Re: [SR-Users] Force RTPProxy
No help at all? It would be a good start point a doc to read concern this aspect. Of course, an example based on the conf could me help a lot better. Thank's to all! Simon Il 06/05/2011 10:40, Simone Felici ha scritto: Hello to all! I need a little help with our ser installation (ser-2.0.0-rc1). Due some network restrictions, we need to force every call to pass trought rtpproxy. Could someone point me to the right place where to force it? Please an example based on our config would be really helpful, but in two scenarios: 1. ALL CALLS trought rtpproxy 2. ONLY CALLS from a given network trought rtpproxy We are in doubt on what choose between the two solutions. Having both examples could me help to implement the needs faster :) Thank's a lot for your help! Simon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Force RTPProxy
Hi, did you look at the manuals? You must simply issue a rtpproxy_offer/_answer to engage the RTPProxy from your route/reply_route, as described in the manual: http://www.kamailio.org/docs/modules/devel/modules/rtpproxy.html#id3056341 Carsten 2011/5/9 Simone Felici : > > No help at all? > It would be a good start point a doc to read concern this aspect. > Of course, an example based on the conf could me help a lot better. > > Thank's to all! > > Simon > > Il 06/05/2011 10:40, Simone Felici ha scritto: >> >> Hello to all! >> >> I need a little help with our ser installation (ser-2.0.0-rc1). >> Due some network restrictions, we need to force every call to pass trought >> rtpproxy. >> Could someone point me to the right place where to force it? >> Please an example based on our config would be really helpful, but in two >> scenarios: >> 1. ALL CALLS trought rtpproxy >> 2. ONLY CALLS from a given network trought rtpproxy >> We are in doubt on what choose between the two solutions. Having both >> examples could me help to >> implement the needs faster :) >> >> Thank's a lot for your help! >> >> Simon >> >> > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Carsten Bock http://www.ng-voice.com mailto:cars...@ng-voice.com Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Force RTPProxy
Hello, I'll start from here and set-up a scenario to test it out. Is it compatible with ser-2.0.0-rc1 used by me? Thank's for the answer. Simon Il 09/05/2011 10:28, Carsten Bock ha scritto: Hi, did you look at the manuals? You must simply issue a rtpproxy_offer/_answer to engage the RTPProxy from your route/reply_route, as described in the manual: http://www.kamailio.org/docs/modules/devel/modules/rtpproxy.html#id3056341 Carsten 2011/5/9 Simone Felici: No help at all? It would be a good start point a doc to read concern this aspect. Of course, an example based on the conf could me help a lot better. Thank's to all! Simon Il 06/05/2011 10:40, Simone Felici ha scritto: Hello to all! I need a little help with our ser installation (ser-2.0.0-rc1). Due some network restrictions, we need to force every call to pass trought rtpproxy. Could someone point me to the right place where to force it? Please an example based on our config would be really helpful, but in two scenarios: 1. ALL CALLS trought rtpproxy 2. ONLY CALLS from a given network trought rtpproxy We are in doubt on what choose between the two solutions. Having both examples could me help to implement the needs faster :) Thank's a lot for your help! Simon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Simone Felici Divisione Tecnica: Progettazione e Sviluppo tel. +39 0461.030.111 fax. +39 0461 030.112 Via Fersina, 23 - 38123 Trento - MC-link S.p.A. Sede Direzionale e Amministrativa Via Carlo Perrier, 9/a - 00157 Roma Sede Legale Via Fersina, 23 - 38123 Trento http://www.mclink.it Save a tree. Don't print this e-mail unless it's really necessary Informativa ai sensi del Codice della proprietà industriale e del Codice dei dati personali. Le informazioni contenute in questa e-mail e negli eventuali allegati, possono contenere informazioni confidenziali e coperte da segreto commerciale/industriale. Esse vengono comunicate nei limiti giuridici dei rapporti in essere fra le parti e pertanto nessun ulteriore diritto di proprietà intellettuale o industriale può essere rivendicato dal ricevente. Le informazioni contenute in questa e-mail e negli eventuali allegati sono indirizzate esclusivamente a coloro che figurano come destinatari. Se avete ricevuto per errore questa e-mail siete pregati di informarci (rispedendola al mittente) e di provvedere alla sua rimozione, a non farne utilizzo e a non conservarne alcuna copia. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Force RTPProxy
Hi Simone, it should be similar for SER 2.0. The commands may be different though, so you probably do one of the following: - upgrade to a more recent SER version - check the readme for your version (should be in modules/nathelper directory) Carsten 2011/5/9 Simone Felici : > > Hello, > > I'll start from here and set-up a scenario to test it out. > Is it compatible with ser-2.0.0-rc1 used by me? > > Thank's for the answer. > > Simon > > > Il 09/05/2011 10:28, Carsten Bock ha scritto: >> >> Hi, >> >> did you look at the manuals? >> You must simply issue a rtpproxy_offer/_answer to engage the RTPProxy >> from your route/reply_route, as described in the manual: >> http://www.kamailio.org/docs/modules/devel/modules/rtpproxy.html#id3056341 >> >> Carsten >> >> 2011/5/9 Simone Felici: >>> >>> No help at all? >>> It would be a good start point a doc to read concern this aspect. >>> Of course, an example based on the conf could me help a lot better. >>> >>> Thank's to all! >>> >>> Simon >>> >>> Il 06/05/2011 10:40, Simone Felici ha scritto: Hello to all! I need a little help with our ser installation (ser-2.0.0-rc1). Due some network restrictions, we need to force every call to pass trought rtpproxy. Could someone point me to the right place where to force it? Please an example based on our config would be really helpful, but in two scenarios: 1. ALL CALLS trought rtpproxy 2. ONLY CALLS from a given network trought rtpproxy We are in doubt on what choose between the two solutions. Having both examples could me help to implement the needs faster :) Thank's a lot for your help! Simon >>> >>> ___ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >> >> >> > > > -- > Simone Felici > Divisione Tecnica: Progettazione e Sviluppo > > tel. +39 0461.030.111 > fax. +39 0461 030.112 > Via Fersina, 23 - 38123 Trento > > - > MC-link S.p.A. > Sede Direzionale e Amministrativa > Via Carlo Perrier, 9/a - 00157 Roma > Sede Legale > Via Fersina, 23 - 38123 Trento > > http://www.mclink.it > > Save a tree. Don't print this e-mail unless it's really necessary > > Informativa ai sensi del Codice della proprietà industriale e del Codice dei > dati personali. > Le informazioni contenute in questa e-mail e negli eventuali allegati, > possono contenere informazioni confidenziali e coperte da segreto > commerciale/industriale. Esse vengono comunicate nei limiti giuridici dei > rapporti in essere fra le parti e pertanto nessun ulteriore diritto di > proprietà intellettuale o industriale può essere rivendicato dal ricevente. > Le informazioni contenute in questa e-mail e negli eventuali allegati sono > indirizzate esclusivamente a coloro che figurano come destinatari. > Se avete ricevuto per errore questa e-mail siete pregati di informarci > (rispedendola al mittente) e di provvedere alla sua rimozione, a non farne > utilizzo e a non conservarne alcuna copia. > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Carsten Bock http://www.ng-voice.com mailto:cars...@ng-voice.com Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Force RTPProxy
Hi Carsten, The upgrade is something I would like to do very! But at the moment there are so many customizations on the structure. Could be maintained, but the work sould be almost long. I search to understand if at least this command is implemented and then look forward. Meanwhile many thank's Simon Il 09/05/2011 11:48, Carsten Bock ha scritto: Hi Simone, it should be similar for SER 2.0. The commands may be different though, so you probably do one of the following: - upgrade to a more recent SER version - check the readme for your version (should be in modules/nathelper directory) Carsten ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio BLF - NOT SIMPLE
Hi Lucas! Am 06.05.2011 22:15, schrieb Lucas Alvarez: > Hi, I've been trying to configure kamailio for enabling BLF for Aastra > phones. Sincerely, it's been a nightmare, till now I'm not sure if I > have to enable pua_dialoginfo or presence_dialoginfo module or both for > handling dialog events. Besides that is not clear which changes should > be made to the route configuration nor which parameters should be > configured. I would appreciate if anyone can enlighten me or could point > me to the right documentation page. For the Kamailio presence server to understand the "dialog" event package you need to use presence_dialoginfo module. (this is necessary to handle the SUBSCRIBE requests). The second thing is: where is the dialing information coming from? Some phone will send PUBLISH requests with the dialog state on dialog changes. Then it should already work. If your phones do not PUBLISH the dialog state to the proxy, then there is a workaround possible in Kamailio. If Kamailio uses the dialog module, then the dialog changes (calls) can be derived from dialog state and published to the presence server (either an external presence server or the same Kamailio proxy instance) by using the pua_dialoginfo module. regards klaus ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Authenticate if receiving 401
Am 07.05.2011 01:03, schrieb Eric Hiller: > The only downside is now to add a new gateway involved editing the > config file and reloading kamailio. At some point I could put this in > SQL and just update the gateways daily ie. DASH. you could store IP adresses in database tables of LCR module and use the from_gw() functions, eg: http://www.kamailio.org/docs/modules/3.1.x/modules/lcr#id2554739 klaus ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio fails to register to a VoIP provider
Am 08.05.2011 23:51, schrieb Pierpaolo Culurciello: > I > tried to analize the two flows of SIP messages but I didn't notice any > difference between them so I can't understand why Kamailio is not able > to register. Take a look at the CSeq Header. >From readme: Known limitations in this version: * authentication does not support qop auth-int, just qop auth; * CSeq not increased during authentication - the response may be rejected. Thus, if the proxy is strict, Kamailio can not authenticate to the other proxy as Kamailio is not standard conform. @developers: in case of REGISTER I think it should be possible for Kamailio to increase CSeq? regards Klaus ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Ser AVP
Hello all,
Have this issue with my ser config. I try to load a fwdbusy value from db
using the $fu class, but it's complaining..
config:
...
if (avp_db_load("$fu", "s:fwdbusy")) {
...
log:
0(20749) ERROR: [pvapi.c:516]: error searching pvar "fu"
0(20749) ERROR: [pvapi.c:720]: wrong char [u/117] in [$fu] at [2 (0)]
0(20749) ERROR: avpops [avpops.c:275]: bad param 1; expected :
$pseudo-variable or int/str value
What's wrong with that? Should it not considered as a ser avp? Not K one. Or
that avp_db_load is an exlusively K function?
Thank you,
Claudio
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Re: [SR-Users] Ser AVP
Hello,
the avpops module is now the one coming via kamailio development.
First, you have to load modules_k/pv.so module (to get $fu), then the
second parameter has to be PV format as well, e.g.,: $avp(fwdbusy)
(note: s: is optional now in the name of the avps, if it is missing then
the name is implicit considered string).
Cheers,
Daniel
On 5/9/11 3:24 PM, Claudio Furrer wrote:
Hello all,
Have this issue with my ser config. I try to load a fwdbusy value from db
using the $fu class, but it's complaining..
config:
...
if (avp_db_load("$fu", "s:fwdbusy")) {
...
log:
0(20749) ERROR: [pvapi.c:516]: error searching pvar "fu"
0(20749) ERROR: [pvapi.c:720]: wrong char [u/117] in [$fu] at [2 (0)]
0(20749) ERROR: avpops [avpops.c:275]: bad param 1; expected :
$pseudo-variable or int/str value
What's wrong with that? Should it not considered as a ser avp? Not K one. Or
that avp_db_load is an exlusively K function?
Thank you,
Claudio
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--
Daniel-Constantin Mierla
http://www.asipto.com
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Re: [SR-Users] SIP Trunk Usage Monitoring
>> I recently interconnected to an upstream carrier using Kamailio 3.0, >> working fine. We have configure 2 SIP trunks for failover/redundnacy. >> I'm using dispatcher module to round robin calls to the carrier. I >> wanted to monitor trunk usage between us. I was reading in the devel >> 3.2 about some native monitoring in the new dispatcher module without >> the need to load the dialog module. >> >> Can I use the new module in version 3.0 or will I need to upgrade the >> core to 3.2? >> >> I typically will not use dev software in production, so can anyone >> suggest a light method of trunk usage monitoring for ver 3.0? I'm not >> opposed to using the dialog module, I'm just not familiar with it and >> not sure the best way to integrate it for usage tracking. I don't >> need to do billing or capture cdr's on the SIP-Routers so the simpler >> the better. > > Honestly, going the dialog module route is your easiest bet. > > -- > Alex Balashov - Principal Thanks Alex, Would there be any usage examples of dialog module? I'm not sure if I really need profiles and [values] and when to set and unset. A push in the right direction would be helpful. Thanks. JR -- JR Richardson Engineering for the Masses ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Ser AVP
Great, thank you.
BTW, is going to be deprecated the module_s/avp_db module (functions
like load_attrs, load_extra_attrs, etc)?
>From the avp_db/README file:
..."You may also want to check the avpops module which is more flexible and
contains more functions. In future SER releases the avp module will be probably
deprecated in favor of avpops module..."
Regards,
Claudio
On Mon, 9 May 2011, Daniel-Constantin Mierla wrote:
> Hello,
>
> the avpops module is now the one coming via kamailio development.
>
> First, you have to load modules_k/pv.so module (to get $fu), then the second
> parameter has to be PV format as well, e.g.,: $avp(fwdbusy) (note: s: is
> optional now in the name of the avps, if it is missing then the name is
> implicit considered string).
>
> Cheers,
> Daniel
>
> On 5/9/11 3:24 PM, Claudio Furrer wrote:
> > Hello all,
> >
> > Have this issue with my ser config. I try to load a fwdbusy value from db
> > using the $fu class, but it's complaining..
> >
> > config:
> > ...
> > if (avp_db_load("$fu", "s:fwdbusy")) {
> > ...
> >
> > log:
> > 0(20749) ERROR: [pvapi.c:516]: error searching pvar "fu"
> > 0(20749) ERROR: [pvapi.c:720]: wrong char [u/117] in [$fu] at [2
> > (0)]
> > 0(20749) ERROR: avpops [avpops.c:275]: bad param 1; expected :
> > $pseudo-variable or int/str value
> >
> > What's wrong with that? Should it not considered as a ser avp? Not K one. Or
> > that avp_db_load is an exlusively K function?
> >
> > Thank you,
> > Claudio
> >
> > ___
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> > sr-users@lists.sip-router.org
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierla
> http://www.asipto.com
>
>
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[SR-Users] Dialog Module Usage Question
Hi All, I'm trying to configure the dialog module with profiles for outbound calls to different Dispatcher domains. I can get the dialog module configured for inbound calls and pull statistics and separate calls into specific profiles, that part is working well. The problem I am having specifically is for outbound calls flowing through a dispatcher, I want to separate calls into a profile for each SIP domain that I send calls to. I'm not sure how do accomplish this. I see in kamctl fifo dlg_list, there is 'caller_bind_addr:: udp:10.10.12.24:5060'. Is there a way to export that into a variable that I can use as the 'profile' for the dialog? Thanks. JR -- JR Richardson Engineering for the Masses ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Dialog Module Usage Question
> I'm trying to configure the dialog module with profiles for outbound > calls to different Dispatcher domains. I can get the dialog module > configured for inbound calls and pull statistics and separate calls > into specific profiles, that part is working well. > > The problem I am having specifically is for outbound calls flowing > through a dispatcher, I want to separate calls into a profile for each > SIP domain that I send calls to. I'm not sure how do accomplish this. > I see in kamctl fifo dlg_list, there is 'caller_bind_addr:: > udp:10.10.12.24:5060'. Is there a way to export that into a variable > that I can use as the 'profile' for the dialog? > Actually, I can't use the 'caller_bind_addr::udp:10.10.12.24:5060' it should be the 'callee_contact:: sip:5551212@10.10.14.101' without the user part of the URI, just the domain. Any suggestions? Thanks. JR -- JR Richardson Engineering for the Masses ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Dialog Module Usage Question
On 05/09/2011 07:16 PM, JR Richardson wrote: The problem I am having specifically is for outbound calls flowing through a dispatcher, I want to separate calls into a profile for each SIP domain that I send calls to. I'm not sure how do accomplish this. I see in kamctl fifo dlg_list, there is 'caller_bind_addr:: udp:10.10.12.24:5060'. Is there a way to export that into a variable that I can use as the 'profile' for the dialog? The domain part of the request URI is stored in a pseudovariable called $rd. I would just use that as your profile 'value'. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
