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37558 discussions

[Users] Re: Got SEMS-svn and OpenSER-cvs to work
by Rick van Rein 28 Apr '06

28 Apr '06

Hi, Just for the (pipermail) record: How I got OpenSER+SEMS to work on FreeBSD. OpenSER is not causing any problems in combination with SEMS-svn anymore. No need to use any specific version, just the CVS-head from openser.org. It now works completely, an IVR app is reading out .wav files and processing the DTMF music that I play for it. In openser.cfg I used ... unix_sock="/tmp/openser.sock" ... if(!t_write_unix("/tmp/sems.sock", "ivr")){ t_reply("500", "error contacting sems"); } ... In sems.conf I used ... socket_name=/tmp/sems.sock reply_socket_name=/tmp/sems.reply.sock ser_socket_name=/tmp/openser.sock send_method=socket ... I started OpenSER with openserctl start I started SEMS with /path/to/sbin/sems -d vr0 -f /path/to/etc/iptel/sems/sems.conf The name vr0 happens to be the ethernet interface that I am using, on Linux you'd mostly use eth0 for the same setup. All this runs on FreeBSD with adaptions that were checked into SVN, rev.39. Greetings, -Rick

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RE: [Users] Force RTP stream to go through mediaproxy
by Sam Lee 28 Apr '06

28 Apr '06

The problem appears when I tried to do that . The error I mentioned below was encountered, session.py didn't show any session establish if this error appears on the syslog. Any idea ? -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Wednesday, April 26, 2006 3:41 PM To: Sam Lee Cc: users(a)openser.org Subject: Re: [Users] Force RTP stream to go through mediaproxy Hi, most probably, in your script, you use mediaproxy only if NAT was detected (and a flag was set). So, you do not have to test the flag (to use mediaproxy only for nated calls), but call mediaproxy for all calls. regards, bogdan Sam Lee wrote: >Can I don't bother about the NAT test and push everything thru >mediaproxy ? >I can't quite understand why are there different NAT_Test , and the >example configs uses almost all of them... > >Regards, >Sam > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Wednesday, April 12, 2006 5:04 PM >To: Sam Lee >Cc: users(a)openser.org >Subject: Re: [Users] Force RTP stream to go through mediaproxy > >Hi, > >maybe you do not perform the proper nat tests. see: > http://openser.org/docs/modules/1.0.x/mediaproxy.html#AEN113 > >regards, >bogdan > > >Sam Lee wrote: > > > >>Hi all, >> >>I got the openser and mediaproxy up working and fine. >>The mediaproxy is able to work with certain NATED configurations, but >>not all. >>When i tried to check what is the problem , those NATED configurations >> >> > > > >>that were not working was found not to have made use of the >>mediaproxy, which causes the problem. >> >>What i am trying to do now, but without much success, is to force all >>the RTP media to go through the mediaproxy. Is there any way i can >>force all the RTP media to go through the mediaproxy without all the >>client_nat_test stuff...? >> >>Any hints will be much appreciated ! >> >>Regards, >>Sam >> >>---------------------------------------------------------------------- >>- >>- >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> > > >

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[Serusers] SERWEB (urgent)...path problem(include)
by fay butt 28 Apr '06

28 Apr '06

Hi, I am new user in ser, I still have a problem with SERWEB. I installed the serweb but it is not working in the browser. Can anybody tell me how to configure the serweb. serweb location folder is : /var/www/html/serweb/ when i keep any .php or .html file in /var/www/html/ it woks , but the serweb is not accessible with command http://localhost/serweb/admin/index.php. If anybody have the Idea about SERWEB configuration,plz let me know.... thanx --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.

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[Users] Regarding on OpenSER and softphone
by raviprakash sunkara 28 Apr '06

28 Apr '06

Hi Everybody. Just a small conformation about OpenSER and Softphone.. I'd installed openser and after so that install the sip softphone(X-Lite). After the settings in X-lite (sip-proxy..domain ), then i make a call ... But shows " Logging failed contact network admin " what's means this ?. I think I need to write a script ( modules ) in openser.cfg file for Register and authentication . how to make a calls and what are the steps should take in openser.cfgfile...? please help in this regarding and please excuse " in My English ".. Thanks and Regards.... Sunkara Ravi Prakash....:-)

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[Serusers] How to send a real phone number to pstn?
by wu yaqian 28 Apr '06

28 Apr '06

Hello,every one: I use ser-0.9.6.I have database authorize.I have accounts like 10000,10001 and so on.I want to send a real phone number (user difined) to pstn.This means when user 1000 call a pstn number he can send his real moblie phone number to the caller.(The caller can see this number). What should i do. Thanks a lot.

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RE: [Serusers] sccp to sip calling and vice versa
by Gould, Aaron 27 Apr '06

27 Apr '06

i just got it! bash-3.00$ serctl alias add 2002 sip:bob@mylab.com this allows calls from skinny ip phones (sccp) handled by a ccm to exit it's mtp/sip trunk then hit the sip proxy/registrar and have a way of resolution from number to name......the line 2 on my sip phone called "bob" rang aaron ________________________________ From: serusers-bounces(a)lists.iptel.org on behalf of Gould, Aaron Sent: Thu 4/27/2006 10:17 AM To: serusers(a)lists.iptel.org Subject: [Serusers] sccp to sip calling and vice versa does anyone understand how i would dial from a sccp phone using letters so that i could dial a sip aor url/uri ? i have a ccm 4.0(2)a configured with a sip trunk and mtp so that i can dial from sccp phones joined up to the ccm to sip phones that are joined up to the sip proxy/registrar that i have the ccm sip trunk pointing to but i can only do this if the username portion of the sip aor uri is numbers........like .... sip:2001@mylab.com then from the sccp phone i dialed 2001 and the sip phone rings. but what if my sip phone address is .... sip:bob@mylab.com ...how then do i dial the digits "bob" from the sccp phone? aaron _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] sccp to sip calling and vice versa
by Gould, Aaron 27 Apr '06

27 Apr '06

does anyone understand how i would dial from a sccp phone using letters so that i could dial a sip aor url/uri ? i have a ccm 4.0(2)a configured with a sip trunk and mtp so that i can dial from sccp phones joined up to the ccm to sip phones that are joined up to the sip proxy/registrar that i have the ccm sip trunk pointing to but i can only do this if the username portion of the sip aor uri is numbers........like .... sip:2001@mylab.com then from the sccp phone i dialed 2001 and the sip phone rings. but what if my sip phone address is .... sip:bob@mylab.com ...how then do i dial the digits "bob" from the sccp phone? aaron

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[Serusers] Integration of serweb with horde/imp
by hgaillac-sip＠yahoo.fr 27 Apr '06

27 Apr '06

Is it possible to integrate serweb to a framework such as horde/imp ? harry ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour retrouver directement vos services préférés : vérifiez vos nouveaux mails, lancez vos recherches et suivez l'actualité en temps réel. Rendez-vous sur http://fr.yahoo.com/set

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RE: [Users] avp_load_radius format
by Ricardo Martinez 27 Apr '06

27 Apr '06

This topic is very similar to my question in a past thread (http://openser.org/pipermail/users/2006-April/004277.html) I would like to add my experience. I'm using RADIATOR as my radius server. When i was trying to return 3 AVP's from my DB i was using the next syntax. SIP-AVP=state1:on SIP-AVP=state2:off SIP-AVP=state3:sip:12345@mydomain.com This was causing an Access-Accept message with the next SIP-AVP ATTRIBUTE: SIP-AVP = "state1:on<9>SIP-AVP=state2:off<9>SIP-AVP=state3:sip:12345@mydomain.com" in Radiator you need to add a "special" syntax (very similar to FreeRadius) SIP-AVP="state1:on" SIP-AVP="state2:off" SIP-AVP="state3:sip:12345@mydomain.com" That solve my problem!!!. Thanks to all. Best Regards, Ricardo Martinez.- > -----Mensaje original----- > De: Helge Waastad [mailto:helge@smartnet.no] > Enviado el: Jueves, 27 de Abril de 2006 6:48 > Para: Klaus Darilion > CC: users(a)openser.org > Asunto: Re: [Users] avp_load_radius format > > > Yes, > I was already thinking the same.... > > br hw > > > tor, 27,.04.2006 kl. 11.49 +0200, skrev Klaus Darilion: > > Maybe you can add your experiences to the Wiki (or Radius > docs)- there > > is already a Radius section. > > > > regards > > klaus > > > > Helge Waastad wrote: > > > Aha, > > > that's it. > > > > > > Thank's again. > > > > > > I had to alter the operator in FreeRADIUS: SIP-AVP += > "avp:value".... > > > > > > This happens when youre not really reading the man pages > thouroughly :-) > > > > > > > > > Again, > > > thanks a lot. > > > > > > br hw > > > > > > > > > ons, 26,.04.2006 kl. 17.59 +0300, skrev Bogdan-Andrei Iancu: > > >> Hi Helge, > > >> > > >> I think more lines with SIP-AVP should be placed into > reply. I can say > > >> this based on the openser code processing the reply - it > looks for more > > >> than one SIP-AVP RADIUS avps, each containing only one SIP avp. > > >> > > >> regards, > > >> bogdan > > >> > > >> > > >> Helge Waastad wrote: > > >> > > >>> Hi, > > >>> and thanks for your answer. > > >>> You'll have to excuse me for not 100% grasping the logic. > > >>> A typical RADIUS answer would be: > > >>> > > >>> rad_recv: Access-Accept packet from host > 195.18.134.150:1812, id=207, > > >>> length=38 > > >>> SIP-AVP = "voicemail:111111" > > >>> > > >>> So should more than one Access-Accept be sent with different > > >>> SIP-AVP=.... (I guess not) > > >>> > > >>> Or should the string value of SIP-AVP be consist of > several avps? > > >>> > > >>> I guess you can only respond with distinct attributes > from RADIUS? > > >>> > > >>> Rigth now I'm trying to see if I can do this by default > FreeRADIUS setup > > >>> or if should move into rlm_perl... > > >>> > > >>> ....probably a revised OpenSER-RADIUS Howto should be made. > > >>> I can, for that sake, help..when I get it working :-) > > >>> > > >>> br hw > > >>> > > >>> > > >>> > > >>> ons, 26,.04.2006 kl. 11.16 +0300, skrev Bogdan-Andrei Iancu: > > >>> > > >>> > > >>>> Hi Helge, > > >>>> > > >>>> you need to include several RADIUS AVPs into the > radius reply - one for > > >>>> each SIP AVP. > > >>>> > > >>>> regards, > > >>>> bogdan > > >>>> > > >>>> Helge Waastad wrote: > > >>>> > > >>>> > > >>>> > > >>>>> Hi, > > >>>>> I just wonder if someone could give me a hint on how > to load several > > >>>>> AVP's from RADIUS. > > >>>>> > > >>>>> I'm currently reponding with: > > >>>>> SIP-AVP=rpid:1111111 > > >>>>> which parses the rpid avp. > > >>>>> > > >>>>> But what should the response look like if I want to > load several avps? > > >>>>> (For group checking, ACl etc)? > > >>>>> > > >>>>> br hw > > >>>>> > > >>>>> > > >>>>> > > >>>>> > > >>>>> > > > -- > Helge Waastad > Senior Konsulent > Smartnet > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Users] speeddial bug?
by Frogger 27 Apr '06

27 Apr '06

I have been unable to use the speeddial module. There is some interesting behavior. Maybe a bug? When using: sd_lookup("speeddial"); The log shows an entry: "openser.speeddial" does not exist. openser db shows the table name as "speed_dial". Even when replacing names and using different combinations I cannot get speeddial to work. Ethereal will show multiple invites like openser is trying to find the record but it cannot. Any thoughts? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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