sr-users February 2008

sr-users@lists.kamailio.org

146 participants
260 discussions

Re: [OpenSER-Users] [OpenSER-Devel] next irc meeting

by Daniel-Constantin Mierla

Hello Henning, On 02/13/08 14:56, Henning Westerholt wrote: > On Tuesday 12 February 2008, Daniel-Constantin Mierla wrote: > >> Hello, >> >> I think is time for a new irc meeting to synchronize on the work for 1.4 >> and discuss 1.3.1 about release. I propose Friday, Feb 15, UTC 13:00. >> >> I created a wiki page to gather the discussion items there and have the >> agenda ready for that date. I added what I would like to touch. Please >> add yours and complete/comment on mine or others. >> > > Hi Daniel, > > good idea! I've already added some of my points to the list. This friday is > not a good date for me, as i'm on vacation this day. Perhaps we can go for > the Tuesday, Feb 19, UTC 13:00? > ok, it is fine with that date, too. I will update the docuwiki pages (added users list so they get the update). Cheers, Daniel > Cheers, > > Henning > >

16 years, 3 months

[OpenSER-Users] Wanted: VoIP Engineer for Switerland

by laurent schweizer

Peoplefone AG offers Voice over IP(VoIP) services with exceptional rates. Peoplefone is a certified partner of Siemens<http://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1&sdc_sid…>and AVM/FRITZ!Box <http://www.fritz-shop.ch/> . Due to our rapid growth, we are currently seeking for: VOIP SPECIALIST Place of work: Zurich *Requirements:* - Graduation from college or university with a Bachelor's degree (preferably IT) - Experience with PHP - Practical knowledge of C and C++ - Practical knowledge of Mysql and Postgresql - Linux experience - Knowledge of IP Networks, UDP, TCP - Experience with tools for Network analysis like Ethereal - VoIP basic knowledge, VoIP servers, configuration of devices - Fluency in English - Ability to interact with individuals and groups at all levels - Detail oriented and analytical - Strong verbal and written communication skills Knowledge of Perl, Java, Asterisk / SER / OPENSER, ability to configure routers, Cisco or Patton gateways, knowledge of SIP and STUN protocol, knowledge of NAT problems, of outbandProxy, knowledge of monitoring tools like Cactus, Nagios, MRTG or high availability tools like DRBD, Hearthbeat would be an additional asset. Interested individuals are requested to send their resume to: christophe.beaud(a)peoplefone.com

16 years, 3 months

[OpenSER-Users] ERROR: couldn't open sequence file /var/run/radius.seq

by Iñaki Baz Castillo

I see this issue in syslog: ---------------- openser: rc_get_seqnbr: couldn't open sequence file /var/run/radius.seq: Permission denied --------------- The problem is that OpenSer run with "openser" user who has not write access to /var/run. If I change permissions to /var/run the the message doesn't appear. Anyway I'd prefer not to modify permissions and owner of a system directory as /var/run. - How important is that "radius.seq" file? - Is not possible to configure OpenSer to write it in /tmp (for example)? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

16 years, 3 months

Re: [OpenSER-Users] FreeRADIUS-CDRTool Prepaid Connector 1.1 Released

by A.smith

Hi Dan, Thanks for the all the info! Ok, question on this part: 3. Accounting it is bit more accurate (you have the session total duration inside the accounting packets), so radius will be no longer responsible of calculating the session durations from timestaps. To interoperate with CDRTool you still need the data in the radacct table formatted as per normal by Freeradius, right? By accouting packet are you referring to the radius accouting packets? If you say Yate puts session times in the accouting packets how does this all stick together? Does OpenSER still do the radius accouting part, or do I need to configure Yate to do it instead? thanks a lot! Andy. --------- Original Message -------- From: Dan-Cristian Bogos <danb.lists(a)googlemail.com> To: A.smith <a.smith(a)ukgrid.net> Cc: users(a)lists.openser.org Subject: Re: [OpenSER-Users] FreeRADIUS-CDRTool Prepaid Connector 1.1 Released Date: 13/02/08 12:15 > Hi Andy,The original config was built with Yate in mind due to openser incapacity (before release 1.3) of disconnecting the calls. Since 1.3.0 the dialog module should be able to timeout the calls, in theory you should no longer need extra software like Yate. > I would still recommend using Yate combined with OpenSER in the case you are doing some sort of "Carrier business", for  the following reasons:1. It creates two different legs for your call (in and out) same as Cisco does, and hides one side from the other (eg. removes the via headers and any revealing ip information inside SDP - so your partners should not know where the traffic comes from). > 2. You have more protocols available in.3. Accounting it is bit more accurate (you have the session total duration inside the accounting packets), so radius will be no longer responsible of calculating the session durations from timestaps. > 4. Yate can work in rtp_forward mode, therefore no extra overhead given by rtp processing.So basically what the connector does (as specified in the documentation), for each call which is authorized by radius, the connector will ask permission from cdrtool. If permission is granted, it will return in a avp to openser the maximum duration allowed for the call (timeout value) plus credit available, for the case of special uas able to display that. > By receiving accounting stop packet, the connector will inform cdrtool about call disconnection therefore clearing the lock and debiting the balance inside cdrtool. The rtp stream has nothing to do with this scenario, so you don't need to touch your NAT support configuration, it's all in the signaling. > Let me know if you need further info.Cheers,DanBOn Feb 13, 2008 12:53 PM ________________________________________________ Message sent using UK Grid Webmail 2.7.9

16 years, 3 months

[OpenSER-Users] Routing Problem "To Many Hops"

by Ali Jawad

Hi All I am trying to achieve the following 1. User Calls a number * If the number is 00+all numbers go to PSTN -> Else check if the user is existent in our SIP Proxy user pool and act accordingly What I achieved so far is the following 2. User Calls a number * Route checks if the number is online if yes the call is routed to him (route[1]) if not it send the call to PSTN if it is 00+[0-9] route [2]. This works fine for 00+[0-9] number however if 123456 is called and the user 123456 is not online, the call goes down through the conditional statements because it does not meet 00+[0-9] and I get an error "To Many Hops" Seems I messed up somewhere, can anyone help me fix this http://pastebin.com/m50328c12 Thanks

16 years, 3 months

Re: [OpenSER-Users] OpenSER - a slow starter?

by Lars Skjærlund

Hi Klaus, > MAybe openser tries t do some DNS lookups which may take long time. > > The best would be to set debug=4 and "tail" to the logfile > (default=/var/log/syslog (or /var/log/messages on RedHat&co)). I was running at debug level 6 already, and dumping the output to screen. However, I actually found the error last night: Beeing new to all this, I had mediaproxy and proxydispatcher running on the machine - primarily as a proof to myself that they where installed and seemed to work. I assumed they wouldn't interfere with OpenSER as long as the support modules wasn't loaded, but alas, I was wrong: As soon as I stopped mediaproxy and proxydispatcher, OpenSER started working as expected. The problem is reproduceable - but I think it's a bit strange behaviour? Regards, Lars -- Lars Skjærlund Skovengen 111 2980 Kokkedal Denmark Tel.: +45 70 25 88 10 http://www.skjaerlund.dk/lars

16 years, 3 months

[OpenSER-Users] Implementing Follow-me

by Cosimo Fadda

Hi, I'm very new to this list. In my setup I use openser as proxy an asterisk as pstn gateway. I want to implement a 'follow-me' service, i.e. I want users to dial a particular number and then they can enter a phone number where receive their incoming calls. So which is the best solution: use asterisk to perform a lookup in the DB and decide to which number route the call, ore use openser maybe with the 'exec_dset' command? Or there is a command that I'm missing that can implement this function? Thanks in advance, Cosimo -- *Ing. Cosimo Fadda System Engineer Klarya s.r.l.* Sede Operativa: Via Agnini 47 - 41037 Mirandola (MO) Tel +39 0535 27325 Fax +39 0535 610050 Mob +39 3316907200 Sede Legale: Via Salgari, 17 - 41100 Modena Tel +39 059 821332 Fax +39 059 821492 email: cosimo.fadda(a)klarya.it web: www.klarya.it * **Open Source IP Communications* __________________________________________________

16 years, 3 months

[Serusers] Do an anonymous call

by inge

Hi all, I've implanted this feature some months ago and it no longer appears to be functional. To do that I have used a subst function to parse and replace sip:XXX| <sip:XXX> by "Anonymous" sip:|<sip:>. It follows to run on some products but not on an other one and only between them (ie. on-net between the products concerned). If someone can help me/guide me. Thanks ! Regards, Adrien

16 years, 3 months

[Serusers] TLS module - memory allocation functions error - rel 2.0.0_rc6

by Tomasz Zieleniewski

HI, I encountered the following error while trying to set up tls in ser. When I start ser I get such error: Feb 12 12:40:51 venom ser[26157]: tls: _init_tls_h: compiled with openssl version "OpenSSL 0.9.8g 19 Oct 2007" (0x0090807f), kerberos support: off, compression: on Feb 12 12:40:51 venom ser[26157]: tls: init_tls_h: installed openssl library version "OpenSSL 0.9.8g 19 Oct 2007" (0x0090807f), kerberos support: off, zlib compression: on compiler: gcc -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -m64 -DL_ENDIAN -DTERMIO -O3 -Wa,--noexecstack -g -Wall -DMD32_REG_T=int -DMD5_ASM Feb 12 12:40:51 venom ser[26157]: ERROR: tls_init.c:366: Unable to set the memory allocation functions Feb 12 12:40:51 venom ser[26157]: could not initialize tls, exiting... Here is my config: # -- tls params -- modparam("tls", "config", "/home/sip/ser/2.0/hsp/ca/tls.cfg") modparam("tls", "tls_force_run", 0) modparam("tls", "tls_log",3) modparam("tls", "handshake_timeout", 10) modparam("tls", "send_timeout", 10) modparam("tls", "tls_disable_compression", 0) #modparam("tls", "private_key", "/etc/certs/key.pem") #modparam("tls", "certificate", "/etc/certs/cert.pem") #modparam("tls", "ca_list", "/etc/certs/ca_list.pem") #modparam("tls", "require_certificate", 0) #modparam("tls", "verify_certificate", 1) #modparam("tls", "tls_method", "TLSv1") and tls.cfg [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /home/sip/ser/2.0/hsp/ca/hsp_key.pem certificate = /home/sip/ser/2.0/hsp/ca/hsp_cert.pem ca_list = /home/sip/ser/2.0/hsp/ca/hsp_calist.pem [client:default] verify_certificate = yes require_certificate = yes Please point me what did I miss if this is the right behaviour? Kind regards Tomasz

16 years, 3 months

[OpenSER-Users] Implemented RFCs

by Victor Pascual Ávila

Hi Folks, Is there any document where I can find the implemented RFCs in Openser? Creating a wiki document would be useful. Cheers, -- Victor Pascual Ávila Research Engineer Tel. +34 93 542 2906 Fax. +34 93 542 2517 Research Group on Network Technologies and Strategies (NeTS) Universitat Pompeu Fabra (UPF) Pg. de Circumval·lació, 8 Office 358 08003 Barcelona (Spain) http://nets.upf.edu/

16 years, 3 months

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sr-users February 2008