[OpenSIPS-Users] Prepaid_lock problem in CDRTool

2010-01-04 Thread ASHWINI NAIDU
Hi all, I have installed CDRtool version 6.9.9. *I have set the Prepaid_lock to 1* to disable multiple calls from a single prepaid account. But the parallel call gets established. How can I solve this problem $RatingEngine=array(socketIP = xxx.xxx.xxx.xxx,

Re: [OpenSIPS-Users] Opensips 1.6.1 dos not start on Ubuntu 9.1 64bits

2010-01-04 Thread Antonio Anderson M. de Souza
Bogdan, I started the Opensips with debug=6 and there weren't changes on the initialization log, I've used the strace (follow trace attached), I'm not an expert in the strace, but the only thing that i could see was a message 14752 read(5, O\0\0\2\377\24\4#42000Access denied for u..., 16384) = 83

[OpenSIPS-Users] Can OpenSips support SBC?

2010-01-04 Thread 尹林
nbsp;Hi, I want to know whether the opensips support SBC with dual NICs, If yes, whether opensips can be used to construct a sbc cluster with the function of load balancing, and is there any performance test data as a sbc? ths, yinlin___ Users

Re: [OpenSIPS-Users] Need help for Call to another network

2010-01-04 Thread John Quick
Ahmed, One way to avoid the problem of Asterisk returning 491 Request Pending would be to authenticate the call between the UA and Opensips (see proxy_authorize and proxy_challenge functions in modules auth and auth_db) then relay it to Asterisk with IP address authentication only. i.e. so

Re: [OpenSIPS-Users] B2BUA not passing ACKs

2010-01-04 Thread Bogdan-Andrei Iancu
Hi Brian, the error you posted has nothing to do with blocking / non-blocking - it simply shows that opensips fails to open a new TCP connection because there was nobody listening on the other side. Regards, Bogdan opensipsl...@encambio.com wrote: Hello list, It seems that in August 2008

Re: [OpenSIPS-Users] need advice on B2b

2010-01-04 Thread Anca Vamanu
Hi Ha`, There is a very simple example in the documentation: route[b2b_request] { xlog(b2b_request ($ci)\n); } route[b2b_reply] { xlog(b2b_reply ($ci)\n); } You can call in these routes any function that you call in a request route. Regards, -- Anca Vamanu www.voice-system.ro ha do

Re: [OpenSIPS-Users] B2BUA not passing ACKs

2010-01-04 Thread Anca Vamanu
Hi Brian, There is a misunderstanding from your side in what the b2b scenario documents are concerned ( please read carefully the documentation - http://www.opensips.org/Resources/B2buaTutorial ). The important thing is that there should only be rules in the scenario for requests that need a

Re: [OpenSIPS-Users] compile deb files 1.6.1

2010-01-04 Thread Bogdan-Andrei Iancu
Hi Jan, are you sure you have the opensips-mysql-module_1.6.1.0_amd64.deb file in the current dir (or name is correct) ?? I just tried the install of the same packages (freshly generated from svn tree) and I had no issue. Regards, Bogdan Jan D. wrote: Bogdan, Thanks for the quick response,

Re: [OpenSIPS-Users] Opensips 1.6.1 dos not start on Ubuntu 9.1 64bits

2010-01-04 Thread Bogdan-Andrei Iancu
Hi Antonio, The relevant part is: 14752 socket(PF_FILE, SOCK_STREAM, 0) = 5 14752 fcntl(5, F_SETFL, O_RDONLY) = 0 14752 fcntl(5, F_GETFL) = 0x2 (flags O_RDWR) 14752 connect(5, {sa_family=AF_FILE, path=/var/run/mysqld/mysqld.sock}, 110) = 0 14752 setsockopt(5, SOL_SOCKET,

Re: [OpenSIPS-Users] Opensips 1.6.1 dos not start on Ubuntu 9.1 64bits

2010-01-04 Thread Antonio Anderson M. de Souza
Bogdan, Thank you very much, the problem was because I forgot to add the module dialplan in the modparam db_url: Before: modparam(domain|alias_db|auth_db|usrloc|drouting,db_url, mysql://user:u...@localhost/db) After: modparam(domain|alias_db|auth_db|usrloc|drouting*|dialplan*,db_url,

Re: [OpenSIPS-Users] Opensips 1.6.1 dos not start on Ubuntu 9.1 64bits

2010-01-04 Thread Bogdan-Andrei Iancu
Hi Antonio, It should not work on any OSmaybe your mysql setups are different on the OSs you tested. NOTE that db_url has a default value which may work if the DB was created with default access users. Regards, Bogdan Antonio Anderson M. de Souza wrote: Bogdan, Thank you very much, the

Re: [OpenSIPS-Users] NAT per call leg

2010-01-04 Thread Bogdan-Andrei Iancu
Hi Daniel, Daniel Goepp wrote: I'm going to look the complete fool here for my lack of understanding how OpenSIPS would handle this via it's route branching, but I am just banging my head on this one. Here is the problem: For simplification I'm using the first octet of the IP to identify

Re: [OpenSIPS-Users] .Can OpenSips support SBC?

2010-01-04 Thread Bogdan-Andrei Iancu
Hi YinLin, Well, for opensips, the cps (calls per second) is the most relevant param. And the value is higly depending on the complexity of the configuration/routing script. You can do a simple SBC for NAT, but also you can do (as have recently implemented) and SBC with TLS, NAT traversal,

Re: [OpenSIPS-Users] re-REGISTER with To tag gets 404

2010-01-04 Thread Bogdan-Andrei Iancu
Hi Jeff, should be ok this way. Regards, Bogdan Jeff Pyle wrote: Is there any danger in adapting the default config from: if (has_totag()) { ... } to: if (has_totag() !is_method(REGISTER)) { ... } - Jeff On 12/31/09 7:46 AM, Olle E. Johansson

Re: [OpenSIPS-Users] something on stun documment

2010-01-04 Thread Bogdan-Andrei Iancu
Hi Ha, the alternate port is whatever port you want to use as secondary port by the STUN server. 3479 is the default alternate port in STUN, so it is ok. Regards, Bogdan ha do wrote: Hi admin http://www.opensips.org/html/docs/modules/devel/stun.html#id227269 1.3.3. |alternate_ip|

Re: [OpenSIPS-Users] Send simultaneous commands with exec module

2010-01-04 Thread Bogdan-Andrei Iancu
Hi Jennifer, A Happy New Year to you too! for CMD1 and CMD2, try from command line (CMD1 CMD2) or (CMD1 CMD2) Regards, Bogdan Jennifer-4 wrote: Hi everybody and Happy New Year!! I´m working with Opensips 1.4. I need to send two commands simultaneously from Opensips,

Re: [OpenSIPS-Users] Users Digest, Vol 18, Issue 2

2010-01-04 Thread Bogdan-Andrei Iancu
Hi Ahmed, Asterisk is confused (by sending the request pending reply to second CALL - csesq 2) because it never received an ACK for the reply it sent for the first call (cseq 1). What is strange is that I do not see any ACK neither from OpenSIPS to Asterisk, nor from UAC to OpenSIPSCould

Re: [OpenSIPS-Users] Load balancer probe _mode=1 bug?

2010-01-04 Thread Bill W
Hey Bogdan, It looks like that fixed it. Thanks so much! Bill Bogdan-Andrei Iancu wrote: Hi Bill, Could you please try the attached patch? It seams that there was an issue with the the probing values in the code. Let me know if the patch does solves your problem and I will upload it

Re: [OpenSIPS-Users] Load balancer probe _mode=1 bug?

2010-01-04 Thread Bogdan-Andrei Iancu
Super! I uploaded the fix on SVN. Thanks and regards, Bogdan Bill W wrote: Hey Bogdan, It looks like that fixed it. Thanks so much! Bill Bogdan-Andrei Iancu wrote: Hi Bill, Could you please try the attached patch? It seams that there was an issue with the the probing values in

Re: [OpenSIPS-Users] Media-dispatcher

2010-01-04 Thread Richard Revels
Interesting. I think you may have mentioned this before Dan but I didn't catch it for some reason. That makes installing media proxy on CentOS / RHEL 5.x easier. I've found that creating an RPM to install python 2.5 (along with the 2.4 rather than upgrading it) and then using the virtualenv

Re: [OpenSIPS-Users] 3rd party Client sending 5xx

2010-01-04 Thread Jeff Pyle
Please see: http://www.apps.ietf.org/rfc/rfc3261.html#sec-21.5 On Jan 4, 2010, at 9:53 PM, Aditya Kumar wrote: Hi all, I am inter working with a 3rd party SIP UA and I see they are sending 503/500. Can any one tell me what are the cases at which a SIP UAS will sent 503 or 500

Re: [OpenSIPS-Users] 3rd party Client sending 5xx

2010-01-04 Thread Aditya Kumar
Thanks Jeff for the link. I did see 3261. Question is : RFC will tel high level about response codes. I am trying to understand in particular when a End User (UAS) will sent. I am clear from a proxy perspective. Was looking some one can share your experiences when a UAS will send 5xx

Re: [OpenSIPS-Users] 3rd party Client sending 5xx

2010-01-04 Thread Jeff Pyle
Aditya, That all depends on the switch. Asterisk sends a 503 if you look at it the wrong way, that is, under many conditions. In my experience most carriers will send a 503 when their upstream paths are full (or your concurrent calls to them is full) and they expect you to route advance to

[OpenSIPS-Users] How to implement access control list on opensips

2010-01-04 Thread Ahmed Munir
Hi, I want to implement ACL on OpenSIPs to accept the call on behalf of source URI + IP address. Can anyone tell me which modules and functions are required for it? Also kindly share some example template with it. -- Regards, Ahmed Munir ___ Users