Iñaki Baz Castillo wrote:
> 2009/9/17 Alex Balashov :
>> insert_hf()
>
> But that is just for request, not for responses...
> There is some function in tm module to add headers in responses (AFAIR).
>
That's not true, AFAIK:
http://www.opensips.org/html/docs/
All changes are made in openser.cfg.
priyank luthra wrote:
> Hi
>
> Can you tell me in detail, which file to make this change in.. i am a
> newbie to opensips I have openSER 1.3.0.
>
>
>
> On Thu, Sep 17, 2009 at 12:03 PM, Alex Balashov
> mailto:abalas
--
> Regards,
> Priyank
>
>
>
>
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he
same way SBCs are often used (unnecessarily so).
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te 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.
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a reply,
the stateful transaction layer used to open the transaction from the
initial request will not know that you did so, and vice versa, etc etc.
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Uwe Kastens wrote:
>> Replies are automatically routed only if they are statefully routed.
>
> Statefull = t_relay() ?
Yes.
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at message via forward("IP:port") to the media gw in on_replyto,
>>>>> busy is processed correctly. If not, the busy is not send to the mediagw.
>>>>>
>>>>> So I was wondering if I had to handle some replyto messages?
>>>>>
>>>>> BR
>>>
ing with SDP as well, is
> this correct?
Yes.
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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.
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different than in-dialog requests (requests arising within a dialog
created by the initial requests).
They are routed manually, not using loose_route() in any way.
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ages coming from the internal IP of
> the OpenSIPS server. Once all of the SIP messages get 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 wher
ut cheating
on your homework assignments.
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how to
> handle.
>
> See what I mean?
>
>
> - Jeff
>
>
>
> On 8/18/09 9:10 AM, "Alex Balashov" wrote:
>
>> The SST module is designed for a scenario in which the proxy serves as
>> the endpoint of the SST negotiation. Otherwise, SST i
;
>
>
> On 8/18/09 9:01 AM, "Alex Balashov" wrote:
>
>> If I'm understanding the documentation correctly, you'd probably have to
>> do this with manual header manipulation.
>>
>> Jeff Pyle wrote:
>>
>>> On 8/18/09 8:51 AM,
If I'm understanding the documentation correctly, you'd probably have to
do this with manual header manipulation.
Jeff Pyle wrote:
> On 8/18/09 8:51 AM, "Alex Balashov" wrote:
>
>> Sure, use a failure route and append_branch().
>
> Ok, but how do I ad
eader?
Sure, use a failure route and append_branch().
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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.
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id228526
>
> Regards,
> Bogdan
>
> Alex Balashov wrote:
> > Yes. Set max_contacts parameter in registrar module to 1.
> >
> > Alex G wrote:
>
mailing list
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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is
basically just QoS reporting and will not change the state of the stream.
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re are other, more OpenSER-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 re
t can and can't 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.
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;
> Josip
>
> On Fri, 7 Aug 2009 12:59:52 -0400, Alex Balashov
>
> wrote:
>> It's certainly possible. But you'd do well to tell us what you're
>> trying to accomplish to get the best advice.
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> 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.
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Web
It's certainly possible. But you'd do well to tell us what you're
trying to accomplish to get the best advice.
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Sent from mobile device
On Aug 7, 2009, at 12:52 PM, wrote:
> Hi there,
>
> I was wondering if there was a way to somehow pipe port mirrored sip
> calls
> to opensips, and the
s and
other significant stakeholders in the commercial ecosystem.
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up the mailling list as "normal" mailling
>> lists that with "Reply" just send the reply to the list and not to personal
>> inbox .. ?
>>
>>
>
>
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oxy isn't
supposed to statefully hide anything. :-)
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urmi lakkad wrote:
> modparam("dispatcher", "ds_ping_method", "INFO")
Asterisk does not respond to these. Try using the OPTIONS method instead.
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ect is deployed.
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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.)
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. Create
a ramdisk and have recordings placed there. Then copy them off to
another server at some interval for the purpose of re-encoding and/or
long-term storage.
I'd be curious to see how that performs. Obviously, RAM is a barrier.
-- Alex
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e opensips and mysql server.
>>
>>
>>
>> --
>> Thanking You,
>> Ashwini BR Naidu
>>
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ell as T.38 and can make the switch.
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de in advance whether the
call needs to go through a special PSTN GW when routing the initial INVITE.
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;
or a reference to a bug report number would be necessary to answer this
question.
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someone please point
> me in the right direction?
>
> Thanks,
>
> Alberto
>
>
>
>
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>
message in context:
> http://n2.nabble.com/accounting-BYE-tp3274605p3274605.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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ion.
>
> 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-
reated before the
request 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)?
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s is true only in the context of non-CANCEL and non-ACK
> requests!
>
> Regards,
> Bogdan
>
> Stanisław Pitucha wrote:
>> 2009/7/14 Alex Balashov :
>>
>>> http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id272150
>>>
>>
>> A bit rela
Stanisław Pitucha wrote:
> 2009/7/14 Alex Balashov :
>> 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 a
Consider:
http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id272150
"[I]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
>
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you spend a lot of time on that. I'd love 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 <mailto:i...@aliax.net>> wrote:
>
>
Iñaki Baz Castillo wrote:
> El Viernes, 10 de Julio de 2009, Alex Balashov escribió:
>>> npdi and rp are *userinfo* parameters (in fact they are TEL URI
>>> paremeters so when converting to SIP URI they become part of the userinfo
>>> part). http://www.tech-invi
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 Balashov
> wrote:
>>>> Yes, you can.
>>>>
>>>> Just beware that you will _have_
Victor Pascual Avila wrote:
> On Fri, Jul 10, 2009 at 8:12 PM, Alex Balashov
> wrote:
>> 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
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.
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On Jul 10, 2009, at 2:09 PM, "Paul Mancheno H."
wrote:
> Hello.
>
> I have a project to do a system
ds, 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.
>>
>>
they want.
>
> http://www.catb.org/~esr/faqs/smart-questions.html
>
> It was also a good read for me.
>
> Regards,
> Adrian
>
>
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s 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; they do different things.
>
> The failure_route[x] won'
o have a t_on_failure inside of a failure_route[x] ? Or
> is there another method I could / should use?
>
>
> Thanks,
>
> Patrick
>
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Specific and well-parameterised questions really are the key.
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On Jul 7, 2009, at 2:00 PM, Uwe Kastens 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 answer
> would
ying to find a more efficient way of achieving this.
>
> Thanks for any inputs you might have !!
>
> --- Jay
>
>
>
>
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ical
operations 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})
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this problem or know whats going on?
>
> Thank you for your time
>
>
>
>
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address. (Since the client is currently passing an
> internal ip for media IP)
>
>
>
>
>
> Anil
>
>
>
>
>
>
>
>
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projects inevitably diverge somewhat, and
assuming their proprietors do not see a mutual interest in module
compatibility.
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Mobile :
ion (based
> on src.ip 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.
>>
&g
of where the client registers, scaling is available by adding more SBCs and
> controlling which users hit which SBCs.
>
>
> - Jeff
>
>
>
> On 6/8/09 8:29 PM, "Alex Balashov" wrote:
>
>> It is absolutely indispensable to separate signaling and me
T scenario this was the case; I had
> never seen it done separately in a NAT scenario. That's good news.
>
>
> - Jeff
>
>
>
> On 6/8/09 8:22 PM, "Alex Balashov" wrote:
>
>> No, it is not necessary.
>>
>> The signaling and the b
Is this necessary? In Opensips/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.
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Gavin Henry wrote:
> Hi,
>
> Does conntrack need to be a module or can it be compiled into the kernel?
>
> On a kvm virtual machine we have it shows errors even though this is
> compiled in.
>
> Thanks.
>
Either.
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> separate module ? I fail to see any...
Expanded database-centric functionality, especially free from 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
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gs + avp_db_load should be able to do much much more than this
> without a problem.. even gflags + the perl module would be better
>
>
> On Thu, May 28, 2009 at 9:18 AM, Alex Balashov
> wrote:
>
> 60,000 times a day is about 41 times a minute (or, a li
Right, but we're talking about the Perl module.
Brett Nemeroff wrote:
> gflags + avp_db_load should be able to do much much more than this
> without a problem.. even gflags + the perl module would be better
>
>
> On Thu, May 28, 2009 at 9:18 AM, Alex Balas
e some custom 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" wrote:
>
&g
gt; This mail was received via Mail-SeCure System.
>
>
>
> This mail was sent via Mail-SeCure System.
>
>
>
>
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nt database operation will block a whole child process. Child
processes handle many requests concurrently in a high-volume scenario.
So, that needs to change.
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for the quick answer. Let me try the avp_db_query. thanks
> again Alex.
>
> On Tue, May 26, 2009 at 2:46 PM, Alex Balashov
> mailto:abalas...@evaristesys.com>> wrote:
>
> No generic database operations module. But you can use
> avp_db_query() from avpops
//lists.opensips.org/cgi-bin/mailman/listinfo/users
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Alex Balashov wrote:
> Iñaki Baz Castillo wrote:
>
>> 2009/5/11 Alex Balashov :
>>> 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
Iñaki Baz Castillo wrote:
> 2009/5/11 Alex Balashov :
>> 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 w
g and OpenSIPS routes
>> it as any other in-dialog request).
>>
>>
>>> but unfortunately I came across some buggy UAs doing this.
>> What do you mean with it? what does this UAS?
>>
>>
>> --
>> Iñaki Baz Castillo
>>
>>
>> ___
t;
> some person that has presented him previously this problem?
>
> help!...
>
>
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t;always be the reason of someones happiness."
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laces of overlap, and like I said, each product has it's
> own strenghs. It's simply a matter of opinion.
>
>
> On Wed, Mar 25, 2009 at 8:33 AM, Alex Balashov
> mailto:abalas...@evaristesys.com>> wrote:
>
> Brett Nemeroff wrote:
>
> Bo
ation (that I can see) between what Asterisk provides - or is
designed for - and what OpenSIPS does. They seem to be most
emphatically dissimilar.
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Andrew Yager wrote:
> On 09/03/2009, at 4:39 PM, Alex Balashov wrote:
>
>> I suggest setting an "rpid" credential for these users in the
>> subscriber table. This can be applied when the outbound call from the
>> user is proxy-challenged.
>
> Thanks. T
; fax: (02) 9037 0591 mob: 0405 152 568
> http://www.rwts.com.au/ or http://www.stonebridgecomputing.com.au/
>
>
>
>
>
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Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
text do not take 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,
>>
a standard with no future and no
> company invests in it anymore.
>
> Maybe I am wrong and this has much more sense in the US.
>
> Adrian
>
>
> On Feb 19, 2009, at 8:43 PM, Alex Balashov wrote:
>
>> To expand on this just a little bit:
>>
>> Whil
egy and implementation for
intercarrier 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 Geo
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 Ne
as used in
> H.248/Megaco? ie: dial 123 on TCIC 10012
> -Brett
>
>
> On Thu, Feb 19, 2009 at 11:27 AM, Alex Balashov
> wrote:
>> That is accurate.
>>
>> Brett Nemeroff wrote:
>>
>>> From what I understand about SIP-T it's SIP + ISUP pa
nipulate using a far wider range of tools.
SIP-T is also becoming an attractive external interconnect option.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775
_
;>
>>> Daviramos Roussenq Fortunato wrote:
>>>
>>>> Hi List.
>>>>
>>>> I have a trunk SIP-T must deliver it to the Asterisk in SIP. The Opens
>>>> is that?
>>>> How should be my opensip.cfg?
>>>>
>>>>
>>>> ---
he 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 A
ld be my opensip.cfg?
>
>
>
>
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Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1)
t;> listing (per sections) and not a place for publicity or
> service/product
>>>> descriptions. The idea is simple: if you do business around OpenSIPS,
>>>> you may list yourself there for people to find you - nothing more.
> Page
>>>> is free to edit, so an
p others 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
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k in 2 hours
> (neither in 8 hours).
> Such kind of miracle doesn't exist.
Then there's the financial issue. People with solid knowledge are
usually quite busy; there is no way it is worth anyone's time to drop
what they're doing and get involved in anything for 2 hour
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