I'm confused on this as well - wouldn't you be effectively placing two
calls (one via a non-T38 gateway, one via a T38 gateway) to the same
destination? Figuring that most T38 is going to terminate to a single
analog device, I would think that were this possible at a SIP level, the
device would
John,
Look in to the get_redirects() function:
http://www.opensips.org/html/docs/modules/devel/uac_redirect.html#id2285
91
Jeff K
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of John Quick
Sent: Wednesday, November 25,
Anca,
Thanks *so* much - I really appreciate your effort!
--
Jeff Kronlage
Data102
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Anca Vamanu
Sent: Friday, November 20, 2009 9:07 AM
To: OpenSIPS users mailling list
Hi Anca,
One of my coworkers is on the devel mailing list and saw the repeated
OK issue I was having with the B2B come through as a bug report.
Don't mean to be a pest or to rush anyone, but we were curious if there
was a timeframe on that being addressed? It would help to know for our
the inclination. It is not.
Anca Vamanu wrote:
Hi Jeff,
I will solve it today or in the worst case tomorrow.
Regards,
Anca
Jeff Kronlage wrote:
Hi Anca,
One of my coworkers is on the devel mailing list and saw the repeated
OK issue I was having with the B2B come through as a bug
Anca,
I've nearly got my B2BUA ready for deployment, however, in a
front-end/back-end solution, with my standard proxy configuration in the
front-end (passing calls to the B2BUA back-end), I'm having trouble
putting a call on hold. Please note that all other functionality is
working great -
John,
I'm not the RADIUS expert at our organization, but we do log to RADIUS
and then use CDRTool to rate calls.
With CDRTool, we set an AVP called $avp(s:billing_party) and send it to
RADIUS, which is then written to MySQL, which is later interpreted by
CDRTool. CDRTool uses this variable to
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Transfer issue
Hi Jeff,
Jeff Kronlage wrote:
Anca,
Thanks for the quick reply. I tried as you suggested, on a development box,
and while it didn't work, it did look a lot more like what I was expecting to
see.
However, I guess I
, path_to_scenario_refer.xml)
The you have to call the b2b_init_request function with the refer
parameter:
b2b_init_request(refer);
Regards,
Anca
Jeff Kronlage wrote:
Anca,
Thanks again for your work on this. I've gotten the b2b modules working and
I'm attempting to use the REFER scenario
[mailto:users-boun...@lists.opensips.org] On Behalf Of Anca Vamanu
Sent: Monday, November 09, 2009 7:17 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Transfer issue
Hi Jeff,
Jeff Kronlage wrote:
Anca,
The key pieces of my config file are:
Loadmodule tm.so
loadmodule
(refer) for the initial Invite message.
I have also updated the documentation page and you can find there also
the scenario document for this feature:
http://www.opensips.org/Resources/B2buaTutorial#toc15.
Regards,
Anca
Jeff Kronlage wrote:
Hi Bogdan,
Thanks for the fantastic news.
I don't
I use Nagios for such things...
- Is my fly down? [5 minute interval]
- Do I have a cowlick? [3 minute interval]
Etc..
:P
Jeff
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Jeff Pyle
Sent: Wednesday,
Hello all,
Easy question -
allow_trusted() has obviously been removed from v1.6. What's an
equivalent function, and is there anything documenting the change?
--
Jeff Kronlage
Senior IT Engineer, Data102
j...@data102.com / http://www.data102.com
Nevermind, just found the answer
'the following functions were introduced: check_address(),
check_source_address(), get_source_group() to replace allow_address(),
allow_source_address(), allow_trusted()'
--
Jeff Kronlage
Senior IT Engineer, Data102
102 South Tejon, Suite #1250
Colorado
Dialplan keeps reporting that I have no data in the db... turning on
MySQL logging produces the following:
select dpid,pr,match_op,match_exp,match_len,subst_exp,repl_exp,attrs
from dialplan order by pr;
Note the extra space after the word dialplan, just before order by
pr. Eliminating this
Getting good at responding to my own posts today-
Wacky. Doing a minor upgrade to MySQL fixed the problem -- seemed
strange that a space would've caused that issue.
Regardless, it might be a good idea to eliminate the space?
--
Jeff Kronlage
Senior IT Engineer, Data102
j...@data102.com / http
the call transfer, totally transparent to the other party.
Regards,
Bogdan
Iñaki Baz Castillo wrote:
El Sábado, 24 de Octubre de 2009, Jeff Kronlage escribió:
Our setup has been initially
engineered for thousands of concurrent calls, and we're hoping to avoid
having a dozen Asterisk
I too am interested in this issue. Most of our users have PBXs that generate
the second INVITE out-of-the-box, but we're about to move into residential
service (hence my slew of NAT-related posts recently), and I'd like to have
transfers/REFERs working natively in OpenSIPS without having to
I'd like to be certain I understand from these last few posts regarding
this topic-
The suggestion is to use Asterisk 'behind' Opensips, transferring calls
to it only when a B2BUA is necessary?
I certainly understand not wanting to post a config, but can anyone
share a general idea of how this
] On Behalf Of Iñaki Baz Castillo
Sent: Friday, October 23, 2009 5:38 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Transfer issue
El Sábado, 24 de Octubre de 2009, Jeff Kronlage escribió:
The suggestion is to use Asterisk 'behind' Opensips, transferring calls
to it only when
or in domain
table ? if so, please remove it!
Regards,
Bogdan
Jeff Kronlage wrote:
Please also note this only happens on reinvites - the initial invite
is
fine.
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Jeff
] Additional info on potential registration
issue
Hi Jeff,
Do you use a shared location table (via multiple registrar servers) ?
Regards,
Bogdan
Jeff Kronlage wrote:
I'm getting this over and over in my syslog:
WARNING:usrloc:get_all_db_ucontacts: non-local socket
udp:HI.DDE.N.12:5060...ignoring
Usrloc mode is 3.
--
Jeff Kronlage
Senior IT Engineer, Data102
102 South Tejon, Suite #1250
Colorado Springs, CO 80903
(719) 387- x 1335 direct
(719) 578-8844 fax
j...@data102.com / http://www.data102.com
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun
=z9hG4bK-e4e5
cd84
I'm having some random problems with the user part of the URI randomly
vanishing after I call loose_route() when NAT is involved, and I'm
thinking these are related.
Thanks,
--
Jeff Kronlage
Senior IT Engineer, Data102
102 South Tejon, Suite #1250
Colorado Springs, CO 80903
(719
and outbound request (to see how the loose_route() is done)
Regards,
Bogdan
Jeff Kronlage wrote:
The RURI.
Thanks,
Jeff
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei
Iancu
Sent: Monday, October 19, 2009 9:13
Please also note this only happens on reinvites - the initial invite is
fine.
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Jeff Kronlage
Sent: Monday, October 19, 2009 10:33 PM
To: OpenSIPS users mailling list
Subject: Re
Hello all,
I'm having a random registration problem I haven't had a chance to fight
yet.
Right now, 100% of my users have their SIP gateways on static, public IP
addresses. We use static entries in the location table to route calls
to these locations presently. We're wanting to deploy Linksys
I'm getting this over and over in my syslog:
WARNING:usrloc:get_all_db_ucontacts: non-local socket
udp:HI.DDE.N.12:5060...ignoring
Thanks,
Jeff
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Yes, location table, sorry.
I appear to have solved my problem, albeit entirely from guessing.
I had usrloc's matching_mode set to 1. I'm not 100% confident I understand the
difference between using the call ID to match on registration, but I do know I
went from perhaps a 50% chance to
We do this with relative success using DNS load balancing. Our two
boxes are randomly load balanced, not precisely half half. We then
use a script that fires off test SIP messages at the boxes every 60
seconds, and run a second script that removes the entry from our DNS
server should one of the
,
Jeff Kronlage
Data102
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Brett Nemeroff
Sent: Tuesday, October 06, 2009 7:15 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Re-invite problem - 491 Request Pending
Bogdan,
I
would be much appreciated.
Jeff Kronlage
Data102
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
();
exit;
}
if (!t_relay()) sl_reply_error();
I wish I could give a more techie explanation on why this works - it was
a hackjob answer for me. Bogdan posted an answer perhaps a week ago
that explained it a bit.
Cheers,
--
Jeff Kronlage
Senior IT Engineer, Data102
faster) and there is no swapping.
--
Jeff Kronlage
Senior IT Engineer, Data102
102 South Tejon, Suite #1250
Colorado Springs, CO 80903
(719) 387- x 1335 direct
(719) 578-8844 fax
j...@data102.com / http://www.data102.com
From: users-boun...@lists.opensips.org
[mailto:users-boun
Brett,
The $var variables are persistent per transaction. For the entire
dialog, use dialog variables:
http://www.opensips.org/html/docs/modules/1.5.x/dialog.html
The key functions here are store_dlg_value() and fetch_dlg_value(), or
set_dlg_flag() and is_dlg_flag_set().
Now if
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