sr-users October 2005

sr-users@lists.kamailio.org

214 participants
458 discussions

[Serusers] Problem with CANCEL and call redirection
by Xavier TRENTIN 19 Oct '05

19 Oct '05

Hi all, I'm trying to do a serial call forwarding if callees don't answer. I've added a field in a table which is the number the users want to forward the call if they don't answer. I've a shell script (myScript.sh) which works like that: If the forward mode is activated for the user, then the address is overwritten and the script returns 0, else it returns 1. My main route ends by "t_on_failure("1");" in my ser.cfg, and I've write these lines after the main route: /failure_route[1] { if (exec_dset("/path/to/myscript/myScript.sh \"$SIP_HF_TO\"")); { route(2); } break; } route[2] { sl_send_reply("300","Redirect"); break; } / I also use the tm module and I've set the fr_inv_timer and the fr_timer to the values I wanted. In my configuration : User 1 wants to forward the call to User 2. User 2 wants to forward the call to User 3. User 3 wants to forward the call to User 1. When User 4 calls User 1, who doesn't answer, the call is well forwarded to User 2 after 5 ringings. User 2 doesn't answer and the call is well forwarded to User 3 . Etc. So it works fine, if at least one of the callee answers. But if the caller hangs up (sending a CANCEL to the callee), the phones are always ringing and forwarding the call in the same order. Does anybody have an idea to solve this problem? Thanks. B.R. Xavier.

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[Serusers] Can I send Binary Data through SIP IM?
by Abhijit A. Mahajani 19 Oct '05

19 Oct '05

Hi, Can I send Binary Data thorough SIP IM? Does The SIP IM has an option to set the data as Binary? As far as i know, SIP IM genarally process text data. Thanks and regards. Abhijit

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[Serusers] [OT] MAX TNT as a Media Gateway
by Rodrigo P. Telles 19 Oct '05

19 Oct '05

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Folks, I used MAX TNT for dial-up users in the past and now I have a choice to update that equipment to support SIP and act as a PSTN/Media Gateway. I have some doubts about it: 1 - Is there some one using it? (what hardware configuration?) 2 - Does it support all CODEC's? 3 - Does it 'speak' SIP well with SER/OpenSER? (any reference about supported RFC's?) 4 - Any other additional information is welcome! Thanks in advance. - -- ============================================ Rodrigo P. Telles <telles(a)devel.it> TI Manager Devel-IT - http://www.devel.it IVOZ # 1029 +55 14 3324-1200 Bestcom Group ============================================ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDVBViiLK8unYgEMQRAtUYAJ9b8c5CAMd+Exxdcc/mtEvP927zIwCff6ZE ajusWOEWVjxGEVzo+SSQyZg= =FHXs -----END PGP SIGNATURE-----

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[Serusers] WM Contact problem
by Geovany Andres Gonzalez Ruiz 18 Oct '05

18 Oct '05

Hello, I am using SER + MySQL, and I had a problem. When I used the WM software to register against the SER in the Location table only had in the contact field the domain of WM, I meant The first message was: 192.168.0.138:5060 -> 192.168.0.184:12469 SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.0.184:12469.. From:<sip:gagonzal@cursovoip.com>;tag=09c07cd1-f595-4fff-bcf9-7b12e6721bf5.. To:<sip:gagonzal@cursovoip.com>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7529..Call-ID: 1ee7bb57-9c25-4e65-b4ac-4264317ebe7d(a)192.168.0.184.. CSeq: 2 REGISTER.. Contact:<sip:192.168.0.184:12469>;expires=1200..Server: Sip EXpress router (0.9.3(i386/linux)).. Content-Length: 0..Warning: 392 192.168.0.138:5060 "Noisy feedback tells: pid=9156 req_src_ip=192.168.0.184 req_src_port=1083 in_uri=sip:cursovoip.com out_uri=sip:cursovoip.com via_cnt==1" serctl ul show ...Record(0xb5ebff88)... domain: 'location' aor : 'gagonzal' ~~~Contact(0xb5ec01f8)~~~ domain : 'location' aor : 'gagonzal' Contact : 'sip:192.168.0.184:12469' Expires : 1157 q : Call-ID : '1ee7bb57-9c25-4e65-b4ac-4264317ebe7d(a)192.168.0.184' CSeq : 2 User-Agent: 'Windows RTC/1.0' received : '' State : CS_SYNC Flags : 0 next : (nil) prev : (nil) Someone knows how could I change the behavior of windows messanger to send the contact field like that (<sip:192.168.0.184:12469>) without specific information of the user? or Someone knows how could I prepare SER to manipulate the REGISTER message and put in the contact field in the location table the information from URI ?? Thanks in advance. Atentamente, Geovany Andrés González Ruiz Curriculum Lead (CL) CATC (Cisco Academy Trainig Center) Colombia-Ecuador Universidad Nacional de Colombia http://cnap.unalmed.edu.co

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[Serusers] 423 message for Register Expires
by share phone 18 Oct '05

18 Oct '05

Hi All, The default_expires is 600 and my min is 300 and max expires is 600 when i send the REGISTER message with 3600 it should send me 423 with 600 the openser.cfg do not do it. openser.cfg modparam("registrar","default_expires",600) modparam("registrar","min_expires",600) modparam("registrar","max_expires",1200) Is there any settings that need to be done. Regards, sunil. --------------------------------- Yahoo! Music Unlimited - Access over 1 million songs. Try it free.

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Re: [Users] Openser as non root
by Daniel-Constantin Mierla 18 Oct '05

18 Oct '05

apart of the command line options (see openser -h) you can set some parameters in the configuration file in order to switch the user and group. Check the dokuwiki: http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook#group_gid http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook#user_uid Cheers, Daniel On 10/18/05 11:40, Nelson Silva wrote: > > Tavis: want to share how you did it ? > > thk > > Nelson Silva > ----------------- > email: nelson.silva(a)neuvex.com > website: http://www.neuvex.com > > > Date: Mon, 17 Oct 2005 11:05:21 -0700 > From: Tavis P <tavis.lists(a)galaxytelecom.net > <javascript:top.opencompose('tavis.lists(a)galaxytelecom.net','','','1')>> > Subject: Re: [Users] Openser as non root > To: Users(a)openser.org > <javascript:top.opencompose('Users(a)openser.org','','','1')> > Message-ID: <4353E7E1.2020109(a)galaxytelecom.net > <javascript:top.opencompose('4353E7E1.2020109(a)galaxytelecom.net','','','1')>> > Content-Type: text/plain; charset=ISO-8859-1 > > I'm currently running OpenSER both as non root and also chrooted without > issue > > Nelson Silva wrote: > > > Anyone got a info how to make openser for non root ? > > > > for example, openser -U openser -G openser ? > > > > > > > > thk > > > > Nelson Silva > > ----------------- > > email: nelson.silva(a)neuvex.com > <javascript:top.opencompose('nelson.silva(a)neuvex.com','','','1')> > > website: http://www.neuvex.com > <http://webmail.neuvex.comparse.pl?redirect=http%3A%2F%2Fwww.neuvex.com> > > > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > <javascript:top.opencompose('Users(a)openser.org','','','1')> > >http://openser.org/cgi-bin/mailman/listinfo/users > <http://webmail.neuvex.comparse.pl?redirect=http%3A%2F%2Fopenser.org%2Fcgi-b…> > > > > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Serusers] call drops after about 30 seconds
by maka 18 Oct '05

18 Oct '05

hello everyone, I am using ser-0.9.0 together wit asterisk-1.0.6. I am testing it with a couple of hardware AT-320 IP phones from Atcom, using a PA168S chip. They seem to be working fine, with stun behind nat, and they can call other user-agents (softphones, asterisk adn the pstn through it), but whenever I make a call between the two phones, everytime the call is dropped after about 34 seconds, even when calling between phones on Public IP addresses. I am actually clueless why this happens, I tried changing the NAT ttl value, bu to no effect, and it is not supposed to be a codec mismatch too since both phones use absolutely the same codecs, in the same order of preference. Appreciate the feedback, cheers -- I'm sick and tired of being sick and tired...

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[Serusers] mangling request URI according to To: and userlocation value
by Francesco Fondelli 18 Oct '05

18 Oct '05

Hi all, let's say SER receives this invite: INVITE sip:foopbx@ip.com SIP/2.0 Via: ... To: "058633345880101" <058633345880101(a)ip.com> [cut] and assume that there is an entry in the user location table for ''foopbx(a)ip.com" (ie it registered with SER) and assume that entry in userloc has a "Server: Bar 1.0 PBX" value I'm wondering if is possible to do something like that: if (!lookup("location")) { log(1, "User not in user database...\n"); route(whatever); break; }; if(get_server_string_from_userloc() =~ "Bar 1.0 PBX") { take the number from the "To:" header and write it in the request uri. So the forwarded msg will look like: INVITE sip:058633345880101@ip.com SIP/2.0 Via: ... To: "058633345880101" <058633345880101(a)ip.com> [cut] }; thank you very much, I really need your help Ciao FF

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[Serusers] Billing System Asterisk works
by Mr Greg Plater 18 Oct '05

18 Oct '05

AstBill http://astbill.com/whatis This billing system I have implemented and tested to work for both prepaid and post paid billing Simple to setup with asterisk and easy to manage. Uses: AstBill Drupal Mysql Apache Calls pass to Asterisk from ser for termination to PSTN. They are rated and billed. I have edited SERWEB to display account records and current prepaid balances from Astbill database Check it out it works

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Re: [Users] Openser as non root
by Nelson Silva 18 Oct '05

18 Oct '05

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sr-users October 2005