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37563 discussions

MAX_BRANCHES
by Alex Balashov 27 May '10

27 May '10

This may seem like an absurd question, but is there a particular reason why MAX_BRANCHES is #defined as 12 in config.h? Any particular reason for 12? Is it okay to make it higher? -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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prefix with kamailio
by alexis heron 27 May '10

27 May '10

hello I want to absorb a number with kamailio just to make out only three but I can not. In fact I send 1941 kamailio and i spring 941 with kamailio so I absorb a number. Here is how I : if ($rU=~"^1+941$") { log(1, "kamailio vers CCM 9xx"); route(10); } route[10] { log(1, "In route[10]"); rewritehostport("142.145.98.138:5060"); append_hf("P-hint: redirection CCM\r\n"); xlog("L_INFO", "$rm from $fu to $tu"); t_on_failure("2"); if (!t_relay()) { sl_reply_error(); }; exit; } Can someone help me?? best regards

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Question about inbound Alias Routing
by dotnetdub 26 May '10

26 May '10

Hi List, At the moment I use dbalias to send inbound from the PSTN <--> subscribers. I have a new opportunity that requires me to send inbound to a URI - SIP/353xxxxxxx(a)customer.proxy - not really sure how to accomplish this. I have a feeling there is a module I should be looking at to do this possibly? If someone could point me in the correct direction. TIA. Stephen.

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A Couple of Questions about carrierroute moduel
by JR Richardson 26 May '10

26 May '10

Hi All, I'm testing carrierroute module on version: kamailio 1.5.4-notls (i386/linux). I'm distributing calls to a group of Asterisk servers, all in the same carrier and domain, wanted to load balance the calls and I'm curious how this works. I see the 'probability' parameter in the carrierroute table and set it to equal value across all servers, I have 4 servers I'm sending calls to, so I set each prob value to '0.25'. I'm testing with sipp and hashing over call_id so this is the same for every call. I send 40 calls and the distribution of calls to the asterisk servers are 11, 12, 6, 11, pretty close to even but not 100%. How does the distribution algorithm work for carrieroute? Does it statefully track all active calls and load balance across each server in destination group? Is there another algorithm that can offer that can get closer to a round robin effect? Another question is do I need to include the record_route() and the t_check_trans() functions prior to calling the carrierroute functions? I see in the debug "DBG:core:forward_reply: found module tm, passing reply to it" and dialogs seem to process fine, so does the carrierroute module handle those requirements for stateful processing? Thanks. JR -- JR Richardson Engineering for the Masses

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Routing from Asterisk to Microsoft UM for voicemail.
by Nelson Pereira 26 May '10

26 May '10

If someone has a known good working route configuration for use with Microsoft Unified Messaging, please post your config. Thanks Nelson Pereira

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Calculate ha1 with mysql trigger
by Ole Kaas 26 May '10

26 May '10

Hello, When editing the subscriber table using phpmyadmin, updating the ha1 fields are no fun and error prone. The trigger below will calculate the hashes if the input fields are left empty. Navigate to the subscriber table in phpmyadmin and drop the code in the query field under the "SQL" tab. Remeber to set the delimiter to // /Ole DROP TRIGGER IF EXISTS calcha1 // CREATE TRIGGER calcha1 BEFORE INSERT ON subscriber FOR EACH ROW BEGIN IF NEW.ha1 = "" THEN SET NEW.ha1 = MD5(CONCAT(NEW.username,":",NEW.domain,":",NEW.password)); END IF; IF NEW.ha1b = "" THEN SET NEW.ha1b = MD5(CONCAT(NEW.username,"@",NEW.domain,":",NEW.domain,":",NEW.password)); END IF; END//

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'serctl add <name> <password> <email>'
by Sanjeev Singh 26 May '10

26 May '10

Hi All, Please help me this i am using below command to add subscribers in the ser so that i can make a phone call. 1) what is the meaning of this command 2) how to use this command? *'serctl add <name> <password> <email>'* can you please help me on this also. 1) how to register my sip client with ser server 2) how to add subscribers so that i can make a phone call. Regards, Sanjeev Singh

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permissions module
by Asim Riaz 25 May '10

25 May '10

Hello List, from the documentation, permissions.allow, allows the calls either by matching From-URI or R-URI and permission.deny denies the call from matching uri. Do I need to divide the file into different sections, for R-URI and for From-URI? Not sure exact syntax could someone explain me that if I want to match in permissions.allow if a phone 10.10.10.10 dial(r-uri) 1234 allow it and everything else is blocked? What is the preference does kamailio check for permissions.deny first if there is no matching then checks permissions.allow? Thanks in advance, Asim

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Re: [SR-Users] sipp and stateful transaction problem
by Daniel-Constantin Mierla 25 May '10

25 May '10

Hello, On 5/22/10 2:22 AM, JR Richardson wrote: > On Fri, May 21, 2010 at 4:46 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: > >> Hello, >> >> On 5/21/10 10:47 PM, JR Richardson wrote: >> >>> Hi All, >>> >>> I'm doing some testing with kamailio 1.5: >>> >>> kamailio1:/etc/kamailio# kamailio -V >>> version: kamailio 1.5.4-notls (i386/linux) >>> flags: STATISTICS, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, >>> SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >>> MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 4194304 >>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >>> svnrevision: 2:6005M >>> @(#) $Id: main.c 5608 2009-02-13 16:48:17Z henningw $ >>> main.c compiled on 10:14:11 May 18 2010 with gcc 4.3.2 >>> >>> Using dispatcher module trying to load balance SIP calls across some >>> Asterisk servers. I have it working fine when I test in this >>> scenario: >>> >>> sip phone dial out><asterisk><kamailio><round robin to several asterisk >>> servers >>> >>> This works stateful and stateless, handles everything gracefully. >>> >>> This scenario is giving me fits: >>> >>> sipp dial out><kamailio><round robin to several asterisk servers >>> >>> I get retransmits on every call back to sipp with errors like >>> "Discarding message which can't be mapped to a known SIPp call" and >>> "SIP/2.0 481 Call leg/transaction does not exist" >>> >>> This happens with kamailio setup stateful or stateless. I'm wondering >>> if sipp is the problem or just doesn't play well with kamailio? >>> >>> I've kept the config as simple as possible for testing, it is listed >>> here http://pastebin.com/BZ8hJvJv >>> >>> Here is my sipp usage: >>> >>> sipp -sn uac 10.10.12.53 -i 10.10.14.97 -s 55 -d 7000 -l 10 -r 1 >>> -trace_err >>> >>> Any insight would be appriciated. >>> >>> >>> >> the problem is in your sipp scenario. The uac calls do not map to uas. >> kamailio does not reply 481, check the uas scenario, that is the one that >> sends back the 481. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> Kamailio (OpenSER) Advanced Training >> Miami, Fl, USA - June 21-23, 2010 >> http://www.asipto.com/index.php/kamailio-advanced-training/ >> >> >> > Thanks Daniel, I reveiwed the sipp docs, '-sn uas' just sits there as > a responder, it will not initiate calls to kamailio. I don't > understand what you are getting at? How would I use this type of > scenario to test? > keeping the mailing list cc-ed is recommended, since others can respond faster and new people can benefit of the discussion. What I wanted to say is that kamailio does not reply 481. So the problem is in the responder of requests sent by UAC and forwarded by Kamailio. Somehow, the dialog is destroyed before the BYE (or other in-dialog request) is sent by UAS. If you can grab the SIP trace of such call (e.g., using ngrep on kamailio server), I can give more hits (try to select the sip flow for one such call only, sending full sip trace will be too big). Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

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Re: [SR-Users] kamailio 3.0.2 for Thu, May 27, 2010
by Daniel-Constantin Mierla 25 May '10

25 May '10

On 5/25/10 8:51 PM, SIP wrote: > Daniel-Constantin Mierla wrote: > >> Hello, >> >> I am planning to release kamailio 3.0.2 this Thursday. There were some >> fixes since 3.0.1 that worth to be packaged. If you have major reports >> for current stable version, please write to sr-dev(a)lists.sip-router.org >> >> Cheers, >> Daniel >> >> > Is this a drop-in upgrade from 3.0.1 (i.e. recompile modules and core > and keep the rest in place)? > > yes, changes to last number in version string mean that config language and database structure are the same. This is a minor release, bringing only fixes to the code. So upgrading from 3.0.0 or 3.0.1 to 3.0.2 is exactly what you said, just recompile and reinstall modules and core, keep the rest in place. Cheers, Daniel -- Daniel-Constantin Mierla Kamailio (OpenSER) Advanced Training Miami, Fl, USA - June 21-23, 2010 http://www.asipto.com/index.php/kamailio-advanced-training/

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