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 al
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, 200
Anca,
I'm having the b2b crash at different moments now. While the system
seems relatively stable if I can get past the initial OK/ACKs, I'm
having what appears to be a problem generated by a to-tag with dashes in
it. For instance:
(IPs/hostnames have been replaced by either (proxy) for o
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
the scope of your authority or capability to rush anyone
if you did have 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,
>>
>>
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
inter
How is the destination route set with the B2BUA? For this scenario, I
have only one gateway and do not require an LCR or drouting-style
system.
I understand that separate functions are required, which I am using.
However, setting $du or using t_relay("destination.com:5060"); doesn't
seem to wo
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 - RE
ember 10, 2009 1:43 AM
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
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
-Original Message-
From: users-boun...@lists.opensips.org
[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 k
ded the refer scenario? You can find it here:
http://www.opensips.org/Resources/B2buaTutorial#toc15. You have to put
in in a file and then set the 'script_scenario' parameter to the path of
the file:
modparam("b2b_logic", "script_scenario", "path_to_scenario_refer
ite 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 sup
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, Nov
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 /
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 thi
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 #125
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.data10
EFER and do 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 hopin
nt: Friday, October 23, 2009 6:08 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Transfer issue
El Sábado, 24 de Octubre de 2009, Jeff Kronlage escribió:
> I follow this, but -and please correct me where I'm misunderstanding- isn't
> the point of using a Proxy versus
m: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] 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
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 is
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
ecognized as local SIP domain
So, have you added "64.111.17.11" IP as alias in script 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.
>
> -Origina
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
with the VIA part..
Also RURI is not to be changed during loose_route(), only if you have a
strict router proxy in front of youmaybe you can post the inbound
and outbound request (to see how the loose_route() is done)
Regards,
Bogdan
Jeff Kronlage wrote:
> The RURI.
>
> Thanks,
>
ds,
Bogdan
Jeff Kronlage wrote:
> Thanks Bogdan,
>
> An unrelated question:
>
> Does anything special need to be done with "via" statements when
> implementing NAT transversal?
>
> Fix_nated_contact() takes care of the contact field for me, but I
still
> end u
.XX;branch=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 Sp
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
] 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
> ...ig
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 recei
I'm getting this over and over in my syslog:
WARNING:usrloc:get_all_db_ucontacts: non-local socket
...ignoring
Thanks,
Jeff
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Users mailing list
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http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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 A
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 t
Any input would be much appreciated.
Jeff Kronlage
Data102
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
good IT/Developer type, I'd like to totally
understand what my script is doing J
Cheers,
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 l
s now much 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:
It didn't break anything else for me in this final configuration. This
code is called right before (and encompasses) where I normally called
t_relay() (see below), which is only once place in my script, in
route[1].
--
Jeff Kronlage
Senior IT Engineer, Data102
102 South Tejon, Suite
()) sl_reply_error();
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
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 you
Greetings all,
I am looking for a simple way to change a SIP response code.
In some circumstances, I'd like to receive a 486 from a UA, and in
failure_route, change it to a 600 before it's passed on to the
destination UA.
Is there a function for this?
Thanks!
Jef
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