramona Modroiu <[EMAIL PROTECTED]> wrote:
> Jonathan K. Creasy wrote:
>> I have this block in my configuration:
>>
>>
>>
>> if(!cr_rewrit
Jonathan K. Creasy wrote:
> I have this block in my configuration:
>
>
>
> if(!cr_rewrite_uri("1", "call_id")){
> t_reply("403", "[EMAIL PROTECTED] Not allowed");
> } else {
># In case of failure, re-route the request
>prefix("+");
>t_on_failure("2");
>if (i
I have this block in my configuration:
if(!cr_rewrite_uri("1", "call_id")){
t_reply("403", "[EMAIL PROTECTED] Not allowed");
} else {
# In case of failure, re-route the request
prefix("+");
t_on_failure("2");
if (is_method("INVITE"))
t_on_reply("1");
Hi Ali,
it seems that the last weeks events created a lot of confusion among
openser users. openser was renamed to kamailio due to trademark issues,
a fork was created shortly after as opensips.
Now the ctl tool is named kamctl, see migration table:
http://kamailio.net/dokuwiki/doku.php/instal
Bastian Schern wrote:
> 4
>
> Daniel-Constantin Mierla schrieb:
>
>> Hello everybody,
>>
>> some people expressed inconveniences regarding the new name, so we would
>> like to get your opinion about, therefore this is a poll that will get
>> us to a decision. Here is the question:
>>
>> Do you
Hi,Is there anyway to get the statistics of different module using "opensipsctl
fifo get_statistics"?For example to obtain dns request/response statistics on
enum module.Thanks,___
Users mailing list
Users@lists.kamailio.org
http://lists.kamailio.org/cg
Hello,
they are available under new domain "kamailio":
http://www.kamailio.org/docs/
or directly:
http://www.kamailio.net/docs/modules/
Cheers,
Daniel
On 08/18/08 20:58, Douglas Garstang wrote:
> The module documentation for versions 1.2, 1.3 etc used to be
> available on the web site.
>
>
The module documentation for versions 1.2, 1.3 etc used to be available on the
web site.
Where has it gone? When you clicked on the documentation link, you used to be
able to choose the version of OpenSER(Kam...OpenSIPS whatever) that you wanted.
Now, your taken immediately to what appears to
Hello,
On 08/18/08 16:29, Ali Jawad wrote:
>
> Hi
>
> I am looking for a accounting solution, that I can use with Openser
> without involving Asterisk. Has anyone tested something similar ? I
> need the solution to work for my prepaid customers, for example he has
> got 10$ once the credit is u
Iñaki Baz Castillo wrote:
El Lunes, 18 de Agosto de 2008, Ali Jawad escribió:
Hi All
Was anyone able to get the accounting part of asterisk2billing to work
with Openser ? This looks like a pretty mature solution to me.
Humm... something says me that Asterisk2Billing is for... Asteri
On 08/18/08 18:54, Iñaki Baz Castillo wrote:
> El Lunes, 18 de Agosto de 2008, Ali Jawad escribió:
>
>> Hi All
>>
>>
>>
>> Was anyone able to get the accounting part of asterisk2billing to work
>> with Openser ? This looks like a pretty mature solution to me.
>>
>
> Humm... something says
Hello,
is your sipp script dealing with record routing properly? Can you post a
sample sip call from invite to bye (e.g., ngrep trace)? Will show better
whose fault is. I doubt it has something to do with openser.
Cheers,
Daniel
On 08/18/08 18:48, sergejf wrote:
> Hello,
>
> I'm attempting to
Answer 4
On Mon, Aug 11, 2008 at 12:50 PM, Daniel-Constantin Mierla <
[EMAIL PROTECTED]> wrote:
> Hello everybody,
>
> some people expressed inconveniences regarding the new name, so we would
> like to get your opinion about, therefore this is a poll that will get
> us to a decision. Here is the
El Lunes, 18 de Agosto de 2008, Ali Jawad escribió:
> Hi All
>
>
>
> Was anyone able to get the accounting part of asterisk2billing to work
> with Openser ? This looks like a pretty mature solution to me.
Humm... something says me that Asterisk2Billing is for... Asterisk?
--
Iñaki Baz Castillo
4
Daniel-Constantin Mierla schrieb:
> Hello everybody,
>
> some people expressed inconveniences regarding the new name, so we would
> like to get your opinion about, therefore this is a poll that will get
> us to a decision. Here is the question:
>
> Do you like the name "Kamailio"?
> 1) Yes,
Hello,
I'm attempting to load test a simple dispatcher script in OpenSER 1.3.x
using SIPp (built-in UAC and UAS scenarios). My dispatcher.list only has one
address 8.XX.XX.12 (a SIPp instance, UAS). SIPp on 8.XX.XX.10 sends the
INVITE messages to 63.XXX.XXX.110. Everything goes well until the UAC
Hi All
Has anyone tried portabilling with Openser ? If so can you please
provide a sample of your config file ?
THanks
With Regards
Ali Jawad
System Administrator
Splendor Telecom (www.splendor.net)
Beirut, Lebanon
Phone: +961 1 373725
Fax: + 961 1 375554
Hi
I am looking for a accounting solution, that I can use with Openser
without involving Asterisk. Has anyone tested something similar ? I need
the solution to work for my prepaid customers, for example he has got
10$ once the credit is used. Deny any more call.s
With Regards
Ali Jawad
Hi All
Was anyone able to get the accounting part of asterisk2billing to work
with Openser ? This looks like a pretty mature solution to me.
Thanks
With Regards
Ali Jawad
System Administrator
Splendor Telecom (www.splendor.net)
Beirut, Lebanon
Phone: +961 1 373725
Fax: + 961 1 375
Hello,
they are debug or warning messages. What do you mean by proxy is running
out of the socket in one hour?
Cheers,
Daniel
On 08/16/08 04:56, toly wrote:
> Greetings,
>
> Proxy compiled with TLS support, tls is not turned on
> Routing script inclulded below. Underlying database is Sybase.
>
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