/libmemcached* to /usr/lib/libmemcached* and it works
properly.
Thank you very much for your responses,
Antonio Anderson Souza
Vizir http://www.vizir.com.br - Blog http://www.antonioams.com/ -
Twitter http://twitter.com/antonioams -
LinkedInhttp://br.linkedin.com/in/antonioams
- Facebook http
.8 form /usr/local/lib to
/usr/local/lib/opensips/modules/
but the problems still the same.
Does anybody has some idea to help me?
Best Regards,
Antonio Anderson Souza
Vizir http://www.vizir.com.br - Blog http://www.antonioams.com/ -
Twitter http://twitter.com/antonioams -
LinkedInhttp
, use the stable
version[1] and apply only the patch that I need, or use the version in the
branch [2]?
[1] -
http://opensips.org/pub/opensips/1.6.3/src/opensips-1.6.3-notls_src.tar.gz
[2] - http://opensips.svn.sourceforge.net/viewvc/opensips/branches/1.6
Best Regards,
Antonio Anderson Souza
http
/docs/modules/1.6.2/tm.html#timer-based-failover
Best Regards,
Antonio Anderson Souza
Call me for free!
http://174.129.217.139:8080/makecallclient/callme?p=YXNvdXphQHZvaWNlLmJyYXN0ZWwuY29tLmJy
Voice Technology http://www.voicetechnology.com.br/ - Blog
http://www.antonioams.com/ - Twitter http
Dear Bogdan,
I finished the tests of the solution suggested, and it's working
perfectly!!!
Thank you very much for the awesome software and support,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
On Fri, May 28, 2010 at 12:45 PM, Bogdan-Andrei Iancu
bog...@voice-system.ro
Hi Bogdan,
Thank you very much for the explanation, I haven't understood that the
changes made in the main branch is inherited by the new ones.
I'll change my logic to concentrate the changes in the Branch route, and
provide a feedback to the list.
Thank you very much!
Antonio Anderson Souza
Hi Bogdan,
I've changed mu script to execute all of the uac_replace only on branch
route, and it's working perfectly.
Thank you very much for the help.
Best regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
On Thu, Jun 10, 2010 at 5:59 PM, Antonio Anderson Souza
;
}
append_branch();
if (!t_relay()) {
sl_reply_error();
};
}
branch_route[1] {
...
uac_replace_from(, sip:$avp(s:from)@$fd);
uac_replace_to(, sip:$avp(s:to)@$td);
...
}
Best regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
(L_INFO, PreRouting to TO: $rm gw:[$si:$sp] ruri:[$ru] from:[$fu]
to:[$tu] sourceip:[$si] callid:[$ci]\n);
}
I'll change the script to check the dp_translate return before execute
uac_replace_to, and I'll send a feedback to the list.
Thank you very much,
Antonio Anderson Souza
Voice Technology
http
is working perfectly.
Best regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
On Fri, May 28, 2010 at 11:37 AM, IƱaki Baz Castillo i...@aliax.net wrote:
2010/5/28 Antonio Anderson Souza anto...@voicetechnology.com.br:
Hi Bogdan,
Which is the best way to check the dialplan
Stream of an stablished call to this new destination.
What do you think? This could be an interesting new feature to Opensis
supports the Lawful interception requirements.
Best regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
On Mon, May 24, 2010 at 9:15 AM, Bogdan
Hi Rajib,
I have several customers using opensips in HA, and I use in all of then
Keepalived (instead of Heartbeat), and the Sipsak to test the service level,
so Sipsak send an OPTIONS request to Opensips and if does not receive the
answer, Sipsak trigger the failover in the Keepalived.
In other
Ahmed,
Could you send an wireshark trace to the list? It will be easier to check
what's going wrong.
Besta regards,
Antonio Anderson M. Souza
Voice Technology
http://www.antonioams.com
Em 29/04/2010 11:47, Ahmed Munir ahmedmunir...@gmail.comescreveu:
Hi,
I've configured OpenSIPs with
The best way is to use the uac module.
Calling the function uac_replace_from before the relay, follow a script
snippet bellow:
uac_replace_from(sip:2...@10.10.1.100 sip%3a2...@10.10.1.100)
Best regards,
Antonio Anderson M. Souza
Voice Technology
http://www.antonioams.com
Em 28/04/2010
Hi Bogdan,
Yes it's a multi part body, and it's content is binary.
Follow a wireshark capture [1].
[1] - http://dl.dropbox.com/u/2134454/SIP_MGC_to_MGC_ISUP.pcap
Best regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
On Wed, Apr 21, 2010 at 5:44 AM, Bogdan-Andrei
module, isn't it?
The textops does not support binary content is it right?
There are some tutorial, documentation to guide me to implement a new
module?
Does anybody can help me?
Thank you very much,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
Anca,
Thank you very much for the Contact suggestion, for me it's the best
scenario, I'll implement it and send the feedback to you as I test the
solution.
Best regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
On Tue, Apr 20, 2010 at 6:21 AM, Anca Vamanu
Register status in the Location table.
[1] - http://www.opensips.org/html/docs/video/webinar005/
[2] - http://www.opensips.org/html/docs/video/webinar004/
[3] - http://www.opensips.org/html/docs/video/webinar002/
Best regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
On Tue
Best regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
On Tue, Apr 20, 2010 at 1:28 PM, samoh dahmani.s...@gmail.com wrote:
Other question, I don't want to create another topic just for a little
question :p.
Does a AGI like in asterisk exist for Opensips?.
Best
Anca,
I've tested the solution and it's OK.
Thank you very much!
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
On Tue, Apr 20, 2010 at 11:45 AM, Antonio Anderson Souza
antonio...@gmail.com wrote:
Anca,
Thank you very much for the Contact suggestion, for me it's
is the Opensips IP, it's possible to
check if the INVITE is from a local IP.
I'd prefer the option 1, but I didn't find in the documentation how to add a
custom header in the B2B scenario.
Does anybody can help me?
Best regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
, this way Opensips will use the view
thinking that it was a real table, and i use the AUTH_DB module as the
documentation [1] explains.
[1] - http://www.opensips.org/html/docs/modules/1.6.x/auth_db.html
Best regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
On Fri, Apr
Dear Anca,
I just tested it, and it's working perfectly!
Thank you very much for your support,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
On Thu, Apr 15, 2010 at 12:17 PM, Anca Vamanu a...@opensips.org wrote:
Antonio Anderson Souza wrote:
Hi,
I have on doubt, my
Hi,
I'd like to know if it's possible to have the allow and deny rules in a
database table instead of in files, Can I use the address table to make the
permissions based on the FROM and RURI rules?
Best regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
Hi Anca,
Thank you very much for the status, We're alredy started to develop the
solution using the B2B modules.
Opensips Rocks!!!
Best regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
On Wed, Apr 14, 2010 at 9:47 AM, Anca Vamanu a...@opensips.org wrote:
Hi
Anca,
Thank you very much for your support, I've made the changes in my script and
I got the refer scenarios working properly.
Regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
On Wed, Apr 14, 2010 at 6:31 AM, Anca Vamanu a...@opensips.org wrote:
Hi Antonio,
You
the From header?
Regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
On Wed, Apr 14, 2010 at 4:44 PM, Antonio Anderson Souza
antonio...@gmail.com wrote:
Anca,
Thank you very much for your support, I've made the changes in my script
and I got the refer scenarios working
have some idea to help me?
Best regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
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?
Best regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
___
Users mailing list
Users@lists.opensips.org
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3 months)?
Best regards,
Antonio Anderson Souza
Voice Technology
http://www.antonioams.com
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