sr-users July 2005

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[Serusers] 2 SER's
by Thomas Britis 25 Jul '05

25 Jul '05

2 1

[Serusers] Questions about ser
by Michael Ulitskiy 25 Jul '05

25 Jul '05

Hello, I'm using several modules that access a database. I've noticed that when I start ser in forked mode with "children=8", it generates 62 connections to database. Does every child open a separate set of database handles (one for every module)? I remember I saw on the mailing list advises to run ser with hundreds of children to avoid problems when no workers are available to process a message. If my guess is true it would generate hundreds (or even thousands) of database connections. I'm really interested to hear how people out there deal with it. On a separate note I'm curious whether number of children in ser is limited to a given startup value or ser is able to fork additional children as needed? Thank you, Michael

3 6

Re: [Serusers] SER + NAT
by Thomas Britis 25 Jul '05

25 Jul '05

1 0

[Users] !!! old 'break' removed
by Daniel-Constantin Mierla 25 Jul '05

25 Jul '05

Hello, from now on, the 'break' cannot be used to return from a route. The only place to use is to end a 'case' block in a 'switch' statement. As it was agreed on the mailing list, 'return' must be used to stop the execution of a route -- these changes brings an appropriate c/shell-like meaning for these commands. A message telling about this new functionality is displayed every time when an unknown command is met. The dokuwiki will be updated to reflect this major change. Daniel

1 0

[Serusers] Bug report in pa module
by Xavier TRENTIN 25 Jul '05

25 Jul '05

Hi all, we're trying to connect Windows Messenger Clients to ser-0.10.99 We've configured ser with pa module. If I understand how it works : -U1 is connecting for the first time to ser. -U1 add his contacts, and each time he adds a contact, ser inserts the data in watcherinfo table. For exemple, he adds U2 inhis contacts. -U2 is connecting for the first time to ser. -presentity_contact table is updated and U1 can see that U2 is online. -U2 add his contacts, and each time he adds a contact, ser inserts the data in watcherinfo table. For exemple, he adds U1 inhis contacts. -Then presentity_contact table is updated and U2 can see that U1 is online. We've tested the two following configurations: -In one hand, if the Usernames of 2 clients are very differents, such as "Mohamed" and "Xavier", it works fine. -In the other hand, if the Usernames are something like "Xavier1" and "Xavier2" and each of them are in the contacts list of the other, then we found a problem while ser was trying to insert data in presentity_contact table. The s_id generated is the same for each user. Therefore, only the first user who was connected the first time will see the status of the other user. B.R. Xavier.

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RE: [Serusers] Does VOVIDA B2BUA works with SER+MEDIAPROXY?
by Miguel Angel Villar 25 Jul '05

25 Jul '05

Alberto, try this one: http://www.yousendit.com Regards Miguel.- > -----Mensaje original----- > De: Alberto Cruz [SMTP:acruz@tekbrain.com] > Enviado el: Lunes, 25 de Julio de 2005 01:06 a.m. > Para: Mike Tkachuk > CC: serusers(a)lists.iptel.org > Asunto: Re: [Serusers] Does VOVIDA B2BUA works with SER+MEDIAPROXY? > Importancia: Alta > > Hi Mike I already have the logfile but it is 187KB Do you know from an > another place to post it that supports this size of file? > > Regards > > Alberto Cruz > > Mike Tkachuk wrote: > > >Hello Alberto, > > > >Try to post also a logfile of b2bua ( -d -v keys). > > > >Saturday, July 23, 2005 2:22:36 AM, you wrote: > > > >AC> Thanks Mike, I really appreciate your feedback but I would > >AC> like to resolve the routing problem that I have at this time > >AC> before to skip to another solution. > >AC> My deployment is like this: > > > >AC> NatedUA --> SER+MEDIAPROXY <----> B2BUA > >AC> | > >AC> | > >AC> PSTN GW > > > >AC> The B2BUA is running in a separeted machine. > >AC> If the UA place a call and follow the progress of the call, > >AC> the call is completed without any trouble > >AC> If the UA place a call but then we decide to CANCEL the call > >AC> for any reason, the CALL continues in progress from SER to the > >AC> PSTN GW > > > >AC> Please I need help I really don't know what I'm doing wrong. > >AC> I guess I'm making a mistake in my routing configuration at > >AC> ser.cfg but I can't find where I already tried many things like > >AC> forwarding only INVITE and BYE from SER to B2BUA but It didn't > >AC> work neither. > > > >AC> I already posted my NGREP trace and my ser.cfg file at : > >AC> http://pastebin.ca/18313 > >AC> http://pastebin.ca/18314 > > > >AC> Regards > > > >AC> Alberto Cruz > >AC> mike(a)yes.net.ua wrote: > > > >AC> Hi! > > > >AC> I beleive that *B2BUA is compatible with vovida in radius attributes > part > >AC> and should work fine with your billing. > > > >AC> P.S. Anyway it *b2bua script is very simple and you can easy > customise it > >AC> for your needs. > > > > > >AC> Thanks for you recommendation. > > > >AC> The issue that I have is that we are using ALEPO as > >AC> Billing/Prepaid/Postpaid Platform. and they recommend us VOVIDA B2BUA > >AC> because the radius attributes format are compatibles betwen them. > > > >AC> It is because I need to use VOVIDA B2BUA. > > > >AC> If I change to other B2BUA like ASTERISK for example, we should to > write > >AC> an additional interface in order the ALEPO could recive the radius > >AC> attribute as it expect. Maybe in the future we could do this but at > this > >AC> time we have the time over us because we are already in production > and > >AC> we need to solve the CANCEL issue. > > > >AC> Does anybody alse has a ser.cfg example for SER+MEDIAPROXY and VOVIDA > >AC> B2BUA? > > > >AC> Regards > > > >AC> Miguel Angel Villar wrote: > > > > > >AC> Alberto. > > > >AC> I tried VOVIDA B2BUA with SER+MEDIAPROXY some time ago and > worked > >AC> fine. I stoped using it because I noticed the VOVIDA B2BUA had a poor > >AC> interface, any custom should be done in the code, so I had to > recompile > >AC> after any change. Due to this I changed it for the Asterisk B2BUA, > really > >AC> nice interface and works great with SER+MEDIAPROXY, you should try > it. > > > >AC> *B2BUA: http://developer.berlios.de/projects/b2bua/*B2BUA > >AC> mailing list: > >AC> http://lists.berlios.de/mailman/listinfo/b2bua-users*B2BUA > >AC> (installation > >AC> > manual):http://lists.berlios.de/pipermail/b2bua-users/2005-April/000047.ht > mlEnjoY. > >AC> MaV.- > > > > > >AC> -----Mensaje original----- > >AC> De: Alberto Cruz [SMTP:acruz@tekbrain.com] > >AC> Enviado el: Jueves, 21 de Julio de 2005 12:35 p.m. > >AC> Para: 'serusers(a)lists.iptel.org' > >AC> Asunto: [Serusers] Does VOVIDA B2BUA works with SER+MEDIAPROXY? > >AC> Importancia: Alta > > > >AC> Hi does anybody is running SER+MEDIAPROXY and VOVIDA B2BUA? > > > >AC> Please I need help to handle the CANCEL message. > > > >AC> I've already try searching > >AC> thehttp://www.archivum.info/serusers@lists.iptel.org/ and > >AC> thehttp://lists.iptel.org/pipermail/serusers/ for any tip or > >AC> configuration > >AC> example but I didn't find anything. > > > >AC> Regards > > > >AC> Alberto Cruz > > > >AC> _______________________________________________ > >AC> Serusers mailing > >AC> listserusers@lists.iptel.orghttp://lists.iptel.org/mailman/listinfo/serusers > > > > > >AC> _______________________________________________ > >AC> Serusers mailing > >AC> listserusers@lists.iptel.orghttp://lists.iptel.org/mailman/listinfo/serusers > > > > > >AC> _______________________________________________ > >AC> Serusers mailing > >AC> listserusers@lists.iptel.orghttp://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > >-- > >Mike Tkachuk > > > > > > > > > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] accounting
by harry gaillac 25 Jul '05

25 Jul '05

I just use asterisk as Pstn gateway. I would prefer than community improve the ser serweb/sems features . Harry --- Iqbal <iqbal(a)gigo.co.uk> a écrit : > Cant comment, use asterisk, which allows me to add > any email address and > bind to any voicemail ID > > Iqbal > > harry gaillac wrote: > > >Hello > > > >I agree you but that don't solve problem. > >if I dial sip:iqbal@sip.domain.com voicemail module > >won't find your email. > > > >Harry > > > >--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > > > > > > > >>You should look at swicthing them, I dont work > with > >>sems, but normally > >>when a user wishes to pick up voicemail, they will > >>have to dial a > >>numeric number, instead of a alias like > >>iqbal(a)sip.domain.com > >> > >>Hence many many eons ago, I switched my logic to > >>storing numeric numbers > >>in subscribers and names in aliases, fits better > >>with the general telco > >>scenario > >> > >>iqbal > >> > >>harry gaillac wrote: > >> > >> > >> > >>>Hello, > >>> > >>>I try to make ser/sems/serweb working together. > >>> > >>>I add account like this: > >>>"user@domain" and aliase "123@domain". > >>> > >>>I ever set "Lookup("aliases");" in ser.cfg. > >>> > >>>It's ok for direct calls. > >>> > >>>However if callee set "forward to voicemail" via > >>>Serweb or if the callee is unavaible or busy > >>>voicemail module just query subscribers table. > >>> > >>>Sems return: > >>> > >>> > >>> > >>> > >>>>Sems[2535]: Error: > >>>>(AmSession.cpp)(startSession)(497): 404 > voicemail: > >>>>no > >>>>email address for user <aliase> > >>>> > >>>> > >>>> > >>>> > >>>I would like sems find email of the alias ! > >>> > >>>Harry > >>> > >>> > >>> > >>>--- Andrei Pelinescu-Onciul <andrei(a)iptel.org> a > >>> > >>> > >>écrit > >> > >> > >>>: > >>> > >>> > >>> > >>> > >>> > >>>>On Jul 24, 2005 at 15:15, harry gaillac > >>>><gaillacharry(a)yahoo.fr> wrote: > >>>> > >>>> > >>>> > >>>> > >>>>>Hello, > >>>>> > >>>>>I use ser/serweb-0.9.3 I added tow accounts > with > >>>>>numerical aliases. > >>>>>When I dial user1@domain to user2@domain it's > ok. > >>>>>if i wish to forward to voicemail sems send me > >>>>> > >>>>> > >>>>> > >>>>> > >>>>back > >>>> > >>>> > >>>> > >>>> > >>>>>voicemessages. > >>>>>However if i dial alias1@domain to > alias2@domain > >>>>> > >>>>> > >>I > >> > >> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>get > >>>> > >>>> > >>>> > >>>> > >>>>>the message below. > >>>>> > >>>>>How can I solve this problem ? > >>>>> > >>>>> > >>>>> > >>>>> > >>>>Why don't you use lookup("aliases") before > >>>>forwarding to voicemail? > >>>> > >>>>Andrei > >>>> > >>>> > >>>> > >>>> > >>>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>___________________________________________________________________________ > >> > >> > >> > >>>Appel audio GRATUIT partout dans le monde avec le > >>> > >>> > >>nouveau Yahoo! Messenger > >> > >> > >>>Téléchargez cette version sur > >>> > >>> > >>http://fr.messenger.yahoo.com > >> > >> > >>>_______________________________________________ > >>>Serusers mailing list > >>>serusers(a)lists.iptel.org > >>>http://lists.iptel.org/mailman/listinfo/serusers > >>> > >>>. > >>> > >>> > >>> > >>> > >>> > > > > > > > > > > > > > > > >___________________________________________________________________________ > > >Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > >Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > >. > > > > > > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] assign numbers
by Kissandrakis S. George 25 Jul '05

25 Jul '05

Hello, I have installed SER and a pstn gateway. SIP clients can make calls to pstn phones I want to give to authenticated clients a number so a pstn user will be able to call sip users is there a ser.cfg configuration example how to do that? (Assume that the gatekeeper allready routes the calls 2831877... to sip server) How can i force that only authenticated users can call to pstn (now unauthnticated users can call too) Thank you George

2 2

[Users] testing
by root 25 Jul '05

25 Jul '05

hi, can you please suggest me a method with which we can actually test the processing of calls through Openser without actually putting it Live. We just wanted to check the way it performs with loads of calls because scalablity is one of the major issue for us. Please help me. Thanx a lot Jayesh

2 1

[Serusers] RE: failure route
by harry gaillac 25 Jul '05

25 Jul '05

Thanks for help, I set in route [3] : if (t_check_status("(486])||(408)")) { t_on_failure("1"); route(1); break; }; When timer expire and caller receive 408 or 486 last replies it should go to failure route !? Regards Harry according to t_check_status TM docs : *in routing block - the code of the last sent reply. *in on_reply block - the code of the current received reply. *in on_failure block - the code of the selected negative final reply. --- Zeus Ng <zeus.ng(a)isquare.com.au> a écrit : > Harry, > > I haven't had much time to look at you config file. > Otherwise, I'll will > answering questions on the list. So busy these days! > > In regards to your question, you have not forward / > relay the request in > your failure route. That is, you revert the uri, > rewrite host port, but no > t_relay. So, the request will disappear once it > reach the end of that block. > Hope that help. > > Zeus Ng, CISSP, CCSA > Principal Consultant > iSquare Technology > Tel: +61 2 9419 3887 > Fax: +61 2 9410 2629 > Mob: 0416 135 794 > sip: zeusng(938764)(a)sip.isquare.com.au > Email: zeus.ng(a)isquare.com.au > > > > > -----Original Message----- > > From: harry gaillac [mailto:gaillacharry@yahoo.fr] > > > Sent: Saturday, 16 July 2005 7:17 AM > > To: zeus.ng(a)isquare.com.au > > Subject: failure route > > > > > > Hello, > > > > i read > > > http://lists.iptel.org/pipermail/serusers/2004-September/011828.html > > however i can't use failure route for busy or > noanswer > > > > How can I forward caller to voicemail when status > 486 > > or 408 are replied to caller ? > > > > Regards > > Harry > > > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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sr-users July 2005