Brad
would you be so kind and show a sample config for how you setup the serial fork?
I have to set this up but have no idea where to start.
Some sort of template would be great just to give an idea of how this is done.
I should be able to figure out the rest as I go along.
Thanks
Kind Regard
I just setup serial hunting and it was pretty easy.
We stored the list of users to be called in memcached, counted how many
total users need to be called and advanced to the next one in
failure_route as needed, set a ring timeout and then have it go to a
voicemail server or wherever it needed to g
I saw a post saying it was possible to do hunt groups with col, but i wasn't
sure how one would keep up with calls to each group user so that one could
do "least called user" or even the user that was next in line to be called.
I am guessing avps and contact array could help, but i was just wonde
I've had a problem dropped in my lap so please forgive my general
ignorance with regards to opensips. We've got a pair of 1.5.1 opensips
servers running on centos 5.5 64bit. After some period of normal
operation, after some number of dr_reloads, we run into the following
problem. The other perso
Looking to use drouting module but need direct access to the destination set
in a similar fashion to what I think the load_contacts function of the lcr
module does.
Is there an internal avp being used that is accessible in the script?
If not I'll look to add a function that does this.
Thanks,
T
Hi Bogdan.
Thanks for your answer.
You may check a trace at http://200.13.254.180/~saguti/fax.cap (Wireshark
format)
if you see, for example frames 6, 10, 13 and 16, have the 200.13.235.178 IP
address at SDP connection, because they are rewritten to be handled by
mediaproxy.
Trouble frame is 26,
Hi Sergio,
So, for the re-INVITE you have a provisional reply before the 200 OK
(both replies for re-INVITE). Is this true ? and the 200 OK does not hit
the on_reply route ?
Do you have a full SIP trace of such a call ?
Regards,
Bogdan
Sergio Gutierrez wrote:
> Hello to all members.
>
> I hav
That is cool - please ping me when code is available - I would like to
play with it and to integrate it into the opensips.org service web page.
Regards,
Bogdan
Doddle WebPhone wrote:
> We can download the application itself and use/deploy it on our web
> servers.
> Code will be available soon
>
Looks like a mixing of stateless and statefull replies to me . load
the "signaling.so" module and replace the sl_send_reply() for 503 with a
send_reply.
Regards,
Bogdan
Brad Bendy wrote:
> Hi Bogdan,
>
> In this case the 503 is being sent from a route block via
> sl_send_reply, then with a
We can download the application itself and use/deploy it on our web servers.
Code will be available soon
Sergio
On Tue, Aug 24, 2010 at 11:47 AM, Bogdan-Andrei Iancu <
bog...@voice-system.ro> wrote:
> yes, but can you download the code of the doddle phone for your self?
>
> regards,
> Bogdan
>
>
Hi Bogdan,
In this case the 503 is being sent from a route block via sl_send_reply,
then with a exit() after the sl_send_reply()
The same behavior happens in both failure route and the standard route
block.
Im 100% sure ive done something wrong in the script :)
On Tue, 2010-08-24 at 17:46 +030
yes, but can you download the code of the doddle phone for your self?
regards,
Bogdan
Doddle WebPhone wrote:
> Hi Bogdan,
> Yes, Doddle Phone provides us with several ways of (free) integration.
> We can embed it as a widget or even hosting on our Web servers and
> integrating with our Web appli
Hi Brad,
I guess you are doing something funny in the script like allowing the
302 reply to be relaid out, but having the 503 generated by opensips -
by chance, do you send the 503 in stateless mode ?
Regards,
Bogdan
Brad Bendy wrote:
> Hi Bogdan,
>
> Here is a full trace, breakdown is like th
Hello to all members.
I have a deployment of OpenSIPS 1.5.3 whose configuration is designed in
such a way as every call uses rtpproxy. I am using force_rtp_proxy() for
initial requests, sequential requests and at on_reply_route().
For the particular case of FAX calls, I see that the first call is
Fixing former link:
http://widget.doddlephone.com/embed/webphone.jsp?sipserver=proxy.ideasip.com&username=deglk1&password=palindru&callto=1234567890&auto=yes&stun=stun.ideasip.com
">
Tel: +1 234 567 890
Sergio
On Tue, Aug 24, 2010 at 10:47 AM, Doddle WebPhone wrote:
> Hi Bogdan,
> Yes, Doddle
Hi Bogdan,
Yes, Doddle Phone provides us with several ways of (free) integration.
We can embed it as a widget or even hosting on our Web servers and
integrating with our Web applications (php, .NET, ruby, JEE).
Easier and faster way to integrate is by adding this code to the web page
application:
I'm no python expert but I figure this is something quite simple I'm doing
wrong.
I have mediaproxy (and mediaproxy2 on another install) and I'm looking for a
way
to prematurely force a call to end based on the call id ... I was hoping I
could send
delete to the mediaproxy or to the dispatche
Hello Bogdan & Dave,
Yes. It works!
avp_db_load("$ruri/username", "s");
xlog("TEST: OFF LINE FORWARD TO: $avp(s:fwdoffline)\n");
xlog("TEST: BUSY FORWARD TO to: $avp(s:fwdbusy)\n");
xlog("TEST: CALL FORWARD TO : $avp(s:callfwd)\n");
On Tue, Aug 24, 2010 at 2:30 PM, Bogdan-Andrei Iancu wrote:
>
Hi Bogdan,
Here is a full trace, breakdown is like this
.2 INVITES to .164
.164 INVITES TO .168
.168 sends a 302 to .164
.164 sends .2 a 503 followed by a 302
.2 should never know about the 302 at all, but it's still getting back
to the originating proxy.
We are not using get_redirects() to do
Hi All,
I'm having an issue with loose routing and call setups.
My call flow looks like the following:
192.168.112.110 (ATA) -> 192.168.110.1:5060 (OpenSIPS) -->
192.168.10.1:5080 (Sippy b2bua) -> 192.168.10.50:5060 (TDM Gateway -
Audiocodes)
Now, OpenSIPS and Sippy B2bua are on
Le Tue, 24 Aug 2010 11:00:13 +0300,
Bogdan-Andrei Iancu a écrit :
Thanks Bogdan!
> The command line options are overwritten by the script options. So if
> you do "-D" in cli and you script has "fork=yes", then opensips will fork.
>
> Regards,
> Bogdan
>
> Jean-Yves F. Barbier wrote:
> > Hi li
Alex,
That is the same with:
rewritehost("real_destination_ip");
t_relay("outbound_proxy_ip:5060");
Regards,
Bogdan
Alex Massover wrote:
>
> Hi,
>
>
>
> I found it out, thanks anyway J
>
>
>
> $du = "sip:outbound_proxy_ip:5060";
>
> rewritehost("real_destination_ip");
>
> t
Hi,
I found it out, thanks anyway ☺
$du = "sip:outbound_proxy_ip:5060";
rewritehost("real_destination_ip");
t_relay();
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Alex Massover
Sent: Tuesday, August 24, 2010 12:14 PM
To: OpenSIP
Hi,
I little bit confused about how to send traffic to outbound proxy (not opensips
as the outbound proxy, but opensips needs to send traffic to outbound proxy).
The flow is:
Incoming call --> OpenSIPS --> outbound_proxy --> real_destination
Will something like this work:
$rd="real_destinatio
Hi Sergio,
Is this web-phone free for download ? like to embed the app (and not a
link) into my web page ?
Regards,
Bogdan
Doddle WebPhone wrote:
> Maybe this can be useful for OpenSIPs users and their applications:
> We can build click2talk / webphone application empowering webpages
> with SI
Hello Adelson,
better upgrade to latest 1.6 version (1.6.3) from SVN - it contains all
recent fixes on 1.6 branch. Take care that the regexp engine for
dialplan module was changed from 1.6.2 to 1.6.3 as old one (trex) was
bogus and unmaintained. See
http://www.opensips.org/Resources/DocsMigrat
Hi Ross,
what do you try to achieve ? failover for LB ? If so, you just need to
call again do_loadbalance() in failure route, in order to select the
next available LB destination
Regards,
Bogdan
Ross Beer wrote:
> Hi,
>
> Is it possible to use the next gateway from within a failure route
> whe
Hi Ross,
See the lb_status MI function:
http://www.opensips.org/html/docs/modules/1.6.x/load_balancer.html#id227044
Regards,
Bogdan
Ross Beer wrote:
> Hi,
>
> Is it possible to disable a load lanance gateway from opensipsctl?
>
> Kind regards,
>
> Ross
>
> __
Hi JY,
The command line options are overwritten by the script options. So if
you do "-D" in cli and you script has "fork=yes", then opensips will fork.
Regards,
Bogdan
Jean-Yves F. Barbier wrote:
> Hi list,
>
> I compiled Debian packages from 1.6.3 source, but command line option '-D'
> does no
Hi Brad,
Maybe I do not fully understand your case, but opensips is not sending a
302 after 200 OK...Maybe you can post the call flow (a SIP trace) from
the SIP server showing the entire scenario.
Regards,
Bogdan
Brad Bendy wrote:
> Hi,
>
> Im having a heck of a time figuring this out:
>
> INV
Hi,
You mean the call does not get established? or the call is established
but terminated after some time ?
Could you post a SIP trace of such a call ?
Regards,
Bogdan
k1028 wrote:
> I am not a expert on this but would like to get some understand what is the
> problem with my configuration.
>
Hi Sujeev,
you can try it:
- avp_db_load("$ruri/username", "a") (see
http://www.opensips.org/html/docs/modules/1.6.x/avpops.html#id228513)
Regards,
Bogdan
Sujeev wrote:
> Hello List,
> Please let me know how to load multiple attribute at one time. let
> say. my "usr_preferences" table has 3 a
Maybe one way round this problem would be for us to replace the incoming RURI
on the ACK with a sips address maybe to force Opensips to forward it on the tls
connection
Not sure how to do this however or is there better way?
--- On Mon, 23/8/10, Nauman Sulaiman wrote:
> From: Nauman Sulaiman
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