the reply somewhere else.
You can, for example, do this (whether in stateless or stateful request
forwarding mode):
onreply_route[1] {
drop;
}
... I think it's along that general train of thought.
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com
are often used (unnecessarily so).
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi
as well, is
this correct?
Yes.
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi
file for
blind call forwarding.
Sure, but first I need you to provide me some amount of money.
Thanks.
And after that amount is provided, please provide me the name and number
of your principal so I can have a conversation with him about cheating
on your homework assignments.
--
Alex
processed I then need
the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy)
between the Asterisk server and VoIP Gateway.
Any helpful hints on where to look?
-Matt
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678
by the initial requests).
They are routed manually, not using loose_route() in any way.
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
___
Users mailing list
Users
Matthew S. Crocker wrote:
Can mediaproxy glue two RTP streams together (CallerA to CallerB)?
Can mediaproxy glue two RTP streams together from different interfaces/IPs
(eth0 eth1) ?
Yes.
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1
smadhoo6 wrote:
How to configure Opensips (version 1.5.0) to use a particular CODEC say..
Speex.?
This is like asking how to put the milk back in the cow with JSON.
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1
and append_branch().
--
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman
to be there with certain values and they
have different values or gone altogether. If these headers are added in the
request route does the same rule apply as with append_hf(), that is, they
cannot be removed?
The whole thing just seems odd.
- Jeff
On 8/18/09 9:01 AM, Alex Balashov abalas
. If
the far end replies with a 422, by default Opensips will relay the 422 to
the UAC who, well, won't know what to do with it.
It just doesn't seem fair to slap the UAC with a 422 it doesn't know how to
handle.
See what I mean?
- Jeff
On 8/18/09 9:10 AM, Alex Balashov abalas
@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
___
Users mailing list
Users@lists.opensips.org
http
,
Bogdan
Alex Balashov wrote:
Yes. Set max_contacts parameter in registrar module to 1.
Alex G wrote:
is there a way to prevent multi-reg of subscribers
, the most hard part would probably be state machine, and connecting
sip and rtp together.
So if you have any idea on how to acomplish that I and I think many others
faced with same challenge would be very gratefull.
Best regards,
Josip
On Fri, 7 Aug 2009 12:59:52 -0400, Alex Balashov
do.
One thing you have to keep in mind is that if you use a SIP proxy (like
OpenSIPS) for this, it is event-driven, so you can't make it shunt a
call to a different place mid-call.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1
-native approaches involving media
proxies as well.
Best regards,
Josip
Alex Balashov wrote:
Josip Djuricic wrote:
Is OrecX source available, or perhaps is it already able to do this
(forward required targeted traffic to mediagw or b2bua instead of
recording it? )
There is an open
It's certainly possible. But you'd do well to tell us what you're
trying to accomplish to get the best advice.
--
Sent from mobile device
On Aug 7, 2009, at 12:52 PM, josip.djuri...@voljatel.hr wrote:
Hi there,
I was wondering if there was a way to somehow pipe port mirrored sip
calls
I would like to know if is there a way to only change the user part of
RURI when doing alias_db_lookup()?
Not intrinsically, but you can always store the old domain prior to
alias_db_lookup() and then revert to it after the lookup completes.
--
Alex Balashov
Evariste Systems
Web: http
or anything
like that.
Just don't do it. It's bad for business, it's bad for both products,
it's bad for everyone. NOBODY wins if commercial adopters see this kind
of petty bickering and egotism, especially from lead developers and
other significant stakeholders in the commercial ecosystem.
--
Alex
urmi lakkad wrote:
modparam(dispatcher, ds_ping_method, INFO)
Asterisk does not respond to these. Try using the OPTIONS method instead.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237
to statefully hide anything. :-)
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
___
Users mailing list
Users@lists.opensips.org
http
list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
there is no knowledge of their
intrinsic codec capability set, there's no way to know what the decision
rendered ultimately is.
Also note that during a call, the codec may change.
By means other than re-INVITEs? (Which can also be inspected for SDP.)
--
Alex Balashov
Evariste Systems
Web : http
It's worth pointing out that no member of the OpenSER project stack has
been a pure SIP proxy for very long; they have certain UAS features
(registrar, PUA, NAT ping, etc.) As Bogdan said, a pure proxy would not
be terribly useful in most scenarios in which the project is deployed.
--
Alex
the switch.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin
would be necessary to answer this
question.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
___
Users mailing list
Users@lists.opensips.org
when routing the initial INVITE.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
___
Users mailing list
Users@lists.opensips.org
http
list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
in the right direction?
Thanks,
Alberto
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste
.
I added this function in 1.0 (?!?) as it was mainly intended for
proper CANCEL and ACK routing.
Regards,
Bogdan
Alex Balashov wrote:
Bogdan,
Are you saying that t_check_trans() will create a new transaction
for a non-ACK/CANCEL retransmission too? Or that it retransmits
the last
forwarding is actually initiated, especially if I cannot change
the request body in any way after I create the transaction manually
(which I understand the documentation to be saying)?
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct
]f the request belongs to a transaction (it's a retransmision), the
function will do a standard processing of the retransmission and ***will
break/stop the script***. The function return false if the request is
not a retransmission.
-- Alex
--
Alex Balashov
Evariste Systems
Web : http
Stanisław Pitucha wrote:
2009/7/14 Alex Balashov abalas...@evaristesys.com:
http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id272150
A bit related question. Since the docs mention:
If the processing of requests may take long time (e.g. DB lookups)
and the retransmission arrives
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
___
Users mailing list
Users@lists.opensips.org
http
://www.catb.org/~esr/faqs/smart-questions.html
It was also a good read for me.
Regards,
Adrian
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web
, this is the only mailing list that acts
this way!
On Fri, 10 Jul 2009 02:48:05 -0400 (EDT), Alex Balashov wrote:
Thank you for posting this. It is something that very, very often needs
to be said and bears repeating.
This a good read for those who show up on mailing lists without any
Yes, you can.
Just beware that you will _have_ to use something like 302s. If you
send the INVITE request back to the switch, it will be considered a
call loop.
--
Sent from mobile device
On Jul 10, 2009, at 2:09 PM, Paul Mancheno H. pmanch...@gmail.com
wrote:
Hello.
I have a project
Iñaki Baz Castillo wrote:
El Viernes, 10 de Julio de 2009, Alex Balashov escribió:
Victor Pascual Avila wrote:
On Fri, Jul 10, 2009 at 8:12 PM, Alex Balashovabalas...@evaristesys.com
wrote:
Yes, you can.
Just beware that you will _have_ to use something like 302s. If you
send the INVITE
to hear
how all of that works for you. I've got plans to do something similar in
the LNP space..
-Brett
On Fri, Jul 10, 2009 at 2:02 PM, Iñaki Baz Castillo i...@aliax.net
mailto:i...@aliax.net wrote:
El Viernes, 10 de Julio de 2009, Alex Balashov escribió:
npdi and rp
] ? Or
is there another method I could / should use?
Thanks,
Patrick
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web : http
there even when I had a third option. Now it is trying
all three options, but just wanted to make sure this was a logical
methodology I have safe guards in place to stop it from endlessly
looping
Patrick
On Jul 9, 2009, at 6:00 PM, Alex Balashov wrote:
You need both
/listinfo/users
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin
Specific and well-parameterised questions really are the key.
--
Sent from mobile device
On Jul 7, 2009, at 2:00 PM, Uwe Kastens ki...@kiste.org wrote:
You are right. We all started from the same point and asked
questions to
learn a lot. The more specific the question is, the better the
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web : http
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
on numerical transformation values. They won't be evaluated
properly. Have to assign them to an outside variable first. For
instance, can't do something like:
$(fU{s.substr,fU{s.len} - 10,10})
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
and geographic location),
- scale with dump server instead of sbcs,
BR
Uwe
Alex Balashov schrieb:
The topology you describe is an alternative, if you've got the capital
to blow on SBCs.
Jeff Pyle wrote:
Alex,
That makes sense, but for NAT? Vonage, for example. Signaling
somewhat, and
assuming their proprietors do not see a mutual interest in module
compatibility.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
/Mediaproxy terms,
does Opensips need to be operating on the same IP address as the media
relay?
No, it is not necessary.
The signaling and the bearer plane can be separate entirely.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678
, scaling is available by adding more SBCs and
controlling which users hit which SBCs.
- Jeff
On 6/8/09 8:29 PM, Alex Balashov abalas...@evaristesys.com wrote:
It is absolutely indispensable to separate signaling and media for
large-scale service delivery platforms. Think about
the
constraints of unrelated AVP constructs, such as the need to define an
avp_table as a modparam even if one is not going to use it.
-- Alex
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
database
operations for custom route decision making. It runs about 60k times per
day in a Xen VM with no memory or performance issues. I've been quite
pleased.
- Jeff
On 5/28/09 8:46 AM, Alex Balashov abalas...@evaristesys.com wrote:
Admittedly, OpenSER 1.3.x was the last time I tried
than this
without a problem.. even gflags + the perl module would be better
On Thu, May 28, 2009 at 9:18 AM, Alex Balashov
abalas...@evaristesys.com wrote:
60,000 times a day is about 41 times a minute (or, a little less
than
one operation per second
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo
in a high-volume scenario.
So, that needs to change.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
___
Users mailing list
Users
-asterisk-interwork
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web : http
Iñaki Baz Castillo wrote:
2009/5/11 Alex Balashov abalas...@evaristesys.com:
It sounds like the CANCEL with the To-tag should have a Route header as
well in order for it to be processed like any other sequential/in-dialog
request -- that is to say, under loose_route().
But it would
Alex Balashov wrote:
Iñaki Baz Castillo wrote:
2009/5/11 Alex Balashov abalas...@evaristesys.com:
It sounds like the CANCEL with the To-tag should have a Route header as
well in order for it to be processed like any other sequential/in-dialog
request -- that is to say, under loose_route
never had this problem )
A SIP middlebox (SBC) - (I Don`t have)
I use opensips with asteriks in the same server but in different port,
and I have asterisk set in mode comedia
any idea?
some person that has presented him previously this problem?
help!...
--
Alex Balashov
Evariste
/listinfo/users
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin
-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
) between what Asterisk provides - or is
designed for - and what OpenSIPS does. They seem to be most
emphatically dissimilar.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
.
On Wed, Mar 25, 2009 at 8:33 AM, Alex Balashov
abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote:
Brett Nemeroff wrote:
Both OpenSIPs and Asterisk are telephony toolkits and both
provide similar features (some better than others). So you're
/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
___
Users mailing list
Users@lists.opensips.org
http
mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
protocol are different, after all SIP-T
carries information that the SIP can not interpret.
The Problem is the following I get a SIP-T trunk and Asterisk to deliver
precise, how best to do.
2009/2/19 Alex Balashov abalas...@evaristesys.com
mailto:abalas...@evaristesys.com
and will work, just taking the resources of ISUP?
SIP-T is not talking with SIP protocol are different, after all SIP-T
carries information that the SIP can not interpret.
The Problem is the following I get a SIP-T trunk and Asterisk to deliver
precise, how best to do.
2009/2/19 Alex Balashov abalas
.248/Megaco? ie: dial 123 on TCIC 10012
-Brett
On Thu, Feb 19, 2009 at 11:27 AM, Alex Balashov
abalas...@evaristesys.com wrote:
That is accurate.
Brett Nemeroff wrote:
From what I understand about SIP-T it's SIP + ISUP params in the
message. The required bits such as RURI and SDP all
See:
http://tools.ietf.org/html/draft-jfp-sip-isup-header-00
Grep for CIC / cic.
Alex Balashov wrote:
Any standard ISUP attribute has a corresponding map into SIP-T. So,
yes, any bearer-related information is going to be in there as well.
Brett Nemeroff wrote:
One question that I'm
settlement. So, for the most part SIP trunking is used for
customer access only. The SS7 information must be conserved in this
type of setup, and that's one of the reasons the sort of thing that
SIP-T is exists.
Alex Balashov wrote:
Adrian Georgescu wrote:
Why should SIP-T still exist
personally mu
comments.
On Feb 19, 2009, at 10:39 PM, Alex Balashov wrote:
The problem is that outside of the VoIP cottage industry, this stuff
isn't legacy by any stretch of the imagination, in any way, shape,
or form. We're just used to fancifully imagining that it is.
Adrian
in the community.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin
Oh, I see. Yes, that would be an inappropriate suggestion then, my
apologies.
Bogdan-Andrei Iancu wrote:
Hi Alex,
I think Matteo is looking for something to generate OpenSIPS config file
and not a simple web interface to add users..
Regards,
Bogdan
Alex Balashov wrote:
http
/listinfo/users
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi
://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
___
Users mailing list
Users
Iñaki Baz Castillo wrote:
El Domingo, 1 de Febrero de 2009, Alex Balashov escribió:
It's very strange that Asterisk answers OPTIONS pings with a 4xx error,
because OPTIONS is the method Asterisk uses to do its own availability
pings -- that's what the qualify= setting for peers in sip.conf
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web: http
mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
?
Adrian Georgescu wrote:
Alex, I am trying to understand what precisely you are trying to
achieve. What precisely are you working around that cannot be done in a
natural way?
Adrian
On Jan 20, 2009, at 7:29 PM, Alex Balashov wrote:
Good workaround is to use translations in the proxy
mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web
kamailio trunk.
See second example here:
http://openser.blogspot.com/2008/10/registrar-enhancements.html
Module documentation at:
http://www.kamailio.org/docs/modules/devel/registrar.html
Cheers,
Daniel
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel
Do you care if it's online, as long as it answers the challenge
successfully with the same credentials it provides when it registers?
Daniel-Constantin Mierla wrote:
On 01/16/2009 03:53 PM, Alex Balashov wrote:
Daniel,
I am curious, what is the intended use case of this:
check
to the function. You get read of (1) useless transit via an
AVP and (2) useless AVP search all the time. Also you get a more compact
and clear scripting
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1
ram wrote:
Hi
is this possible with Opensips Multi Tenant system ( integrating with
Asterisk or Freeswitch)
Yes.
if yes, any advise how this can be achived ? any documents
Well, you integrate OpenSIPS with Asterisk or FreeSWITCH.
--
Alex Balashov
Evariste Systems
Web: http
in dbaliases table
Thanks,
Julian
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct
.
I guess that solves my issue.
Thanks,
Julian
On Sun, Jan 4, 2009 at 5:58 PM, Alex Balashov abalas...@evaristesys.com
wrote:
Have you tried adding both combinations to the DB manually without using
opensipsctl?
On Jan 4, 2009, at 10:55 PM, Julian Yap julianok...@gmail.com wrote:
So
to check that it is unique:
if is_value_in_db $DA_TABLE $DA_ALIAS_USER_COLUMN $TMP_OSIPSUSER; then
minfo $TMP_OSIPSUSER alias already in $DA_TABLE table
exit 0
fi
On Sun, Jan 4, 2009 at 6:11 PM, Alex Balashov abalas...@evaristesys.com
wrote:
I'm not sure if opensipsctl is broken
Most likely you are relaying INVITEs or other end-to-end requests
without altering the Request URI domain, thus forwarding them back to
the proxy.
On Dec 29, 2008, at 3:02 PM, Jeff Pyle jp...@fidelityvoice.com wrote:
Most likely there is a problem in your routing logic that is causing
a
?
Thank You,
Bruno Rodrigues
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
OpenSIPS does not control this signaling, and is not involved at the
MTP3 level.
Bruno Rodrigues wrote:
I mean use Opensips like a SCP. The GW using SS7 sending the mtp3 to
Opensips using sip-t and Opensips controlling this signaling.
-Original Message-
From: Alex Balashov
signaling using any protocol
(H.248/SIP-T)
-Original Message-
From: Alex Balashov [mailto:abalas...@evaristesys.com]
Sent: segunda-feira, 22 de dezembro de 2008 23:37
To: Bruno Rodrigues
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] SIP - Transaction
OpenSIPS does
://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
___
Users mailing list
Users@lists.opensips.org
need to know please from where I can download the rtpproxy package and
how
I can configure it
Regards
___
Users mailing list
[EMAIL PROTECTED]
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct
Alex R.S.M wrote:
The INVITE request to End-point B generated with append_branch() within
openSIP.
So how openSIP knows to generate a CANCEL message when one End-point
answers the call?
Are you generating it manually or using the registrar's forking mechanism?
--
Alex Balashov
Evariste
ID to route call to
OpenSIPS::AVP::add(369,$routeid);
}
And then in the OpenSIPS script opensips.cfg, I can read it:
if ($avp(i:369) == whatever) {
..
}
Ditto. That is the only way I have gotten it to work.
--
Alex Balashov
Evariste Systems
Web: http
101 - 198 of 198 matches
Mail list logo
