Re: [OpenSIPS-Users] Contact list in REGISTER response
On Tue, Feb 16, 2010 at 12:54 AM, mayamatakeshi wrote: > Hello, > about registration, is it possible somehow to make opensips to reply with a > "200 OK" containing only the Contact of the registering UA instead of all > contacts from usrloc? > > I'm having a problem with eyebeam. It has a bug (at least the version I'm > testing) and it doesn't parse the Contact header correctly and it always > gets the value of expires from the first contact listed. > So in the case of a "200 OK" with a Contact like this: > > Contact: ;expires=5;received="sip: > 192.168.2.5:5050", ;expires=40;received="sip: > 192.168.128.33:61717" > > it should get expires=40 but it is getting expires=5. And since eyebeam > re-registers 5 seconds before expirations, it sends REGISTER immediately and > this goes on in a loop till the expires of the first contact gets greater > than 5 (when the other terminal re-registers). > I have not tested yet, but I think I got it: I have to call the function save with the flag "r" (no Reply), compose the Contact header myself and send the reply. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Aliases to multiple users
Thanks, it helps...not the answer I was hoping for, but it helps ;) -dg On Mon, Feb 15, 2010 at 5:49 PM, Duane Larson wrote: > I don't believe the Alias_DB was meant to do what you are asking. You > would need to use > append_branch() along with serialize_branches() and some routing logic to > get the funtionality you want. > > Hope that helps. > > On Mon, Feb 15, 2010 at 7:36 PM, Daniel Goepp wrote: > >> We are currently using aliases for URI dialing into our network, and it >> works great, but I have a question. I'm not able to create two rows with >> the same username and domain values due to key constraints which I assume >> are put in there to match functionality in the module. However this is kind >> of limiting as I would like to have an alias ring to multiple endpoints. Is >> this possible? For example, have an alias m...@example.com, and ring my >> desk phone 2125551...@example.com and my soft client >> 4135551...@example.com? I'm aware that I could register two endpoints >> with the same line id, and that would work, but for other reasons they need >> to be separate registrations and numbers. Ideas? >> >> Thanks. >> >> -dg >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > -- > -- > *--*--*--*--*--* > Duane > *--*--*--*--*--* > -- > > ___ > Users 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
Re: [OpenSIPS-Users] Aliases to multiple users
I don't believe the Alias_DB was meant to do what you are asking. You would need to use append_branch() along with serialize_branches() and some routing logic to get the funtionality you want. Hope that helps. On Mon, Feb 15, 2010 at 7:36 PM, Daniel Goepp wrote: > We are currently using aliases for URI dialing into our network, and it > works great, but I have a question. I'm not able to create two rows with > the same username and domain values due to key constraints which I assume > are put in there to match functionality in the module. However this is kind > of limiting as I would like to have an alias ring to multiple endpoints. Is > this possible? For example, have an alias m...@example.com, and ring my > desk phone 2125551...@example.com and my soft client > 4135551...@example.com? I'm aware that I could register two endpoints > with the same line id, and that would work, but for other reasons they need > to be separate registrations and numbers. Ideas? > > Thanks. > > -dg > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Aliases to multiple users
We are currently using aliases for URI dialing into our network, and it works great, but I have a question. I'm not able to create two rows with the same username and domain values due to key constraints which I assume are put in there to match functionality in the module. However this is kind of limiting as I would like to have an alias ring to multiple endpoints. Is this possible? For example, have an alias m...@example.com, and ring my desk phone 2125551...@example.com and my soft client 4135551...@example.com? I'm aware that I could register two endpoints with the same line id, and that would work, but for other reasons they need to be separate registrations and numbers. Ideas? Thanks. -dg ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Via with "received=YYY.YYY.YYY.YYY"
We are using a commercial load balancer in front of a pair of OpenSIPs proxies
(V1.6.1).
The proxies add a Via header of the form:
Via: SIP/2.0/UDP
XXX.XXX.XXX.XXX:5060;received=YYY.YYY.YYY.YYY;branch=z9hG4bK0eB35ab4ec1c3e0e2d3
... and forward() on the request.
The replies received by the proxy are then sent to YYY.YYY.YYY.YYY (the load
balancer) but it can't figure out where to send them (so they appear to get
dropped). The replies need to get sent to XXX.XXX.XXX.XXX instead.
Is there some way I can remove the "received=YYY.YYY.YYY.YYY;" portion from the
Via?
I've tried using subst("/received=YYY\.YYY\.YYY\.YYY//") in the local_route and
adding the same subst() call just before the forward() but those didn't work.
Thanks!
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[OpenSIPS-Users] How to get dialed number after call start?
Hi. Opensips 1.6.1 How to get entered via dialpad numbers after call start(as DTMF??)? I want to create call parking with asterisk and need to get requested number for forwarding call to asterisk extension. Thx. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Route calls based on CPS rate
Yes, you can use the ratelimit module to control the cps.
You will need to assign a pipe for each outbound destination.
Regards,
Ovidiu Sas
On Mon, Feb 15, 2010 at 11:50 AM, rajib deka wrote:
> Hi Bogdan,
>
> I agree with you. But I have seen that RATELIMIT module is doing something
> like that. Can we use that module for each gateway by identifying the
> gateway at run-time, like
>
> after LB selected the destination, we can have something like
> if($du == ) {
>
>
> if (!rl_check_pipe("1") {
> rl_drop();
> exit;
> };
>
> }
>
> where the pipe is with some cps value and INVITE queue. Is this will be
> efficient.
>
> Regards
> Rajib
>
> On Mon, Feb 15, 2010 at 6:58 PM, Bogdan-Andrei Iancu
> wrote:
>>
>> Hi Rajib,
>>
>> LB module is doing routing based on the load as current ongoing calls per
>> destination.
>>
>> To compute the CPS for a destination can be a bit tricky - there is no
>> module for doing it, Probably you can try to count the call using some
>> shared mem variable (directly in script) and to try to calculate on the fly
>> the CPS, but as said, does not seams an easy one (from mathematical
>> perspective).
>>
>> Regards,
>> Bogdan
>>
>> rajib deka wrote:
>>>
>>> Hello all,
>>> Is it possible to route calls based on cps rate using OpenSIPS load
>>> balancer module. We have an enterprise implementation here using OpsnSIPS
>>> load_balancer, which is handling 100 cps using our different trunks. So we
>>> want to place calls according to trunks cps capacity. Is there any other
>>> module to handle this situation. your suggestion will be much appreciable.
>>> --
>>> Rajib Deka
>>> Software Engineer
>>> Servion Global Solution
>>> Chennai, India
>>>
>>> Mobile No: + 91 80157 09130
>>
>>
>> --
>> Bogdan-Andrei Iancu
>> www.voice-system.ro
>>
>
>
>
> --
> Rajib Deka
> Software Engineer
> Servion Global Solution
> Chennai, India
>
> Mobile No: + 91 80157 09130
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] Route calls based on CPS rate
Hi Bogdan,
I agree with you. But I have seen that RATELIMIT module is doing something
like that. Can we use that module for each gateway by identifying the
gateway at run-time, like
after LB selected the destination, we can have something like
if($du == ) {
if (!rl_check_pipe("1") {
rl_drop();
exit;
};
}
where the pipe is with some cps value and INVITE queue. Is this will be
efficient.
Regards
Rajib
On Mon, Feb 15, 2010 at 6:58 PM, Bogdan-Andrei Iancu wrote:
> Hi Rajib,
>
> LB module is doing routing based on the load as current ongoing calls per
> destination.
>
> To compute the CPS for a destination can be a bit tricky - there is no
> module for doing it, Probably you can try to count the call using some
> shared mem variable (directly in script) and to try to calculate on the fly
> the CPS, but as said, does not seams an easy one (from mathematical
> perspective).
>
> Regards,
> Bogdan
>
>
> rajib deka wrote:
>
>> Hello all,
>> Is it possible to route calls based on cps rate using OpenSIPS load
>> balancer module. We have an enterprise implementation here using OpsnSIPS
>> load_balancer, which is handling 100 cps using our different trunks. So we
>> want to place calls according to trunks cps capacity. Is there any other
>> module to handle this situation. your suggestion will be much appreciable.
>>
>> --
>> Rajib Deka
>> Software Engineer
>> Servion Global Solution
>> Chennai, India
>>
>> Mobile No: + 91 80157 09130
>>
>
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro
>
>
--
Rajib Deka
Software Engineer
Servion Global Solution
Chennai, India
Mobile No: + 91 80157 09130
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[OpenSIPS-Users] Unveiling the design for OpenSIPS 2.0
Hi all, After words come the facts. Following the discussions and the suggestions about the new OpenSIPS design, the work in that direction started. The first step was to synthesize all the information and to come up with a realistic proposal for a new design - a design that will address all the know issue and requests. After couple of months of discussions, investigations and meetings, the draft for the new design was put together - a combination between simplicity, scalability and flexibility - a full technical description of the can be found at http://www.opensips.org/Development/NewDesignDescription The main actors behind the new design are: Bogdan-Andrei Iancu Dan Pascu Andrei Dragus Anca Vamanu The implementation work is already planned (timeline and manpower) and ready to kick off. Of course, this will impact on the current release policy of the project, but all the changes in the area will be the subject of a different email. We encourage as many people as possible to read the draft and to comment on it. Next step will be presenting a description of the core internals (threading, level, APIs, flow). Best regards, Bogdan -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Contact list in REGISTER response
Hello, about registration, is it possible somehow to make opensips to reply with a "200 OK" containing only the Contact of the registering UA instead of all contacts from usrloc? I'm having a problem with eyebeam. It has a bug (at least the version I'm testing) and it doesn't parse the Contact header correctly and it always gets the value of expires from the first contact listed. So in the case of a "200 OK" with a Contact like this: Contact: ;expires=5;received="sip: 192.168.2.5:5050", ;expires=40;received="sip: 192.168.128.33:61717" it should get expires=40 but it is getting expires=5. And since eyebeam re-registers 5 seconds before expirations, it sends REGISTER immediately and this goes on in a loop till the expires of the first contact gets greater than 5 (when the other terminal re-registers). regards, takeshi ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] registrar save "f" flag missing in current builds
Ha! I crack myself up. May I request this? - Jeff On Feb 15, 2010, at 8:37 AM, Bogdan-Andrei Iancu wrote: > Just checked, 1.6 does not have this addition - there was no such > backport done > > Regards, > Bogdan > > Jeff Pyle wrote: >> Interesting, I really thought I had it working on 1.6... >> >> On Feb 15, 2010, at 8:25 AM, Bogdan-Andrei Iancu wrote: >> >> >>> Hi Jeff, >>> >>> The "f" flag is present only on the devel / trunk version (SVN). >>> >>> Regards, >>> Bogdan >>> >>> Jeff Pyle wrote: >>> Bogdan, I updated to 1.6 SVN 6594 tonight. The new "f" flag for save() you wrote into 6527 is no longer there. I poked around some earlier builds and they appear to be missing there as well. Was there a problem with it? - Jeff ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> -- >>> Bogdan-Andrei Iancu >>> www.voice-system.ro >>> >>> >>> ___ >>> Users 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 >> >> > > > -- > Bogdan-Andrei Iancu > www.voice-system.ro > > > ___ > Users 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
Re: [OpenSIPS-Users] "SIP introduction" is the next webinar
The recording of this webinar is available under : http://www.opensips.org/html/docs/video/webinar004/ Regards, Bogdan Bogdan-Andrei Iancu wrote: > Next webinar is scheduled for 28th of January 2010. > > The topic is "SIP Introduction" - detailed explanation and examples of > SIP fundamentals: Requests and Replies, Initial and sequential requests, > SIP transactions, SIP dialogs, SIP and RTP; A good understanding of SIP > protocol is essential for working with OpenSIPS. > > Free registration - http://www.opensips.org/Training/Webinars#toc5 > > > The list with all the next scheduled webinars is available under > http://www.opensips.org/Training/Webinars#toc4 > > > Best regards, > Bogdan > > -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] softphone to opensips proxy connection
Hi Franck , Could you explain a bit in more details what you try to do ? as for me the goal and the setup are still unclear. Regards, Bogdan live-school support wrote: > Thanks Bogdan, > > I saw your opensips conference in Germany, > very intersting, thanks. > Well, I try since yesterday with 2 x-lite, 2 iptel sip accounts and > mediaproxy in proxy field (with no port specified) > and it works one time on 5. nothing on syslog unless > > >> b 9 11:03:44 node250 /usr/local/sbin/opensips[16633]: new branch at >> sip:xx...@iptel.org >> > > I set > debug=5 (how many level there is ?) > log_stderror=no > log_facility=LOG_LOCAL0 > > but randonly it works.. > > Any idea ? > > Thanks > > Franck > > > > > > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] registrar save "f" flag missing in current builds
Just checked, 1.6 does not have this addition - there was no such backport done Regards, Bogdan Jeff Pyle wrote: > Interesting, I really thought I had it working on 1.6... > > On Feb 15, 2010, at 8:25 AM, Bogdan-Andrei Iancu wrote: > > >> Hi Jeff, >> >> The "f" flag is present only on the devel / trunk version (SVN). >> >> Regards, >> Bogdan >> >> Jeff Pyle wrote: >> >>> Bogdan, >>> >>> I updated to 1.6 SVN 6594 tonight. The new "f" flag for save() you wrote >>> into 6527 is no longer there. I poked around some earlier builds and they >>> appear to be missing there as well. Was there a problem with it? >>> >>> >>> - Jeff >>> >>> >>> ___ >>> Users mailing list >>> Users@lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> -- >> Bogdan-Andrei Iancu >> www.voice-system.ro >> >> >> ___ >> Users 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 > > -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips Hardware Requirements
Hi, This is a kind of "question known to have no answer" - it was asked many times in the past and the answer is : hardware spec and performance depends a lot of your configures (script) and mainly of what exactly you want to your opensips to do. More or less you need to answer by yourself to this question as you better what you intend to put your opensips to do. Regards, Bogdan Indiver wrote: > Hello Everyone, > > I want to know the scalability of opensips 1.6.0. Such as its hardware > requirements and number of calls per second. Can any one provide this info. > Thanks in advance. > > Regards, > Nehru. > -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] registrar save "f" flag missing in current builds
Interesting, I really thought I had it working on 1.6... On Feb 15, 2010, at 8:25 AM, Bogdan-Andrei Iancu wrote: > Hi Jeff, > > The "f" flag is present only on the devel / trunk version (SVN). > > Regards, > Bogdan > > Jeff Pyle wrote: >> Bogdan, >> >> I updated to 1.6 SVN 6594 tonight. The new "f" flag for save() you wrote >> into 6527 is no longer there. I poked around some earlier builds and they >> appear to be missing there as well. Was there a problem with it? >> >> >> - Jeff >> >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > > -- > Bogdan-Andrei Iancu > www.voice-system.ro > > > ___ > Users 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
Re: [OpenSIPS-Users] Opensips for Embedded application (ARM9)
Hi Jirka, yes, ARM is a supported platform. What are the exact issues you found there? Regarding the mem and number of procs, all these can be configured: - private memory - compile time parameter - see config.h PKG_MEM_POOL_SIZE -> default val is 1M per process - share mem - run time param (see -m cli param) - default is 32 M for whole opensips - no of processes - see fork and processes config script params - default it starts in fork mode with 4 procs. Regards, Bogdan Jirka Hlavacek wrote: > Hello everybody, > I've found following thread which confirms the possibility of using > OpenSIPS on ARM 9. > > I would like to use OpenSIPS on ARM 9 with TLS support, but on x86 > it seems to be quite memory consuming (and some forks are > necessary...). > > Please, does anybody have more info (config, patch)? > > Thanks, > Jirka > > > > > Wed, 19 Aug 2009 06:31:36 -0700 > ... I want to port Opensips on ARM 9 board,so need some references on the > same. > >> OpenSIPS is known to work on ARM arch. >> > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Route calls based on CPS rate
Hi Rajib, LB module is doing routing based on the load as current ongoing calls per destination. To compute the CPS for a destination can be a bit tricky - there is no module for doing it, Probably you can try to count the call using some shared mem variable (directly in script) and to try to calculate on the fly the CPS, but as said, does not seams an easy one (from mathematical perspective). Regards, Bogdan rajib deka wrote: > Hello all, > Is it possible to route calls based on cps rate using OpenSIPS load balancer > module. We have an enterprise implementation here using OpsnSIPS > load_balancer, which is handling 100 cps using our different trunks. So we > want to place calls according to trunks cps capacity. Is there any other > module to handle this situation. your suggestion will be much appreciable. > -- > Rajib Deka > Software Engineer > Servion Global Solution > Chennai, India > > Mobile No: + 91 80157 09130 -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] registrar save "f" flag missing in current builds
Hi Jeff, The "f" flag is present only on the devel / trunk version (SVN). Regards, Bogdan Jeff Pyle wrote: > Bogdan, > > I updated to 1.6 SVN 6594 tonight. The new "f" flag for save() you wrote > into 6527 is no longer there. I poked around some earlier builds and they > appear to be missing there as well. Was there a problem with it? > > > - Jeff > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- Bogdan-Andrei Iancu www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy feature request - option to rely on SDP for connection info
On 12 Feb 2010, at 09:02, Jeff Pyle wrote: > Hello, > > I'm not sure if this is the correct forum for a Mediaproxy feature > request, but if it is, here's what I'm thinking. > > I use Mediaproxy mostly for accurate accounting of non-NAT users. > The problem is that some of my carriers begin their RTP by sending > it to me from a source port other than the one specified in the RTP > (which Mediaproxy learns) before switching to the port specified in > the SDP. By this point it's too late because the relay has already > locked to the first port number. What those carriers do is illegal. They should not change the media IP/ port without a renegotiation (re-INVITE or UPDATE request). Mediaproxy can deal just fine with changing the IP/port if done following a SDP renegotiation. > > I'd love to see the option to specify a flag as a modparam where, if > set, Mediaproxy would skip the auto-learning of the addresses and > port numbers and rely on those specified in the SDP. I would set > this flag for calls from/to non-NAT'd users so calls terminating to > these carriers would produce audio once they send traffic from the > port in their SDP. > > > Thanks, > Jeff > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dan ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Remote Party ID
Hi David, Yes, you must add it for all requests (even in dialog ones). There is no way to specify that a change on the message( like adding a header) should be done for all the requests in a dialog. Regards, -- Anca Vamanu www.voice-system.ro David J. wrote: > Where is the best place to use the append_rpid_hf()? > It seems I add it once, but subsequent messages dont include the header. > > Perhaps I have to add it for each request? > > Thanks. > > ___ > Users 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
