sr-users May 2010

sr-users@lists.kamailio.org

67 participants
106 discussions

functions for modifying pAsserted
by Alex rsm 27 May '10

27 May '10

Hi, I couldn't find the module that has the functions for modifying pAsserted field. Your help would be appreciated. Thanks, AR _________________________________________________________________ Hotmail has tools for the New Busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:…

3 2

Siremis 1.0 login fails (or, does nothing)
by Joe Micciche 27 May '10

27 May '10

RHEL5.5 server Kamailio 3.0 built from source via Git Apache/2.2.3 PHP 5.3.2 MySQL 5.1.47 chown -R apache /var/www/html/siremis-1.0.0, apache is the httpd user. Used the instructions from Asipto's site for Siremis install and configure, added the alias to httpd.conf, Config.xml is: <?xml version="1.0" standalone="no"?> <Config> <DataSource> <Database Name="Default" Driver="Pdo_Mysql" Server="127.0.0.1" DBName="openser" User="openser" Password="*****" /> <Database Name="Siremis" Driver="Pdo_Mysql" Server="127.0.0.1" DBName="openser" User="openser" Password="*****" /> </DataSource> </Config> I navigate to the Siremis page, http://10.5.10.21/siremis/ and get the login screen, enter the login credentials, Submit, and nothing happens. httpd/error_log: [Wed May 26 12:49:45 2010] [error] [client 10.3.226.168] File does not exist: /var/www/html/siremis-1.0.0/siremis-web/js/sound.js, referer: http://10.5.10.21/siremis/bin/ [Wed May 26 12:49:45 2010] [error] [client 10.3.226.168] File does not exist: /var/www/html/siremis-1.0.0/siremis-web/js/sound.js, referer: http://10.5.10.21/siremis/bin/ I have wiped Siremis and done a fresh install, restarted httpd, and still nothing. Does anyone have ideas or suggestions on where else to look? -- ================================================================== Joe Micciche jmiccich(a)redhat.com Red Hat, Inc. http://www.redhat.com Senior Communications Engineer X(81) 44554 +1.919.754.4554 ==================================================================

6 15

t_check_trans return false on CANCEL
by Ernest Mavrel 27 May '10

27 May '10

3 2

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/

4 5

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

3 4

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.

2 1

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

2 2

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

2 1

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//

4 3

'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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sr-users May 2010