Re: [OpenSIPS-Users] How to fix Content-Length header?
On Martes 19 Octubre 2010 20:08:11 Dmitry Kravchenko escribió: > I have D-Link DIR-615. It's not in the list but SIP ALG is on by > default. My friend has the Linksys, not in the list. Off course, not all models that have SIP ALG are on the list, it's a wiki, you could add them. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to fix Content-Length header?
On Martes 19 Octubre 2010 18:24:24 Dmitry Kravchenko escribió: > But doesn't all modern routers are equipped with SIP ALG and > condsequently it is unportable to require it being OFF? On most models, SIP-ALG is disabled by default, because it simply doesn't work. Check here for a list them: http://www.voip-info.org/wiki/view/Routers+SIP+ALG And don't try to fix the request, because the problem is not only related to Content-Leght, SIP-ALG implementations are full-broken Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How to fix Content-Length header?
On Martes 19 Octubre 2010 17:47:43 Dmitry Kravchenko escribió: > I wrote in the following way: I advise you, don't try to fix SIP-ALG 'fixed' request, you will NEVER, EVER get to the right way. If you have SIP-ALG enabled routers, just change them or disable SIP-ALG. Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Adding data to a request before relaying it
On Viernes 01 Octubre 2010 09:55:05 Najib Hara escribió: > Hi everybody, > > I'm a newer in the OpenSIPS world and I'm trying to learn how to use it > efficiently. I'm working on a project where I have to modify incoming > requests before relaying them to their first destination. By modifying, I > mean sending those requests to a server which will send back messages with > the additional data to implement in the requests. The next step is to > collect those informations from the responses and add them to the initial > requests which will be relayed to their initial destination. My question > is: is OpenSIPS capable of doing this ? > > For more detail, here is a scheme. > > Thank you in advance for your responses The short answer is yes, OpenSIPS is able to do what you describe. Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] "Asterisk Contexts" in OpenSIPS
On Viernes 01 Octubre 2010 07:53:38 Deon Vermeulen escribió: > Hi Raul > > Thanks for the clarification and response. Really appreciate it. > > Have been looking at the siptraces provided by SIP Trace in Opensips > Control Panel. > > I'm guessing I still have a NAT Traversal issue. > > What is really strange is that I can only phone from us...@domaina.com > to us...@domain.com, but not visa-versa. > When I answer the call on us...@domain.com the call does not setup but > times out with error 408 on both ends. If as I suppose, you are new to OpenSIPS, I suggest you to begin with the standar config file, it does nat-fixing-handling, and when you undestand what it does, try to modify it for adding what youe need. Also a bunch of SIP knowleadge is "a must". Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] "Asterisk Contexts" in OpenSIPS
On Viernes 01 Octubre 2010 06:11:48 Deon Vermeulen escribió: > Pardon for asking but how do I do this? > > Don't understand what you mean by "dump your SIP traffic and make sure > that you route the ACK message to the correct destination" or how to > do this. It means using tshark, ngrep, tcpdump or witchever traffic capture tool you feal confortable with Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Problems with CDRTool 8.0.4
On Domingo 05 Septiembre 2010 12:39:59 Saúl Ibarra Corretgé escribió:
> On 09/05/2010 01:08 PM, Raúl Alexis Betancor Santana wrote:
> > Hi all,
> >
> > I'm testing CDRTool 8.0.4 and getting some problems with it ..
> >
> > Let first say what I have done:
> >
> > - Install cdrtool 8.0.4 from darcs repo
> > - Install freeradius-xs from ag-projects deb's repo
> > - Setup mysql, tables created, users, stored procs, all that it's ok
> > - Setup OpenSIPS as Install.txt from cdrtool said ... with the
> > radius_extra param of acc module and so on ..
> >
> > Now, if I run freeradius -X for testing ... I get ...
> >
> > rad_recv: Accounting-Request packet from host XXX.XXX.XXX.XXX:60397,
> > id=182, length=485
> >
> > Acct-Status-Type = Start
> > Service-Type = Sip-Session
> > Sip-Response-Code = 200
> > Sip-Method = Invite
> > Attr-55 = 0x4c837891
> > Sip-From-Tag = "y5jps2ys0j"
> > Sip-To-Tag = "as79e94c07"
> > Acct-Session-Id = "3c4d8c5b7b6e-qq6pz2331syr"
> > User-Name = "8orriz...@test.com"
> > Calling-Station-Id = "sip:8orriz...@sip.test.com"
> > Called-Station-Id = "sip:349yyzzz...@sip.test.com;user=phone"
> > Sip-Translated-Request-URI = "sip:s...@192.168.0.2"
> > Canonical-URI = "sip:34928382...@sip.test.com;user=phone"
> > User-Agent = "snom360/7.3.30"
> > Contact =
> > ";reg-id=1"
> > From-Header = ""Ra\303\272l Alexis Betancor Santana"
> >
> > ;tag=y5jps2ys0j"
> >
> > NAS-Port = 5060
> > Acct-Delay-Time = 0
> > NAS-IP-Address = XXX.XXX.XXX.XXX
> >
> >Processing the preacct section of radiusd.conf
> >
> > modcall: entering group preacct for request 0
> >
> >modcall[preacct]: module "preprocess" returns noop for request 0
> >
> > rlm_acct_unique: Hashing 'Sip-To-Tag = "as79e94c07",Sip-From-Tag =
> > "y5jps2ys0j",Client-IP-Address = XXX.XXX.XXX.XXX,Acct-Session-Id =
> > "3c4d8c5b7b6e-qq6pz2331syr"'
> > rlm_acct_unique: Acct-Unique-Session-ID = "c9518f17b9dd1b07".
> >
> >modcall[preacct]: module "acct_unique" returns ok for request 0
> >
> > rlm_realm: Looking up realm "test.com" for User-Name =
> >
> > "8orriz...@test.com"
> >
> > rlm_realm: No such realm "test.com"
> >
> >modcall[preacct]: module "suffix" returns noop for request 0
> >
> > modcall: leaving group preacct (returns ok) for request 0
> >
> >Processing the accounting section of radiusd.conf
> >
> > modcall: entering group accounting for request 0
> > radius_xlat:
> > '/var/log/freeradius/radacct/XXX.XXX.XXX.XXX/detail-20100905'
> > rlm_detail:
> > /var/log/freeradius/radacct/%{Client-IP-Address}/detail-%Y%m%d expands
> > to /var/log/freeradius/radacct/XXX.XXX.XXX.XXX/detail-20100905
> >
> >modcall[accounting]: module "detail" returns ok for request 0
> >
> > WARNING: Attempt to use unknown xlat function, or non-existent attribute
> > in string %{Sip-Application-Type}
> >
> > I don't see Sip-Application-Type defined in dictionary.opensips and it's
> > clear that OpenSIPS it's not sending it to freeradius, also ... who I
> > get it value from the opensips script? ... I mean where do I have to put
> > a
> > $avp(s:sip_application_type) = "something" ? and what should be
> > "something"?
> >
> > I have checked cdrtool code ... and don't see where it use
> > sip_application_type ... maybe I loosed something obvious.
>
> Sip-Application-Type is meant to contain what type of application the
> SIP session is performing. This can be "audio", "video", "file-transfer"
> and "chat". You need to set this from the OpenSIPS script as the
> Sip-Application-Type radius attribute.
But is not defined in the dictionary.opensip that cames with cdrtool ... I
suppose I should put it there or Freeradius and the radiusclient-ng will not
know what to do with it.
Also ... does cdrtool have a way of billing based on sip-application-type? ...
because I have not seen anything about that on the code.
Also, I get errors with the accounting_stop_query in sql.conf of freeradius,
because of Connect-Info that is also not transmited to the radius server
Best regards.
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[OpenSIPS-Users] Problems with CDRTool 8.0.4
Hi all,
I'm testing CDRTool 8.0.4 and getting some problems with it ..
Let first say what I have done:
- Install cdrtool 8.0.4 from darcs repo
- Install freeradius-xs from ag-projects deb's repo
- Setup mysql, tables created, users, stored procs, all that it's ok
- Setup OpenSIPS as Install.txt from cdrtool said ... with the radius_extra
param of acc module and so on ..
Now, if I run freeradius -X for testing ... I get ...
rad_recv: Accounting-Request packet from host XXX.XXX.XXX.XXX:60397, id=182,
length=485
Acct-Status-Type = Start
Service-Type = Sip-Session
Sip-Response-Code = 200
Sip-Method = Invite
Attr-55 = 0x4c837891
Sip-From-Tag = "y5jps2ys0j"
Sip-To-Tag = "as79e94c07"
Acct-Session-Id = "3c4d8c5b7b6e-qq6pz2331syr"
User-Name = "8orriz...@test.com"
Calling-Station-Id = "sip:8orriz...@sip.test.com"
Called-Station-Id = "sip:349yyzzz...@sip.test.com;user=phone"
Sip-Translated-Request-URI = "sip:s...@192.168.0.2"
Canonical-URI = "sip:34928382...@sip.test.com;user=phone"
User-Agent = "snom360/7.3.30"
Contact = ";reg-id=1"
From-Header = ""Ra\303\272l Alexis Betancor Santana"
;tag=y5jps2ys0j"
NAS-Port = 5060
Acct-Delay-Time = 0
NAS-IP-Address = XXX.XXX.XXX.XXX
Processing the preacct section of radiusd.conf
modcall: entering group preacct for request 0
modcall[preacct]: module "preprocess" returns noop for request 0
rlm_acct_unique: Hashing 'Sip-To-Tag = "as79e94c07",Sip-From-Tag =
"y5jps2ys0j",Client-IP-Address = XXX.XXX.XXX.XXX,Acct-Session-Id =
"3c4d8c5b7b6e-qq6pz2331syr"'
rlm_acct_unique: Acct-Unique-Session-ID = "c9518f17b9dd1b07".
modcall[preacct]: module "acct_unique" returns ok for request 0
rlm_realm: Looking up realm "test.com" for User-Name =
"8orriz...@test.com"
rlm_realm: No such realm "test.com"
modcall[preacct]: module "suffix" returns noop for request 0
modcall: leaving group preacct (returns ok) for request 0
Processing the accounting section of radiusd.conf
modcall: entering group accounting for request 0
radius_xlat: '/var/log/freeradius/radacct/XXX.XXX.XXX.XXX/detail-20100905'
rlm_detail: /var/log/freeradius/radacct/%{Client-IP-Address}/detail-%Y%m%d
expands to /var/log/freeradius/radacct/XXX.XXX.XXX.XXX/detail-20100905
modcall[accounting]: module "detail" returns ok for request 0
WARNING: Attempt to use unknown xlat function, or non-existent attribute in
string %{Sip-Application-Type}
I don't see Sip-Application-Type defined in dictionary.opensips and it's clear
that OpenSIPS it's not sending it to freeradius, also ... who I get it value
from the opensips script? ... I mean where do I have to put a
$avp(s:sip_application_type) = "something" ? and what should be "something"?
I have checked cdrtool code ... and don't see where it use
sip_application_type ... maybe I loosed something obvious.
Any hit or clue?
Best regards
--
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Dimensión Virtual
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[OpenSIPS-Users] CDRTool without FreeRadius
Does anyone tryed to use CDRTool + mediaproxy without Freeradius support? ... I mean only using MySQL. It's possible? Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] using OpenSIPS as outbound proxy
On Martes 31 Agosto 2010 04:42:39 S. Millard escribió: > Iñaki Baz Castillo > > > Asking for free support/consultance or customized full configuration > > scripts is not the purpose of this mailist, neither asking for > > configuration files for *another* software (FreeSwitch). > > > > If you ask here for help not related to the purpose of this maillist, > > other members CAN complaint. We are not forced just to help or to be > > quiet. > > > > Regards. > > I think i was very clear in the begining, I addressed my questions to > Bogdan as a novice and not for profit, so please keep it quiet, i never > asked and will never ask for your help. > Millard Sorry, but you don't ask on a list to SOMEONE but to EVERYONE, if you want to ask someone in particular, leave him a private message, I'm sure Bogan will not answer you either. I you don't know how to use a tool (a maillist), learn first before complaining. You was asking for a custom-ready-to-use solution, you was asking for free- consultance, and thats what you received ... nothing. Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Removing a Header in INVITE before forwarding
On Tuesday 10 August 2010 23:18:04 Aditya Kumar wrote:
> Hi All,
> I am using opensips as a outound Proxy.
>
> I am using exec_s5 in example script.
>
> Most of the cases the following script is working based on my requirement:
> route:
> if (uri==myself) {
> //removed
> };
> # user found, forward to his current uri now
> if (!t_relay()) {
> sl_reply_error();
> };
>
>
> Issue:
> when the proxy sees that the RURI is not that of "myself" it is relaying.
> which is what I want.
>
> But I see that Route Header in INVITE is also getting forwarded.
> Route is that of the opensips IP address.
>
> Can any one help me in telling /pointing script where in,
> When I see that uri is not myself, Repy the message as it is .
> But remove the Route header?
You should NEVER, EVER IN YOUR LIFE to remove the Router header, it you do so,
it's a 99.9% that you will be in troubles.
> pl let me know if I am not clear
If you paste an ngrep trace of what you get and try to explain more clear what
you whant, maybe we could help you.
Best regards
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Re: [OpenSIPS-Users] Removing a Header in INVITE before forwarding
On Thursday 12 August 2010 02:20:04 Richard Revels wrote:
> I think this also happens (route header on initial invite) when you set an
> outbound proxy on some clients. My soft phone does this. I just yank
> route headers with the remove header field command on initial invites as I
> don't want clients deciding how to route. As TR mentioned, that can be
> bad'ish.
>
> remove_hf("Route");
It's a very, very bad idea to remove the Route Header before relaying, because
if it get back (it will not, because you are removing the returning path),
your proxy will not know where to relay that request.
Best regards
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Re: [OpenSIPS-Users] Opensips: multi and séquanti al ringing
On Monday 07 June 2010 14:20:56 mehdi boudou wrote: > Hello, > > Is it possible to do multi and sequential ringing just with the > configuration file without any other module ? Yes, lookup at the docs how to do "serial forking" and "parallel forking" -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips forward register
On Thursday 22 April 2010 21:57:20 info wrote: > Thanks that worked. > One question though the IP registered to asterisk shows the openips IP > address is there any way to show the correct IP of the device or does it > always have to go to the opensips ip Sure you whant to do that ? ... take into account Asterisk limitations on handling REGISTERS. Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Article on mediaproxy
On Wednesday 10 February 2010 14:49:51 Madovsky wrote: > I saw a patch for RTPproxy that transcode, I think it's included in the > last version. RTPProxy it's not able to transcode, what you had seen it's a patch that allow to use DSP cards with RTPProxy, so it will be able to transcode, but that patch doesn't add transcoding capabilities to RTPProxy, only to use transcoding hardware. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Article on mediaproxy
On Wednesday 10 February 2010 04:57:04 Infos Madovsky wrote: > Hi, > > is any MediaProxy dev can comment this article please ? > > http://hans.fugal.net/blog/2007/03/06/adios-mediaproxy/ I'm not a Mediaproxy developer but I could say that that article it's totally worng. First, it's a 3 year old article, things changes. Second, no matter if that was a test 3 years ago ... 3 years ago and today Asterisk DOESN'T SCALE, and it's not a good solution as rtp proxy, it haven't been a good solution for that ever. Asterisk "it's good" for other things but not for media-proxing. I could only say that the one that wrote that artcile doesn't have any idea about what he was writing, moreover taking into account that it doesn't give any test results or either describe a test scenario, it seams only to talk that way because "someone else told him to do so" ... Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] opensips performance
On Friday 05 February 2010 12:42:31 Neo Anderson wrote: > Hi, > > 1. CPU : Quad Core 64 bit CPU > 2. 8 GB RAM > 3. Centos 5.4 64 bit OS > 4. OpenSIPS will have db lookup with radius integration for accounting. > > Please let me know how many simultaneous calls OpenSIPS can process for > above mentioned situations? Dear VoipExpert ... let's talk first about how much you would have to pay for that information, taking into account that it's obvious you don't want to do the tests yourself, so this would be a payed assistance. Best Regards -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] MediaProxy 2.3.10: T38 Fax using zoiper 2.14 for MAC
On Thursday 28 January 2010 11:22:52 Saúl Ibarra Corretgé wrote: > Hi Mike, > > I received your trace, but let me answer you on the list, I won't expose > anything:) > > Here is what happens: > > - Zoiper sends an INVITE with audio proposal. > - Asterisk answers with 200OK and audio answer (m=audio on the SDP) > - Zoiper sends a re-INVITE to change to T.38 with a proposal (m=image on > SDP) > - Asterisk answers with 200OK but m=audio on the SDP For general knowleadge ... that's normal with Asterisk T.38 support, that is full-broken, with T.38 Asterisk it's a pain in the ass. Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Music On Hold Opensips
On Thursday 21 January 2010 17:30:36 opensipsl...@encambio.com wrote: > Hello, > > An jeu., janv 21, 2010, Alex Balashov schrieb: > >On 01/21/2010 12:00 PM, Mehdi Bouchefra wrote: > >> I'd like to implement the function Music on hold on Opensips. > > > >OpenSIPS is a SIP proxy, not a media endpoint. So, it doesn't do > >that. > > If while OpenSIPS routes a call a person presses 'hold' on a > telephone, often it will send a INVITE (I think it's called a > REINVITE.) What about detecting this 'hold' REINVITE in the > route script and redirecting the message to the media server? > > In the end, another software will have to do the job of the media > server (as Alex points out), but it would seem a logical role for > OpenSIPS to play in some redirection for phones and their 'hold' > buttons. No, no, no and no ... OpenSIPS it's a proxy and COULD not do anything in a middle of an stablished dialog (an ongoing call) For doing a MoH server, you will need to use the B2BUA module (and I don't know if could be used for this ..). -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RES: Billing System
On Wednesday 20 January 2010 17:40:53 AsteriskGuide wrote: > A new book about OpenSIPS was released today > > http://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book > > Sample chapter: > http://www.packtpub.com/article/installation-of-opensips-1.6 > > Regards, > > Flavio E. Goncalves I have just ordered my copy today ... ebook + paperback ... Just a very, very good book Flavio, congratulations Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Feature request: additional authorization conditions in auth_db
On Tuesday 05 January 2010 12:59:31 Alexander wrote:
> Hello all!
>
> I have a question about authorization: is there ability to specify
> additional conditions? For example, like it's done in postfix?
> We check only username and password in OpenSIPS, but there may be
> additional info, like "status" (active, suspended, blocked, etc...). So,
> if name/password match, but status is not "active" - authorization fails.
>
> It may look like this in config file:
>
> modparam("auth_db", "additional_conditions", "fiStatus = 200").
>
> As I see, now this feature is not supported. If it's not
> planned to develop, I can try to implement this feature and provide
> patch or modified source files. Will it be useful for someone else but
> me? :)
I think that everyone off us use the group module for that task of ACL's, also
you could use a view as the table for auth.
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Re: [OpenSIPS-Users] Media-dispatcher
On Thursday 31 December 2009 09:24:51 Saúl Ibarra Corretgé wrote: > However Python 2.4 is getting quite old now so you better upgrade to > version 2.5 at least because at some future point Python 2.4 support > will be dropped. Umm ... it had been dropped, since it doesn't run on python 2.4 ... so no "will be dropped", better .. "it had been dropped" ... ;-) -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mediaproxy benchmarks?
On Saturday 28 November 2009 17:50:32 Brad Bendy wrote: > If it's all done on the kernel level, then RAM won't really matter either > correct? No need for massive amounts, just good NIC cards and a decent > processor id assume, correct? Yes, you are ok. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Mediaproxy benchmarks?
On Saturday 28 November 2009 14:55:36 Brad Bendy wrote: > Hi list, > > Does anyone have benchmarks on Mediaproxy? Im trying to see what kind of > throughput I can get on say a dual Quad Core 55xx system. From what Ive > read I need to run one instance per core because Python can't really > multithread. Has anyone done this, or tried running several instances > under something like Xen to get max use of the hardware. Those were my > two initial ideals. Taking into account that mediaproxy does not handle the RTP, because all what it does is to insert contrack rules into the kernel ... I see no improvements on running more that one instance per server. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Modify SDP
On Monday 16 November 2009 21:55:16 Daryl G. Jurbala wrote: > Not sure how that would make a difference, as it actually limits the subset > of messages the replace_body could act on (properly, of course, but I'm > just testing/debugging here). In any case, I gave it a shot and it acts > the same. Also, take into account that you must do that onreply_route block and not on the main route block -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Question about AUTH in Deregister and Update
On Monday 16 November 2009 03:08:50 jia li wrote: > Hi, > > Is there any possibility to disable authentication in DEREGSITER and update > the register flow in OpenSIPS server, and maintain authentication in > REGISTER flow? > Any configuration needed? Thanks! Better if you try to explain what you need ... because a "DEREGISTER" doesn't exits as it, it's a REGISTER with contact info and expire time set to 0, so in fact it's a REGISTER request. You could check for the presecense of the expire header and then do no ask for AUTH ... but that will be a very, very, very, very, very, very, very, bad idea. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] static nat and 1 to 1 nat
On Wednesday 11 November 2009 11:25:07 John Quick wrote: > Alex > > As far as I can tell, the records in location table that have null in the > received column are those where the client device is aware of its NAT > environment and has already substituted the correct external IP address and > port number. For example IP phones that are using STUN. Or a router doing SIP-ALG and not working ... -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] INVITE not forwarded, call fails
On Sunday 08 November 2009 15:47:38 lorenzo wrote:
> On 04/11/09 13:41, lorenzo wrote:
> > On 04/11/09 13:32, Iñaki Baz Castillo wrote:
> >> El Miércoles, 4 de Noviembre de 2009, lorenzo escribió:
> >>> and this is the wireshark capture of the "conversation":
> >>> http://pastie.org/683069
> >>
> >> this shows a call which is cancelled (CANCEL) and later a call which
> >> receives 180 Ringing,
> >>
> >> It's difficukt to inspect trace if you don't say what exactly occurs in
> >> that trace,
> >
> > hi Iñaki, thanks for your interest!
>
> guys, nobody got any advice on where to look for to solve this problem?
> if anybody needs more info, just let me know!
What I see from you trace, (better if next time you put a ngrep-sip trace ..),
is:
Apple DLink GW
INVITE ->
<- 100 Trying
CANCEL->
<- 200 Canceling
<- 486 Request Canceled
ACK ->
-
New INVITE ->
<- 100 Trying
[the next is supposed, not in trace]
INVITE ->
<- 100 Trying
[... end supposing]
<- INVITE
<- INVITE
180 Ringing ->
Maybe you are rerouting the second invite back to your UAC, or you do some
aliasing ... or something similar.
--
Raúl Alexis Betancor Santana
Dimensión Virtual
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Re: [OpenSIPS-Users] RTPPROXY
On Tuesday 03 November 2009 12:19:36 Olle E. Johansson wrote: > 3 nov 2009 kl. 12.36 skrev Raúl Alexis Betancor Santana: > > On Tuesday 03 November 2009 10:15:31 michel freiha wrote: > >> Dear All, > >> I would like to ask you please if there is a way to use OpenSIPs > >> with a > >> proxy for rtp other than rtpproxy? Mean if i can use a propriatery > >> rtp > >> proxy > > > > All you need to use a "propietary" rtp proxy is to develop a module > > for > > OpenSIPs who can "speak" to that rtp proxy. > > Is it clear that the API between OpenSIPS and the rtp-proxy is > unaffected by the GPL? > Any risk that the proprietary RTP proxy will be GPL at runtime? It shouldn't got GPL at runtime, because if you follow the same way as other modules ... you have a .c module (GPL) that speak to your rtpproxy over unix-socket, or TCP/IP socket ... so your rtpproxy doesn't need to be GPL, your "interface module" MUST be GPL, because it will be compiled against GPL code. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RTPPROXY
On Tuesday 03 November 2009 10:15:31 michel freiha wrote: > Dear All, > I would like to ask you please if there is a way to use OpenSIPs with a > proxy for rtp other than rtpproxy? Mean if i can use a propriatery rtp > proxy All you need to use a "propietary" rtp proxy is to develop a module for OpenSIPs who can "speak" to that rtp proxy. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] solved ; ( Re: parallel fork and mediaproxy => false behaviour?
Uwe Kastens escribió:
> Yes, I am sure about this. Is there any good tutorial out there?
>
>
No, as far as i known, only the docs about how branches should work.
> no, sorr, a typo, should be:
> new:
> route[1] {
> ...
> engage_media_proxy();
> ...
> t_on_branch(1);
> t_on_reply(1);
> route(3),
> ...
> }
>
> route[3] {
> ...
> ..
> t_on_reply(2);
>
> t_relay();
>
> ...
> }
>
So .. first you set t_on_reply to 1 .. and then to 2 .. so t_on_reply 2
will be the one that get runned, and so .. is on that route block where
your "fix" lives ... ;-)
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Re: [OpenSIPS-Users] solved ; ( Re: parallel fork and mediaproxy => false behaviour?
Uwe Kastens escribió:
> I was not able to get it up and running with use_media_proxy and
> end_media_session, since the mediasession was ended, if a BYE for the
> 2nd branch arrived.
>
That could only occurs if you get something wrong with the branches. For
sure :-)
> I changed a lot in my script and now its working with engage_media_proxy
> as expected. I have no idea why its working now. The only relevant think
> I have changed was. Strange, I will try which changed fixed that point.
>
> old:
> route[1] {
> ...
> engage_media_proxy();
> ...
> t_on_branch(1);
> t_relay(),
> ...
> }
>
> new:
> route[1] {
> ...
> engage_media_proxy();
> ...
> t_on_branch(1);
> t_on_reply(1);
> route(3),
> ...
> }
>
> route[1] {
> ...
> engage_media_proxy();
> ...
> t_on_reply(2);
>
> t_relay();
>
> ...
> }
>
Do you have a duplicated route[1] ?? ...
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Re: [OpenSIPS-Users] parallel fork and mediaproxy => false behaviour?
Uwe Kastens escribió: > Raúl Alexis Betancor Santana schrieb: > >> On Friday 23 October 2009 14:14:52 Uwe Kastens wrote: >> >>> Hi, >>> >>> >>>>>>> Is there an option to prevent this behavior with mediaproxy? >>>>>>> >>>>>>> opensips: 10.20.20.159 and 10.20.30.159 >>>>>>> UACs: 10.20.20.25 and 10.20.20.26 >>>>>>> UAS: 17.17.17.167 >>>>>>> >>>>>> You have something wrong on you opensips.cfg, for sure ... We have lot >>>>>> of UAC's working on that scenario you described, without any problem. >>>>>> >>>>> Good to hear that. >>>>> >>>>> >>>>>> Are you using the dialog module? ... if yes, take into account it >>>>>> limitations to work with parallel forking. >>>>>> >>>>> You are working not with engage_media_proxy() then? >>>>> >>>> Of course, becasue engage_media_proxy NEEDS the dialog module ... and >>>> it's a know limitation of dialog module that it doesn't work AT ALL with >>>> parallel forking, or with multiple 1XX replies. >>>> >>>> Better if you limit your uses of the dialog module to the minimum ... let >>>> say ... to 0 .. ;-) >>>> >>> At which time are you calling use_media_proxy() then? On the 1st INVITE >>> or later with the 200 OK with SDP? Is there any example out there? >>> >> On First invite after checking if needed, on_reply for 1XX or 200 with SDP, >> on >> re-invites >> >> You could see and example on sipwise.com >> >> > > Thanks. How would you handle parallel forking in that case? If I match > for end_media_session() on BYE the media_session is dropped if the UAS > sends the BYE for the 2nd call leg > You need to work with branches, so when on_reply route is excecuted, it will be executed for each branch reply, so no problems at all, because mediaproxy module knows that they are diffent branches, and work ok. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] One Way Audio Using X-Lite, OpenSIPS & Asterisk (Ross Beer)
On Friday 23 October 2009 14:19:45 Ross Beer wrote: > Hi, > > Here is the sip trace for the calls, it strange as all other phone work > except X-Lite, it appears that sound is not getting from the softphone to > the server. Though if the softphone talks directly to asterisk it works ok. > This is the case if MediaProxy is used or not. > > There is a lack of codecs in the invite which is strange as I have 4 > enabled on the server and the softphone. Seems you don't look on the wright place .. because your X-Lite is offering PCMU ... just look at the m= line on the first invite. > INVITE sip:160@ SIP/2.0 > Via: SIP/2.0/UDP ADDRESS>:9302;branch=z9hG4bK-d8754z-4e352701ef65e42a-1---d8754z-;rport > Max-Forwards: 69 > Contact: :9302> > To: "160"> > From: "Ross">;tag=ad038800 > Call-ID: ZjQ0MTE2MzI1OTE0NjA5MDEzYWExYTljNDM5ODFmNjM. > CSeq: 1 INVITE > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO Content-Type: application/sdp > User-Agent: X-Lite release 1103k stamp 53621 > Content-Length: 236 > v=0 > o=- 0 2 IN IP4 192.168.1.222 > s=CounterPath X-Lite 3.0 > c=IN IP4 > t=0 0 > m=audio 10006 RTP/AVP 0 101 > a=alt:1 1 : 9ImKPFaa h9RvD+sl 192.168.1.222 10006 > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=sendrecv Umm ... umm ... by ROUTER IP ADDRESS do you mean your public IP on your NAT router ? ... does your router have SIP-ALG activated ?, that could explain the problem. This INVITE is the one that just arrive at your proxy? ... if so .. your NAT router is doing sip-alg .. so mangling all the SIP dialog, and they usually does a very bad job with that. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parallel fork and mediaproxy => false behaviour?
On Friday 23 October 2009 14:14:52 Uwe Kastens wrote: > Hi, > > >>>> Is there an option to prevent this behavior with mediaproxy? > >>>> > >>>> opensips: 10.20.20.159 and 10.20.30.159 > >>>> UACs: 10.20.20.25 and 10.20.20.26 > >>>> UAS: 17.17.17.167 > >>> > >>> You have something wrong on you opensips.cfg, for sure ... We have lot > >>> of UAC's working on that scenario you described, without any problem. > >> > >> Good to hear that. > >> > >>> Are you using the dialog module? ... if yes, take into account it > >>> limitations to work with parallel forking. > >> > >> You are working not with engage_media_proxy() then? > > > > Of course, becasue engage_media_proxy NEEDS the dialog module ... and > > it's a know limitation of dialog module that it doesn't work AT ALL with > > parallel forking, or with multiple 1XX replies. > > > > Better if you limit your uses of the dialog module to the minimum ... let > > say ... to 0 .. ;-) > > At which time are you calling use_media_proxy() then? On the 1st INVITE > or later with the 200 OK with SDP? Is there any example out there? On First invite after checking if needed, on_reply for 1XX or 200 with SDP, on re-invites You could see and example on sipwise.com -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Transfer issue
Peter den Hartog escribió: > 77 is the sip trunk, 90 are my IP's. > So, it's your trunk provider the one who refuses to do the REFER ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Transfer issue
On Friday 23 October 2009 11:39:44 Peter den Hartog wrote: > I moved my opensips in the network, it's now directly connected to my sip > trunk, i can call inside, i can call outside. I can transfer inside. But > when i try to tranfser an outside nummer i get to see this ngrep: [..] > It makes sense to me that i forgot something in my config, a refer module > or something? any toughts/pushes in the right direction would be greatly > appreciated! As far as I see ... 77.73.226.254 it's refusing to do the REFER, what are you "inside" IP's, and what are you "outside" one ? ... -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] No solution with asterisk possible? New feature in TM? Re: parallel forking and CANCEL/BYE
On Friday 23 October 2009 11:25:57 Iñaki Baz Castillo wrote: > El Viernes, 23 de Octubre de 2009, Uwe Kastens escribió: > > Hi, > > > > > IMHO a proxy shouldn't behave as a UAC. Perhaps it can monitor dialogs > > > and so because this features just requires requests inspection, there > > > is no intrusion (adding a Record-Route parameter is not intrusion XD). > > > But behaving as an UAC is 100% intrusion. > > > > > > Yes, OpenSIPS is very flexible and can be used to solve some UA > > > problems, but the proxy shouldn't be the key for this purpose (IMHO). > > > > Ok. I am with you. > > > > But for example looking at the problem with mediaproxy (see email from > > this morning), opensips is doing to much or to less ATM. So > > mediaproxy/opensips will talk to the wrong SDP Ports, since its using > > the 2nd 200 OK with SDP from the UAC answer. > > Yes, I've replied to that mail right now. It seems to be a bug in > mediaproxy. No, it's not a bug in mediaproxy. I think it's a bug in engage_media_proxy(), I'm pretty sure ... because I use mediaproxy with parallel forking without any problem on the same scenario, but using use_mediaproxy -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parallel fork and mediaproxy = > false behaviour?
On Friday 23 October 2009 11:10:48 Uwe Kastens wrote: > Hi, > > >> Is there an option to prevent this behavior with mediaproxy? > >> > >> opensips: 10.20.20.159 and 10.20.30.159 > >> UACs: 10.20.20.25 and 10.20.20.26 > >> UAS: 17.17.17.167 > > > > You have something wrong on you opensips.cfg, for sure ... We have lot of > > UAC's working on that scenario you described, without any problem. > > Good to hear that. > > > Are you using the dialog module? ... if yes, take into account it > > limitations to work with parallel forking. > > You are working not with engage_media_proxy() then? Of course, becasue engage_media_proxy NEEDS the dialog module ... and it's a know limitation of dialog module that it doesn't work AT ALL with parallel forking, or with multiple 1XX replies. Better if you limit your uses of the dialog module to the minimum ... let say ... to 0 .. ;-) -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] One Way Audio Using X-Lite, OpenSIPS & Asterisk
On Friday 23 October 2009 11:01:11 Ross Beer wrote: > Hi, > > I am using the following config with Opensips and having a problem with one > way audio. When connecting the softphone directly to asterisk that runs on > the same machine audio passes without any problems. Firewalls are all open > and Zoiper/Snom phones connect without issue. > > If anyone could offer some advice it would be much appreciated as I'm > tearing my hear out :-) Better if you attach some SIP trace of a working an a non working call -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parallel fork and mediaproxy => false behaviour?
On Friday 23 October 2009 09:40:07 Uwe Kastens wrote: > Hello @all, > > My favorite szenario again:) > > Parallel forked INVITE to 2 UACs, both are sending back 200 OK with SDP, > UAS sends an bye for the 2nd call leg. Mediaproxy is using the SDP > information from the 2nd UACs, which has dropped the call already. > > Is there an option to prevent this behavior with mediaproxy? > > opensips: 10.20.20.159 and 10.20.30.159 > UACs: 10.20.20.25 and 10.20.20.26 > UAS: 17.17.17.167 You have something wrong on you opensips.cfg, for sure ... We have lot of UAC's working on that scenario you described, without any problem. Are you using the dialog module? ... if yes, take into account it limitations to work with parallel forking. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Audio problem
On Wednesday 21 October 2009 23:13:36 Justin L wrote: > Hi, > > I have a question related to my load balancing configuration of opensips. > > I have an X-Lite softphone that connects to Opensips server, which > transfers the INVITE request to one of the asterisk boxes. > All of them are behind firewall on the same network. Then asterisk calls to > my cell phone through the voip provider. > > The SIP balancing works fine and I get the call, but there is no audio. The > firewall should be configured correctly to transfer the SIP and RTP ports. > > Since I just started to use opensips it sounds to me like a very basic > problem, that many people probably have faced. > Could you please recommend me a way to troubleshoot this issue? > > Thanks a lot, > > Justin. Some SIP trace would be nice to begin ... -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Load balancer and Access control list
On Monday 19 October 2009 23:04:30 Peter P GMX wrote: > I now implemented a solution in Freeswitch. I created a whitelist based > on defined external gateways and IPs of registered UAs. > Now only whitelisted IPs are routed inside Freeswitch. > > I am looking forward to seeing the memcached implementation in the next > OpenSIPS release. That way I can share the whitelisted IPs quickly with > OpenSIPS and may block Invites in OpenSIPS based on a whitelist in > memcached. This is less CPU intensive. Buff ... so you like killing flyes with a bazokka ... ? ... well it's up to you. But IMHO, it's easier, faster, cleaner and scales better, if you use allow_trusted() from the opensips.cfg and work with your "whitelist" from the proxy POV, it's just and advice ... you will sleep better ;-) -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Load balancer and Access control list
Bogdan-Andrei Iancu escribió: > So, in such a case, a real working solution will be to have opensips to > spoof the source IP of the outbound request? but this may really break > the transaction Only if the spoofing is done with skipping the VIA > insertion. > > Regards, > Bogdan > Spoofing IP's is the worst solution, you could not asure that packages will arrive at the gw ... imagine there are firewalls between ... and lot of more problems. From my POV, if he need to go throught the proxy ... better to translate the acl validations to the proxy. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Load balancer and Access control list
Bogdan-Andrei Iancu escribió: > Peter, > > You can add a header with whatever name, depends more on FS what kind of > header is able to use to take the src IP from. > > Regards, > Bogdan > That will not run, FS as Asterisk, Yate and most of B2BUA/PBX outthere ..takes socket IP for validating it's ACL, as OpensSIPS does with the permissions/group module, of couse with a proxy one can chage that behavoir ... but usually with FS/Asterisk/Whatever .. that's not possible. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Load balancer and Access control list
Peter P GMX escribió:
> Hello,
>
> I tried the 302 redirect now in 2 different methods:
> $ru = "sip:06912345...@yy.yyy.yyy.163:5062"; # rewite complete
> contact address
> rewritehostport("yy.yyy.yyy.163:5062"); # rewite host
> and port
> and then
> sl_send_reply("302", "redirected");
>
> In all cases OpenSIPS sends back a 302 message with the new contact
> information yy.yyy.yyy.163.
> The originating UA then takes the contact information into consideration
> in the TO: header, however it still sends the message to the previous IP
> yy.yyy.yyy.165 (OpenSIPS again) instead of 163 (Freeswitch).
>
> I tried this with 2 SIP clients, they both behave the same.
> What am I doing wrong?
>
> Best regards
> Peter
>
If the UAC's are configured to ALWAYS use .165 as outbound proxy .. no
matter what you try with the 302 ... that will not run.
Trying to do loadbalancing with 302 or any other 30X code it's an error,
there will be lot of situations in witch it will not run.
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Re: [OpenSIPS-Users] Issue with incoming calls.
Peter den Hartog escribió: > Well, the server is in a network, i have a phone connected to that server in > the same network.. that network has 1 router, and an outside IP adres > connects to that router.. That's not a strange network solution right? I > mean, i don't see how that's diffrent to any home/office network out there? > > the ip adres is not directly connected to the server, it's connected to the > router. The outside phone calls come over that line. > I have not said anything about been or not a "normal" scenario, it is ... but you must know that SIP is not a firewall/nat friendly protocol, if you want it to work on you scenario ... that is, you recive calls from the "outside" network, you must setup your proxy as I told you. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Issue with incoming calls.
On Monday 19 October 2009 12:50:02 Peter den Hartog wrote: > Ok this is realy strange, now i've changed the router to a diffrent router, > same settings and now the phone DOES ring, but i can't answer it.. if i > press answer nothing happens, i can only reject the call. > > i'm beginning to think i forgot a port, or anything like that.. i just > opened up the port 5060 towards my opensips server.. is this enough? So, you are using opensips on a private IP ?, behing a NAT router? ... buff .. It have been discused lot of times ... that will give you TONS of problems. First: - You must use a NAT router that doesn't do SIP-ALG - You must redirect port 5060 or the port you are using for sip-signaling - You must modify your opensips.cfg to use your public IP as adversited_ip - You must add your public and private IP as local domains - You must pray all you know ... -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Issue with incoming calls.
On Monday 19 October 2009 09:44:22 Peter den Hartog wrote: > Bogdan, > > If you mean ping errors, i can ping everything. > > The sip trunk, comes in on a modem, (90.x.x.x) and goes to a router > (90.x.x.x -> 10.x.x.x) then the router routes every 5060 udp signal from > outside to the opensips server (10.x.x.x -> 10.x.x.x) the server has a > 10.x.x.x adres, so does the phone. > > I've added, the sip trunk as trusted domain, and i've added the router as > trusted domain. > But still, i get the to many hops message, and the exact same logging on > the sip call as above.. > > any ideas? "Too many hops" usually means that you are looping yourself. Have you checked with ngrep that you are not routing the petition over yourself again and again ? -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Media proxy basic setup
On Friday 16 October 2009 21:41:56 Daniel Goepp wrote: > Unfortunately yes, in this situation I am. The firewall is not PAT, but > rather a full 1 to 1 NAT. So any request coming in on port 1 or 10001 > will go straight to MP. From a network perspective I know it works as long > as the application is aware of what's going on. I'm able to get FreeSWITCH > running next to this and it handles the packets fine, but FS has a way to > tell it what the external IP is. Trust me, I know I'm fighting an uphill > battle here...but I'm trying to do some proof of concept work that would > depend on this setup. And at some point I will just admit defeat, I'm just > not there yet. I have gone through the source for MP, and it actually > doesn't seem like it would be too much to change, but unfortunately I'm no > Python expert. Changing mediaproxy code will not fix your problem, beleive me ... mediaproxy works inserting conntrack rules on the kernel, it doesn't open any port. Continue with the previous example ... RTP flow will be like .. UAC1 -> RTP to 222.222.222.222:1 -> NAT 1:1 Router -> 10.10.10.10:1 -> 10.10.10.10:10001 -> NAT Router -> UAC2 Who assure you that the NAT Router will map the outoing RTP stream as 222.222.222.222:10001 that is what UAC2 is specting ? anyone, because NAT 1:1 mapping are for INCOMING from outsideworld to insideworld ... but not for OUTGOING RTP flows .. Taking into account that on the near-end routers (UAC's side) will not "magicaly" open any port or something similiar ... you will end 99 of 100 times with a OWA (One Way Audio) situation. That's why mediaproxy and rtpproxy was designed to work on public IP's, for avoiding deadlocks on RTP forwarding and to be able to work even if near-end routers doesn't map RTP flow correctly. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Media proxy basic setup
On Friday 16 October 2009 18:51:12 Daniel Goepp wrote: > Yes, I do realize that I'm breaking the rules...but that never stopped me > from trying before. And yes, RTPproxy has the same problem, which is why I > decided to give media proxy a try to see if I could get it working, same > problem though as you note. I guess next I'm going to have to get in and > hack up some code...would seem a straight forward fix though, just have two > parameters in the config, one to offer and one to bind to. > > -dg Just a question ... are you trying to use mediaproxy on a private IP, but to announce a public IP? ... that's your goal? .. if yes, it will NEVER run, mediaproxy/rtpproxy was designed to work on public IP's. Try to image what will occurs on this situation: UAC -> Proxy MP With MP working on a private IP, let say ... 10.10.10.10 behing a NAT router ... let say 222.222.222.222 with the port range 10.000-20.000 redirected to mediaproxy IP. Now you setup mediaproxy to announce 222.222.222.222 as it's media-ip and let it bind to 10.10.10.10 (first of all, as you have seen that's not possible with current code) Now a nat-fixing request cames from the proxy ... and mediaproxy decides to use 1:10001 and you send out that on the SDP to your UAC's, so your UAC1 send it's RTP to 222.222.222.222:1 and your UAC2 should do the same but to 222.222.222.222:10001, how do you think that your NAT router would behalf ? ... wrongly on 99% of times, for sure. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Share Registrations Accross Servers
Ross Beer escribió:
> Hi,
>
> I have registrations replicated over two servers using a MySQL cluster
> and have also tried replication at SIP level however I can not call
> between servers.
>
> The setup is as follows:
>
>
> Soft phone A Reg > OpenSips A
> |
> Soft A calls Soft B
> |
> Soft phone B Reg > OpenSips B
>
>
> NAT checks are performed at the point of registration and when calls
> are being made.
>
> It doesn't look like the request from Soft pone A that is passed to
> OpenSips A is going any further than OpenSips A. I can see that it
> looks up the location of soft phone B and tries to call it directly
> instead of looking a the 'socket' and passing the call to OpenSips B.
>
> Do I have to perform any checks to see if the socket is remote and
> then forward the request instead of just doing a lookup("location")?
> If so can some one point me in the right direction :-)
>
> Thanks,
>
> Ross
You should use the add_path() functions before using t_replicate, and
also all the related stuff to the path module
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Re: [OpenSIPS-Users] ERROR:mediaproxy:use_media_proxy: insufficient ports returned from mediaproxy: got 1, expected 2
On Viernes, 16 de Octubre de 2009 07:37:27 Krunal Patel escribió: > Hello, > > I have given 1:2 port range in mediaproxy. > We are having almost 1500 registered users. > Even few of them are behind NAT. > So logically there can be maximum 3000 open ports. No so "logically" ... take into account that a call may have more than 1 RTP stream, also take into account that usually if there is re-invites during the call, new ports are needed, not to mention that when a port is "closed" by mediaproxy, may be marked by the kernel as "not available" during some time. > Would you please let me know what else can be the cause of the issue? > Meanwhile, I am checking the source code, If I could find anything helpful. As mediaproxy now uses conntrack rules for doing RTP forwarding, try to use conntrack to check how many rules are inserted and are live on your system when the error occurs. -- Raúl Alexis Betancor Santana Dimensión Virtual S.L. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] parallel forking and CANCEL/BYE
On Jueves, 15 de Octubre de 2009 10:52:12 Uwe Kastens escribió: > Hi, > > I am using opensips to fork calls to UAs which are registrered from > different IPs/Ports. > > If one UA accepts the INVITE the other UAs will get a CANCEL. > > Now I have one subscriber with 2 asterisk server which asked me to send > a BYE after the CANCEL. Otherwise he wants me to send an BYE which could > not be processed correctly on the opensips. > > I am pretty sure, that this kind of handling would not be RFC conform > and so its not possible to handle this inside the routing script. Or did > I missed something? You are wright, that will be non-RFC conform ... moreover .. I don't undestand why your subscriber needs that ... because Asterisk is non-RFC conform on lot of things .. but not on that one. -- Raúl Alexis Betancor Santana Dimensión Virtual S.L. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPs behind a firewall
On Jueves, 15 de Octubre de 2009 00:31:10 Daniel Goepp escribió: > Route: Have you declared as a local domain?, if not ... OpenSIP will try to route it, so thats were you have the loop of the ACK. -- Raúl Alexis Betancor Santana Dimensión Virtual S.L. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Loadbalancer Query
On Jueves, 8 de Octubre de 2009 14:50:30 ASHWINI NAIDU escribió: > Hi all, > > I wanted to know how many stateful transactions can opensips handle if > loadbalancer is involved. That's a too generic question, you have to take into account lot of more things to do that estimation -- Raúl Alexis Betancor Santana Dimensión Virtual S.L. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPs behind a firewall
On Wednesday 07 October 2009 22:56:06 Daniel Goepp wrote: > Thanks for the information, that seems to at least change the Via, but > RTPproxy is still not handling this well. It will only give it's > private IP back to OpenSIPS, so I end up wtih: > > Connection Information (c): IN IP4 10.250.7.164 > > I will try and see if I can work around this on the RTPproxy side of > things. You will have tons and tons of problems runing an rtpproxy on a private IP, more than the ones you will be able to solve. RTPProxy was designed to BE on a public IP. Ask your provider to give you some sort of /29 or /28 IP pool and work with that, otherwise you will end with headhache. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] List of online users
On Thursday 01 October 2009 05:31:43 smadhoo6 wrote: > Oh yeah!.. I am developing a SIP-Client that will be using this > information.. using pjsip.. > Once I know how I 'll be able to send this information.. I can make changes > in the client accordingly.. :) > Thanks! If you are developing a SIP-Client ... better to follow the standars and just use SUBSCRIBE to retreive other users status -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] List of online users
On Wednesday 30 September 2009 12:05:51 smadhoo6 wrote: > Hi, > I want to send this information(ie., the subscribers list) to all the > registered users within that domain.. That is useless ..., if you are not using a custom-developed SIP-client that expect that information ... any SIP-compliant UAC will refuse that information and you will be wasting your time and resources. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Polycom 650 can't Register behind NAT
On Sunday 27 September 2009 06:11:53 osiris123d wrote: > So I went to the web GUI clicked on SIP and then clicked on Local Settings > and set the "Local SIP Port" to be 60111, but the polycom is still not > registering. Here is the ngrep output now. So, again ... you proxy it's not challenging the polycom on the 401 response ... that's why you polycom do not send it credentials and never get registered. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Polycom 650 can't Register behind NAT
On Sunday 27 September 2009 04:46:52 osiris123d wrote: > I am sure this is not a OpenSIPs issue, but I was just wondering if people > have run into this and what they have done to resolve this. I currently > have been testing my OpenSIPs config with Counterpaths Bria client and also > the SJphone. They don't have any issues registering behind NAT. I just > purchases a Polycom 650 phone and I cannot get it to register. When I do > an ngrep I see that the phone is wanting OpenSIPS to talk back to it via > port 5060. When I do an ngrep with Bria or SJphone I see that when > OpenSIPS replies back to the softphone it is not on port 5060. It's not a matter of on witch port the Polycom listen ... in your ngrep trace, the polycom SHOULD ACK the 401 and it isn't do it. Also ... the 401 should contain challenge information from the proxy and it isn't ... so your proxy it's not doing it work well ... our you NAT router it's doing some kind of SIP ALG and disturbing the normal working of the proxy. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Multi domain registration.
On Monday 14 September 2009 09:11:23 Chris Maciejewski wrote: > Hi Saúl, thanks for suggestion, but this is exactly the code I currently > use. I posted it in my previous message in this thread: > http://lists.opensips.org/pipermail/users/2009-September/008034.html > > ...unfortunately it doesn't work the way I want. It works, maybe you have not set some options of the modules that let it works but It works, trust me ;-) -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy relay on none default route interface failed
On Thursday 10 September 2009 22:25:39 Dan Pascu wrote: > On 10 Sep 2009, at 18:00, Raúl Alexis Betancor Santana wrote: > > So ... "killing flies with bazookas" ... > > So ... when having to update, need to update n instances, instead of > > only > > one .. > > So ... wasting hard disk (I know it's cheap ... but wasting > > anyway) ... > > I'm pretty sure you felt smart and clever while wording your indignant > reply. Indeed, you absolutely had to put me into my rightful place and > I deserved every dot and comma in it. What was in my mind trying to > suggest some alternatives that work with the software as it is and do > not require patching or a new version? But for a strange reason I > cannot pinpoint, I do not feel motivated by your rhetoric. No sir. > Shamefully I must admit that it made me think that I can spend my time > better elsewhere. Imagine that good sir. I'm so ashamed. It also made > me have unclean thoughts about you. It made me think that maybe you > have a size problem and you're compensating. Can you imagine that? So, > good sir, maybe you should fire more of your aggressive indignation > towards me to break this unclean thinking in my mind and finally make > me see the light. Forget about asking nicely or, God forbid, > contributing back. That's for pussies and it only weakens one's > character. Every time we have "talked" about this ... you always react the same ... as If I was fighting against you ... I don't know from where you get that feeling from my words. I'm only expressing my opinion, your are free to ignore it if you don't like it. Yes you gave a solucion and I arge that a better solucion (and from a desing point of view and from the KISS principles, easier) it's so simple add a -c config param. Do you want the patch? ... they are just 4 lines, I did it for 2.3.4 version of mediaproxy, also for callcontrol and other very good software you have developed but that lacks THAT SIMPLE option, to be able to set the config file to use from the command line ... as 99% of other software allow you. > > I still think that a simple param like -c configfile is better. > > That may be, but I made a principle of not helping people that hate me > or abuse me verbally. I don't think I have abused you ... if you think soo it's your problem. I only express my opinion and comment other possible solutions. > Besides, free in "free software" doesn't stand > for let's use that software that is free of charge and then feel free > to abuse the developer. It stands for fell free to improve it and > contribute back. Wait, scratch that. From you I wouldn't even want > patches. The contribution you made is already more than enough... Dan, sincerily .. you don't like to hear opinions that are agains yours ... and that's not good. If you don't want my patches, you are free to ignore them. Not always a patch is needed to improve a software or to contribute to it, most of time another point of view about a problem it's enought. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy relay on none default route interface failed
On Thursday 10 September 2009 15:02:21 Dan Pascu wrote: > You don't need to patch anything. Just unpack mediaproxy in as many > different directories as you need, run ./build_inplace and modify each > config.ini in those directories as needed. Then run mediaproxy from > those directories and each of them will use the local config.ini from > its own directory. So ... "killing flies with bazookas" ... So ... when having to update, need to update n instances, instead of only one .. So ... wasting hard disk (I know it's cheap ... but wasting anyway) ... I think it's easier to patch media-relay to be able to pass the config file as a param, as on many other programs around the world .. > Alternatively, if you want to use a system wide installation, you can > copy the binaries from /usr/bin to a number of different directories > and add a config.ini in each directory. Then run those binaries from > those directories instead of /usr/bin/ and each binary will use the > config.ini file in its own directory to overwrite settings from the > global /etc/mediaproxy/config.ini. ... no coments > Mediaproxy uses 2 configuration files. The global one resides in /etc/ > mediaproxy/config.ini. On top of that if a config.ini is present in > the same directory as the binary (media-relay & media-dispatcher) that > one will be used to overwrite the settings from the global one having > priority over it. I have no tested that ... but anyway it requires me to been duplicating installations ... I still think that a simple param like -c configfile is better. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mediaproxy relay on none default route interface failed
On Thursday 10 September 2009 11:56:00 Ghaith ALKAYYEM wrote: > Hello, > > I think it's not possible to use two separate relays on the same server, > I tried that a lot then I switched to RTPproxy. That's not true, you could run as many Realys as you want on the same server, only have to patch mediaproxy-relay to be able to call it with a diferent .cfg as the default one, have diferent listen ports and no more. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mailing list reply-to : list or sender
On Wednesday 05 August 2009 17:09:56 Dan Pascu wrote: > On Wednesday 05 August 2009, Raúl Alexis Betancor Santana wrote: > > On Wednesday 05 August 2009 14:58:32 Brett Nemeroff wrote: > > > And if you set to "The List", it's very hard to reply to the poster. > > > Since neither reply to reply-all will get you the email of the poster. > > > > I don't know any mail client that when you hit "reply all", don't send > > the reply to ALL, including list and poster, at least all common MUA does > > it that way. > > Indeed. And how does that apply to what Brett (and everybody else > advocating "Poster") said, that he wants to reply to the poster alone? I don't know if I undestood you question ..., what they told is that they allways use "Reply all", for sending to "the list" and to "the poster", and what I told is that for me it's more logical to use "Reply" and been sended to the list. Nowadays if you use "Reply", you will reply only to the poster. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mailing list reply-to : list or sender
On Wednesday 05 August 2009 17:03:01 Dan Pascu wrote: > On Wednesday 05 August 2009, Raúl Alexis Betancor Santana wrote: > > On Wednesday 05 August 2009 10:07:47 Dan Pascu wrote: > > > > I have no problem to setup my Kmail to work with the "List mode", I'll > > doit as most of list-admins don't want to change that. > > That's contradictory. You first said that 99% of the lists you're on use > "The List" and now you say that you have to setup kmail to automatically > reply to the list because _most_ list-admins don't want to change it like > that?!? NO, you missundestood me, when I said "_most_ list-admins don't want to change it" I was talking about THIS LIST ADMINS. 99% of the lists I'm subscribed to work as I described, using "The List" mode, in fact .. only this list is the one that doesn't do it in "Poster" mode. > Have you ever wondered why list-admins don't want to change that? As I told, I mean opensips-users-admins so this question have no sence, because on all the other lists I'm member of work in "List mode" > "Some" are you and another user. Everybody else found "Poster" to be a more > natural setting that gets less in the way and it's easier to use. I have count 2 more users appart from me, don't know how many users are subscribed now, but I hope we are more than the 5 that have written on this thread. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mailing list reply-to : list or sender
On Wednesday 05 August 2009 15:56:19 Brett Nemeroff wrote: > Actually,I just checked with gmail on a list with "The List" set and when > you hit "reply all" it ONLY places the list address in the To: X-DDD ..., sorry I must laugh ..., Gmail it's a webmail with lot of funcy "features", but also well know because of all the things it doesn't handle very well as other MUA does. Gmail doesn't handle threads on a friendly way, so only for that, it isn't very good for mananing maillists mails. Also Gmail doesn't handle at all, the extra headers that mailmain and other maillist managers add, so that why when you do a "Reply all" with gmail, it doesn't include all address on the reply. All this things about gmail are well documented on Google LABS forums, they have been requested as "basic features" by lot of users. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mailing list reply-to : list or sender
On Wednesday 05 August 2009 14:58:32 Brett Nemeroff wrote: > I agree with you here. Even if it's "different" than the way most of your > other lists work, "Poster" simply makes more sense. And I don't think > another way should be picked because the poster might do the wrong thing. I > think it's up to all of us to reply to the list as appropriate. > For me, If I'm reading a message from a specific person, and I hit "reply" > I'm replying to that person. If I hit "reply to all" I'm replying to > everyone. This just makes sense to me. I don't agree here, that my POV > And if you set to "The List", it's very hard to reply to the poster. Since > neither reply to reply-all will get you the email of the poster. I don't know any mail client that when you hit "reply all", don't send the reply to ALL, including list and poster, at least all common MUA does it that way. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Contact header
Alex Balashov wrote: > If we are going to have a cultured and dignified relationship between > the Kamailio and OpenSIPS camps, which I assume is the goal of everyone > for reasons of commercial self-preservation if nothing else, then the > provocations need to stop from both sides. I Agree. But also thinks that talking about what I consider a big limitation of the dialog module is not a provocation. > No, it is not very upstanding to come on the OpenSIPS list only to > remind its members that you don't use OpenSIPS and that Kamailio is much > better. Whether you think it's true or not, the OpenSIPS list is not > the appropriate forum in which to air that thought; it's just not > polite. Sorry, but I have not come here telling that, what I said at the beginning of this post was about dialog module and it's limitations, then comented that I use Kamailio, not OpenSIPS, but I NEVER said anything about one been better than the other, only that I use the other, just that. > The values and focus of every community must be respected, and this > mailing list belongs to the OpenSIPS community and development team. > There's a certain degree of "when in Rome..." that should be obeyed. But while "standing in Rome ..." could why talk about that "Fontana di Trevi" should be cleaned or not ?, because I haven't say anything about not to use dialog module for other purposes, I told than on my enviroments I could not use it because it have some limitations, just that. > I'm a very committed Debian user, and intensely dislike Redhat-derived > distributions. But if I am on a mailing list centered chiefly around > Fedora, CentOS, RHEL, etc. or products based on them, it's just not my > place to bring up Debian or invite an RH vs. Debian flame war. That's > just not what the list is for, and my ability to join it and ask a > question is a privilege, not a right. I Agree with you, but again I think that I have not begin any flame war, appart from the one about dialog module. I have not said nothing about other features/bugs, was only talking about dialog module. > 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. Full Agree. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Contact header
On Wednesday 05 August 2009 08:54:27 Adrian Georgescu wrote: > Raul, > > Your recurrent negative comments on this mailing list about things you > do not even use are absolutely disgusting. I use them, but not in production enviroments, because they are useless to me in their actual implementation, that's why I made comments about them. > Keep your fancy opinions > for yourself or take them some place else where people enjoy what you > have to say. Never ever in my life, you don't have to be agree with my opinions (it obvious that you don't), but I'll not shut up if I think I have something to say that could help (I said that dialog module have limitations, I said that I tested it, so next step is to try to fix that limitations and send a patch, that's something I have on my TODO) -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mailing list reply-to : list or sender
On Wednesday 05 August 2009 10:07:47 Dan Pascu wrote: > That's in your world. I was never on any mailing list that used "The List". Uff ... I don't know what to reply to this statement ... nowdays I'm on more than 60 list, including Debian Project ones, Kernel, VLC and some other high traffic lists, NONE of them use "The Poster" > However I burned myself when I set the email client to reply to the list by > default for a folder that was for a mailing list and I unwillingly sent a > private email to the list. I find this to be more problematic than the fact > that I may have to resend. "The List" creates more problems than it solves. I have no problem to setup my Kmail to work with the "List mode", I'll doit as most of list-admins don't want to change that. Some users have said they prefer "List mode", maybe if more users reply we could reach and agreement or not if admins don't want at all. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Contact header
On Wednesday 05 August 2009 00:07:30 Dan Pascu wrote: > On Tuesday 04 August 2009, Raúl Alexis Betancor Santana wrote: > > I wonder how other could use dialog module and other modules that depends > > on it with such "limitations" ... > > That's because you lack imagination and because your purpose doesn't seem > to be to have a problem solved or to help solve a problem but to poke > around with a stick. Too hard answer to a simple question, I don't have any lack of imagination, moreover I'm full of it. I try to follow KISS principles when designing a system, not only to "try to solve it as soon as possible, no matter the way and without thinking about future problems ..." For example, one of the uses of dialog module is to "limit" the number of concurrent dialogs allowed to a certain user (using some kind of "profiling" for example), so ... if dialog module does not handle correctly early dialogs and forking, how could I use it in an enviroment where I could have lot of UAC's registered with the same credentials? it's imposible, because dialog module will do a mesh with the parallel forking. It's just an example, there could be lot of them. > If the dialog limitations bother you so much, be a chap and lend a hand in > improving it, or simply ignore it if you don't care. But your whining is > useless and helps nothing unless your purpose here is to throw shit over a > fan and watch it splash. One way to improve the module is to let developers known REAL limitations on other enviroments more than the ones they usualy work on. Testing it's other way to help ... as far as I know, and that is what I do. It wasn't my intention "to throw shit over a fan", as you said. Just to let you know where problems are. This is free software, so it's up to you if you want to implement it/solve it or let the module as it is now and let others (like me) fix that part, but you have to be open to criticize, if not ... better not to expose yourself to the open world. I don't mean you have to "eat all the people throw you", I just mean that you have to be open minded. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] mailing list reply-to : list or sender
On Tuesday 04 August 2009 23:41:57 Dan Pascu wrote: > On Wednesday 05 August 2009, Bogdan-Andrei Iancu wrote: > > "normal" is a fuzzy word. I can do the change, not a problem, but just > > to past what mailman says: > > > > "Where are replies to list messages directed? Poster is *strongly* > > recommended for most mailing lists. -> Poster This list > > Explicit address " > > > > Currently is "Poster"(which is recommended) and you suggest "This List". > > > > To be honest I consider logical the "Poster" setting as primarily I'm > > talking with a person on the list and not to the list. > > > > But any other opinions are welcome here. > > "The List" is a worse choice than "Poster" because it limits my choice of > whom I want to reply to and makes it more complicated to reply to the > sender alone. When using "The List", if I want to reply to the sender > alone, I need to employ manual copy/paste of his address, after I manually > delete the list address which was automatically added when replying. This > is awkward and annoying. > > With "Poster", I can easily use reply, to reply to the sender alone, or > "reply all" to send to both the poster and the list and the list will make > sure the poster will receive a single copy anyway. > > I don't think nowadays it could be an issue to anyone, as I can't imagine a > mail user agent without a "Reply all" function. 99% of list I'm member of works as "The List", because on 99% of replies you want to send are to the list. It's not a matter of a mail client having or not "Reply All" function, it's a matter of logic, if I'm in a list and I talking on a thread I want "to Reply" to the list .. not to one participant that reply my mail, if I want that the poster recives an annoying copy (because, as far as I remember, noone could send a message to the list if it isn't subscribed, so it will allways receive a copy from the list) ... I could allways use "Reply All". Well .. that's just my opinion. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Contact header
On Tuesday 04 August 2009 21:15:54 you wrote: > saying that the dialog module is broken is an over-statement - I admit > there are some issues that still need to be worked out in the module > (forking and early dialogs) - see the thread on the devel mailing list, > but otherwise the module is pretty fine. For what I try to use it .. if it have no support for earsly dialogs and forking ... it's broken and it's useless (for me) I wonder how other could use dialog module and other modules that depends on it with such "limitations" ... > >> The good news is many if not all of these features have an "off" switch > >> for the uninterested parties to throw. It's not often that everyone > >> wins. > > > > I know, maybe someday in a near (or not so near) future, I could use some > > of that fancy features, by now I prefer to have a > > SIP-Proxy+Registrar+Presence on Kamailio (sorry I don't use OpenSIPS > > That explains why you have such a good opinion over the dialog module > ;). So I understand your statements on the OpenSIPS dialog module is > done without actually testing it... Your are wrong here, both modules have same problems, in fact Kamailio module have some fixes I have not seen on OpenSIPS one. I follow and test both projects, I prefer to use kamailio and I port some modules/fixes from OpenSIPS to Kamailio, so I know what I'm speaking about :-) > you can do the same think in many ways. Depends of what you aim to do > and how suitable to components you are using are for your purpose - a > B2BUA is a powerful component (and intensive resource consumer) you can > use for changing a header in a request or for doing complex call/media > bridging . Or for doing accurate accounting (and please .. let us not beging that thread again, each one have it's own oppinion about this ...), or for T.38 support, or many other things. We use Callweaver as B2BUA just for PSTN relaying, accouting and T.38 support, for other things as VoiceMail,VirtualPBX, Conferences, IVR, etc. we use SEMS. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Fwd: Re: Contact header
Please ... could be possible to setup the mailling list as "normal" mailling lists that with "Reply" just send the reply to the list and not to personal inbox .. ? -- Raúl Alexis Betancor Santana Dimensión Virtual --- Begin Message --- On Tuesday 04 August 2009 21:08:03 you wrote: > > Maybe it's because we use a topology on witch ANYONE could directly reach > > our GW's (at SIP Level), only our proxies, so for us (I'm always speaking > > from the point of view of our arquitecture), it's not importat if you see > > the IP's of the GW's on the replies (are OUR GW's, not anyone else) > > that is more or less true - even if your GW is not directly accessible, > your proxy may have a flow that allow an attacker to route (through your > proxy) calls to GW. Only if we made a mistake on our proxies scripts ... by now .. we have not found any hole :-) > > Also, consider that in most of the case you do not have direct control > of the GW (wholesale providers) and you do routing over public internet. > > >> Or if you are doing wholesale PSTN termination, you might want to hide > >> the PSTN providers you are working with (business protection).. > > > > We hide our PSTN providers ... behind our GW's :-) > > well, after all you do hide :P Yes, I don't said that we do not did it ... just that we do no do it at proxy level, it's not it task (from our POV) > I would not go for "broken"I admit there are some issue with the > dialog module (mainly related to the early dialogs - see the thread on > devel list), but not critical ones There more issues than the ones related to early dialogs, I'm follow the devel list about that. -- Raúl Alexis Betancor Santana Dimensión Virtual --- End Message --- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Contact header
On Tuesday 04 August 2009 13:03:02 Jeff Pyle wrote: > Raul, perhaps I can explain from the perspective of someone who may benefit > from such misplaced features. > "Better" is in the eye of the beholder. Or, perhaps in these cases, in the > eye of the implementer. If topology hiding is the only B2BUA-style feature > I need and all my other requirements are satisfied by a proxy like > Opensips, why would I re-engineer my whole system on a different platform? Maybe because "hidding" module of OpenSIPS is based on a broken dialog module ? ... that's a good reason for me. > The good news is many if not all of these features have an "off" switch for > the uninterested parties to throw. It's not often that everyone wins. I know, maybe someday in a near (or not so near) future, I could use some of that fancy features, by now I prefer to have a SIP-Proxy+Registrar+Presence on Kamailio (sorry I don't use OpenSIPS) and B2BUA features on Callweaver GW's -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Contact header
On Tuesday 04 August 2009 03:44:33 Bogdan-Andrei Iancu wrote: > Hi Michel, > > You can not change the IP in the contact without breaking the routing of > the sequential requests. It might be possible by using dialog support > for these calls and to store the contact into the dialog and to restore > it laterbut it is a manual work. > > Soon, a topology hiding module will be available for such purposes. I wonder why people want to do such strange things ... no matter .. Also, a "topology hidding module" ? ... for what?, I think that such kind of "features" are very useless, IMHO. If people want to hide it's topology, better to use a B2BUA and not a proxy. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CDRTool : Memory error
On Monday 27 July 2009 11:22:22 Yves Premel-Cabic wrote: > Hello, > > I have some new problem with CDRTool : when I was testing my conf, i > didn't load heavy destinations/billing_rates tables. Now I'm installing > all this stuff in prod, I loaded a 200 000 entries mysql table in > "destinations" as well as in billing_rates. I restarted CDRTool, but it > never started again : > > *Fatal error*: Allowed memory size of 134217728 bytes exhausted (tried > to allocate 71 bytes) in */var/www/CDRTool/library/cdr_generic.php* on > line *527 > > *and this line 527 is a part of the function load destinations > > I guess this is a memory allocation problem as it said, but can somebody > tell me what is exactly the problem? It seems cdrtool doesn't manage to > cache the needed infos, but my server has something like 4 GB ram... You should increase you max memory allowed to be used by a php script, check your php.ini files -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Upgrade version problems in a production system (mediaproxy)
On Monday 27 July 2009 10:01:45 Adrian Georgescu wrote: > Inaki, > > If this discusion took place two years ago you would have said the > opposite simply because the 'stable' distribution was so obsolete. Now > the stable is pretty much in sync (still) with unstable hence it seems > a good idea to use stable at this very moment. If we look again over > 12 and 24 months, maybe then we have a better measurement. > > Adrian Never, on production enviroments ALLWAYS Debian stable, and if I have to backport, I do it .. I backported postgresl 8.3 with pgcluster support to a Debian Stable more than a year ago ... and that's a big job, doing the same with AG soft are very, very simple tasks it we compare them. But that are just POV's. I have no problem at all if AG continues to use unstable ... it's up to you and your customers, I'll port the packages I need to my working enviroments. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Upgrade version problems in a production system (mediaproxy)
On Monday 27 July 2009 09:54:42 Iñaki Baz Castillo wrote: > 2009/7/27 Adrian Georgescu : > > The software we are talking about, is natively developed, tested and > > deployed on unstable distribution. > > On *which* unstable distribution? on yesterday version? or on a > version of a week/month ago? That's the problem of "ustable" ... today unstable maybe not the same as this afternoon unstable ... an so on. > I've found a bug in libdbd-mysql-perl package which creates apache > coredumps [*]. Perhaps OpenXCAP runs Ok in that *exact* Debian > unstable version, but people also requiring apache and mod_perl would > have a critical problem. > > [*] http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=520406 There are other relevant bugs open. > Sincerelly, is really to hard to build AG python packages to work on > Debian Stable? (Lenny is not so old, is it?). No ... just dpkg-buildpackage and go on ..., Lenny it's a month old, no more. > For example: I have OpenXCAP 0.10 running on Lenny and the only I > needed to install is the python-application (AG package) version > 1.1.1-1 since Lenny version is too old (1.0.9-4). Just it. For sure I > prefer to do that instead of upgrading my system to Debian unstable > and leave other applications running with critical bugs. Most, if not all AG app's could be used on Lenny or Sid with only updating python-application. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Upgrade version problems in a production system (mediaproxy)
On Monday 27 July 2009 09:34:50 Adrian Georgescu wrote: > > My question is: can I use a debian unstable in production "mission- > > critical" telephony systems? > > We do use only unstable for more than five years now for highly > available SIP platforms, if this gives you some indication of > reliability. That's more than walking on the blending edge, just my opinion. With my personal systems I do really scrapy things ... "waling far away from the blending egde" ... but not with production eviroments. > The software we are talking about, is natively developed, tested and > deployed on unstable distribution. So the fact that it works on stable > today is a matter of temporary luck No, it's a matter of been well done, moreover taking into account that are python scripts, not C or C++ code that depends on lot of libs. > (the previous debian stable was > several years old, nothing recent enough worked on it). That's not true, lattest Debian Stable release 5.0r0 that is from 6/29/2009, so not "years ago" as you suggested. > Based on > experience, sonner or later some library will not be available in > stable and you will need to repackage all dependency chain by > yourself. And that's an admin task ... so what's the problem with this? > The amount of incremental work you will need to do, will > exceed the 'safe feeling' you get from running a so called stable > distribution. Also not true, at least with Debian. > There is no magic bullet, 'stable' does not mean it will > be trouble free especially when it comes to running such bleeding edge > software. Mediaproxy it's not a blending edge software (from my point of view), it's true that NOTHING could be called "stable" for sure, but on Debian unstable, packages are not warranted to run, ever to install without crashing all the system (on a normal way, that does not occurs, but you don't have any warrant of so) But this is only a matter of POV's. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Upgrade version problems in a production system (mediaproxy)
On Monday 27 July 2009 08:09:14 Carlo Dimaggio wrote: > > Hi Raúl, > > Can you share this task? How do you satisfy the dependencies? We have a debian stable "build server" on witch we compile all we need for our debian stable systems, then we test that packages on preproduction enviroment before installing on the production ones. That's all. Most of times it's a matter of getting the .tar.gz, .diff and .dsc files of the packages we need to port and then on the build system: dpkg -x package1.dsc cd package1-1.0.0 dpkg-buildpackage -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Upgrade version problems in a production system (mediaproxy)
On Monday 27 July 2009 08:05:38 Carlo Dimaggio wrote: > My question is: can I use a debian unstable in production "mission- > critical" telephony systems? NO, I don't suggest you to do so if you don't want to be in troubles > Now, with these statements, my choice is: use a debian unstable or use > an ubuntu system without the possibility to upgrade mediaproxy (and in > future, other ag-projects softwares)... Better a Debian stable. > How other people keep their not-debian systems updated? By hand, using debian stable and backporting the packages you need (just libconntrack and python-application, as far as I remember) -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Upgrade version problems in a production system (mediaproxy)
On Friday 24 July 2009 16:49:40 Adrian Georgescu wrote: > Carlo, > > We use only Debian development branch both for the development of our > software and for our deployments. If you use Debian unstable you are > the safest as we are testing and using our packages only this > environment. > > While the name 'stable' brings some confidence for the inexperienced, > in practice when it comes to bleeding edge developments like we do, > the stable version can become obsolete fast and you could be stuck by > running it because of the dependencies that do not make it to the > stable branch as soon as we develop our software. When things do not > work you will hear the same story "your distribution is too old" to > run this software. > > So the most stable distribution is unstable. That's too wilde Adrian !!! ... and abosolut wrong, I don't agree at all. All the systems I run/install run Debian stable, and if I need some blending edge soft I backport it to stable. Backporting mediaproxy it's a very simple task, as it's a "very simple" dependency software. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] unix odbc problem
On Monday 20 July 2009 12:23:29 ASHWINI NAIDU wrote: > hi Inaki > > if you are reffering to the > mysql://opensips:opensip...@localhost/opensips line in opensips.cfg file > and DBHodt in opensips ctlrc both are edited respectively And what have you put there ?, also take into account that EACH module that use DB backend, have an db_url param ... if you read carefully the log you attached, you could see that the group module is the one that it's giving you the error, have you checked what do you have in the db_url param of the group module ? -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Regarding implementing presence
On Thursday 16 July 2009 23:52:49 Srikanth Rajagopalan wrote: > Hi, > > I first used opensiops to implement a sip server without any presence > capabilities. Now I needed to add presence. > My clients are implemented using Pjsip. I try to subscribing to one another > and they turn out successful. But when I check out the database I dont see > any entry in the active_watchers table or watchers table. > In the log file I saw the error > > Can't Connect to local mysql server through socket > /var/lib/mysql/mysql.sock > I think this might be because of maybe the changes I made to the config > file while trying to enable presence > These are the changes I made > > Uncommented out the 2 loadmodule lines > > Uncommented the 3 modparam lines related to presence > > In the modparam line with db_url I changed the url to > "mysql://opensips:opensip...@localhost:mysql/opensips" > Removed the lones related to presence from the exclude modules in Makefile > > Finally I made it using include_modules including all presence related > modules > Also, next the clients tried to publish some info and I got the error > message 503: service unavailable > But the other client still received Notify messages. > > I am not sure if that was because of opensips or pjsip. > > It would really be helpful if some one could guide me where I am wrong. The problem it's obvious, you don't have mysql server running localy or the path to the socket file it's incorrect. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Asterisk reinvite confuses Mediaproxy
On Sunday 12 July 2009 00:48:26 Jeff Pyle wrote: > On 7/11/09 5:33 PM, "Raúl Alexis Betancor Santana" > > wrote: > > I will not question why are you trying to use Mediaproxy if not for NAT > > fixing .. X-) > > Because I can. :) Or, rather, I thought I could... Real world > applications include more accurate accounting and documenting the codecs in > use. For accurate accounting I prefer to use a B2BUA, for documenting the codecs in use, that could be done, just with the Proxy, but you get a good idea :-) > > For properly handling the re-invite, did you call force_rtp_proxy INSIDE > > the in-dialog procces ? > > force_rtp_proxy() is for rtpproxy; this is Mediaproxy. I told you without reading the docs ... just from memory. > I'm calling only > engage_media_proxy() at the initial INVITE. I suggest you to stay appart from the dialog module as far as you can. > Perhaps I might have more luck > with the less automated use_media_proxy()/end_media_session() approach. Yes, that is a better way. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Asterisk reinvite confuses Mediaproxy
On Saturday 11 July 2009 22:16:27 you wrote: > Yeah, I suppose so... :) There is no NAT here, however. All public IPs. > The root of the issue isn't NAT fixing, it's that Mediaproxy doesn't seem > to use properly the new information from a reinvite. I will not question why are you trying to use Mediaproxy if not for NAT fixing .. X-) > Failing call flow is: > PSTN Gateway -> Asterisk w/ reinvites -> Opensips -> SIP Phone* > > * Note: "SIP Phone" is really an Asterisk box with a Polycom behind it, > but it's not doing anything screwy. No reinvites from this one. I can > reproduce the same behavior with a Sipura or Polycom registered directly to > Opensips. It's just much harder to test because I don't have any extra > public IPs available in my "home" lab. For properly handling the re-invite, did you call force_rtp_proxy INSIDE the in-dialog procces ? -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Asterisk reinvite confuses Mediaproxy
On Saturday 11 July 2009 20:18:06 Jeff Pyle wrote: > Ngrep output is attached. It looks okay to me. > > I've indicated which IP addresses belong to which entity at the top of the > file, and commented who's talking to whom at the # before the packet in an > attempt to improve readability. If you want to improve readability .. just don't use IP's from a same range in a capture that is supposed to be about NAT fixing ... ;-) After a first read ... your call flow is Asterisk -> OpenSIPS -> Asterisk -> SIP Phone ? ... or Asterisk -> OpenSIPS -> SIP Phone (beging the same NAT router as Asterisk) ? -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] How To Ask Questions The Smart Way
On Friday 10 July 2009 12:33:31 li...@grounded.net wrote: > Bunch of self important blowhards, this is the only mailing list that acts > this way! That's maybe your are not subscribed to technical lists, because on my more than 18 year old internet knowleadge, on all technicnal (and on non technical also) list I'm been subscrided to, work on the same way, anyone is forced to answer you. Here, more than one of us have told you to read some docs and guide you to the path you should follow to get the NEEDED knowleage and enought background to be able to reach your goals. You could ask doubds, post confings and traces of what you don't get running and I'm pretty sure you will get answers that will help you, but if you insists on doing "too general" questions you will allways get the same answers. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Is opensips a front end to asterisk?
On Thursday 09 July 2009 18:53:19 Brett Nemeroff wrote: > I know this may sound like a pretty lame answer, but you'll get a lot of > benefit from reading the definition of a SIP PROXY from RFC3261. > You can't do much with OpenSIPS (properly) if you dont' know the underlying > RFC. This is very different from other SIP software packages, like Asterisk > where you pretty much can't break RFC compliance on purpose (hah, it may > just already be broken).. I will suppose that this answer is not for me, just because thats what I told to the other user ... "without a deep knowleage of SIP, you will get into lot of troubles trying to do something with a SIP PROXY" -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Is opensips a front end to asterisk?
Sorry I must answer on this way .. but as the list don't set the reply-to correctly and this usser's mail server refuses direct mails .. -- Raúl Alexis Betancor Santana Dimensión Virtual --- Begin Message --- On Tuesday 07 July 2009 20:15:01 li...@grounded.net wrote: > On Tue, 7 Jul 2009 14:02:11 -0400, Alex Balashov wrote: > > Specific and well-parameterised questions really are the key. > > I'll certainly do that, once I start understanding the product but for now, > I'm just trying to get a handle on basics, not deep in depth > understandings, just enough to formulate a plan. You should go into deep knowleadge, it's a MUST to work with a sip proxy. You could begin reading the "Getting starting gide" and so. > One was asking about the viability of using opensips on ESXi. Because of > how easy it is to use snapshots, backup and so on, this would be the best > working environment. So my question was, does opensips have any hardware > timing requirement issues such as asterisk does. If timing is not critical, > as a voip server is, then opensips must run nicely in a virtual manner. Yes, you could use it into a VM, no timming issues like Asterisk. Talking about backup and so ... you only need to do a backup of the .cfg file and the database backend you use, so you will not get any real advantage of running inside a VM from backup point of view. > I don't have any numbers to work with, which is why I say scalable. I'm > looking for something which can help me to scale a voip based application > to many users. So let's say hundreds of users so that we have a number. I > know many of you are running many thousands so this should be a good > starting point. It depends on lot of variables, like available mem, CPU power, network (the most important part), but also how complex is your .cfg about request proccessing, how do you handle database request, etc. So there is not a magic formula, but there are sip-proxies around the world working with Million of users. > This is how I would have approached this, until I started looking for a sip > gateway/load balancer. That's a setup, not direcly related with the software you use. > This should be pretty straight forward to those who have pro setups and > want as much reliability as possible. I want to have two separate locations > so that I can fail over, simple as that really. There are not "simple" scenarios in SIP world and faiolver is very-complex one. > >-From what I can tell, opensips could act as a pbx on it's own but it can > > act as a proxy/load >balancer/gateway to asterisk systems as well. > > This is what I asked about in this thread a couple of times now. It's not > fully clear to me, even after reading. It sometimes sounds like opensips > can be a voip server though it does not provide other media services such > as voice mail and so on. I get that it is a gateway but I'm trying to get a > better understanding of FROM that point on. That's because Opensips it's a proxy, not a PBX, not a B2BUA, etc., it doesn't manage media, so you need some "complements" to have a "full-featured VoIP system" -- Raúl Alexis Betancor Santana Dimensión Virtual --- End Message --- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Voicemail system
On Wednesday 08 July 2009 12:59:40 Paul Mancheno H. wrote: > Hi friends. > > I want to implement a voicemail system for the telecommunications company I > work, I tried but it seems that Asterisk supports only 150 concurrent > calls. > > Could it be better to use Asterisk and OpenSIPS to improve this system?, > Can I use SEMS? For building a voipmail system, better to use SEMS than Asterisk, less resources needed, more concurrent calls. Anyhow, all depends on your budget and how do you need it to scale. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] codec manipulation feature
On Tuesday 16 June 2009 12:48:20 Jeff Pyle wrote: > Exactly. This would be so incredibly useful. > > Imagine a customer with a SIP-based PRI and a T1¹s worth of bandwidth > behind it. This is a common scenario for my customers. We try to avoid > running all G.729 wherever possible because of the obvious and ugly audio > quality hit. At G.711 I can only let them use 17 channels on their 23 > channel PRI. That¹s fine for many customers but a fraction think they need > or actually need all 23 channels. Some can afford second T1 to allow them > access to the remaining 6 channels. How about dynamic > codec selection by the proxy. > > We¹ve already worked out the math and routing logic in Opensips, provided > we have the following items in the customer¹s profile: > * Total bandwidth available to customer (pick a circuit type, or input raw > kbps) > * Minimum percentage bandwidth reserved for data > * Total number of calls required > > Given this information, Opensips can compute how many calls we can leave at > G.711 and after what point we have to start compressing them into G.729 (by > verifying G.729 in the SDP, and removing G.711). Thank you dialog module. > For example, on a 1.5 meg T1 with 15% reserved for data, the formula says I > can allow 12 channels at G.711, but have to compress the remaining 11. > With a 5% data margin I can get away with 15 uncompressed and 8 compressed. > Everything¹s just about ready except for the ability to work on the SDP to > make it happen. > > > - Jeff Just as a comment ... if you have very bad quality with G.729, maybe your problem is on other place. Having T1 line's for data, just for putting VoIP traffic ast SIP+G.711 is a waste of bandwidth, but that just my opinion. Our customers uses ADSL lines ranging from 3Mb/320Kb to 10Mb/1Mb, and using the same line for Inet navitation + VoIP, over a 3M/320Kb line you could drive 5 G729 calls without any problem, and over a 10Mb/1Mb line, more than 12 ... so I don't undestand where you have problems with a 1.5Mb/1.5Mb line. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Patch: add support for telling to call-control witch config file to use
If anyone is interested on this .. this little, little patch.
--
Raúl Alexis Betancor Santana
Dimensión Virtual
diff -rN old-callcontrol/call-control new-callcontrol/call-control
55a56,58
> parser.add_option("--config", dest="configuration_filename",
> default=callcontrol.configuration_filename,
> help="config file", metavar="File")
58a62
> callcontrol.configuration_filename = options.configuration_filename
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Re: [OpenSIPS-Users] What is this error : msg_send: ERROR: udp_sendfailed?
On Friday 12 June 2009 08:46:49 Tung Tran wrote: > Dear, > > I am running Openser as root, > I cannot find out what kind of permission is it, and how to solve it. > > BTW, this server has 6 NIC in total and only one of them are working > (connected), Do you think that should be problem if server has many NIC > card? > > Anyone has experience about this problem pls jumbing, I'm sure at 99,9% that the error is caused by some firewall rules you have on that machine, or a network missconfiguration. -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Inject REFER
On Thursday 11 June 2009 22:57:14 Brett Nemeroff wrote: > 2009/6/11 Raúl Alexis Betancor Santana > > > On Thursday 11 June 2009 22:27:23 Brett Nemeroff wrote: > > > Does anyone know if there is an actual QSIG implementation of "REFER"? > > > > That > > > > > may make this whole issue moot. > > > > Could you describe what scenario are you trying to implement ? > > A call comes in to opensips, which sends the call to a TDM gateway. This > terminates to a ACD. And agent utilizes some mechanism to deflect the call. > It's necessary that the deflection occur on the far end of the sip > connection (so there is a sip to tdm translation in there, of which I don't > know if it can support a deflection). Possible options for the deflection > is either something that is supported natively in a qsig to sip mapping, or > an out of band solution. I was expecting this to be out of band such that a > script would poke at the fifo to perform such deflection. > > I know.. it's a weird setup. So, you have ... SIP <-> TDM <-> SIP, yes is a weird and stupid setup ... You have 3 possible solutions: 1) SIP <-> B2BUA <-> TDM <-> SIP, Call-Deflection by DTMF controlled at B2BUA side 2) SIP <-> TDM <-> SIP ^ ^ |- SIP_faked_UAC -| So, for example, you could use some sipp script that "insert" a faked REFER from the "origin", but controlled by the far end SIP agent. 3) SIP <-> TDM <-> SIP, trying to use call-deflection features that the TDM gateway support (AFAIK, only usable with SS7 links or QSIG links but not with PRI, ANALOG or BRI links) -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Inject REFER
On Thursday 11 June 2009 22:27:23 Brett Nemeroff wrote: > Does anyone know if there is an actual QSIG implementation of "REFER"? That > may make this whole issue moot. Could you describe what scenario are you trying to implement ? -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Inject REFER
On Thursday 11 June 2009 18:05:39 Brett Nemeroff wrote: > All,Is it possible to inject a REFER message into a call? I'll be proxying > calls to a TDM gateway.. I know this seems like an odd request, but I'd > like to essentially hijack the call with a REFER. > > I believe the UAC module can handle the request, but do I need to cleanup > the dialog somehow? > > Thanks, > Brett You could not do that without a B2BUA, because you will break CSeq on the stablished dialogs -- Raúl Alexis Betancor Santana Dimensión Virtual ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
