Re: [SR-Users] dialplan question

Hi, Below in red…. BR, Uri ------------------------------------------------------------------------ *From:*Daniel-Constantin Mierla [mailto:daniel@asipto.com] *Sent:* Thursday, August 25, 2011 1:56 PM *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List *Cc:* אורי שקד- חטיבת הנדסה ורשת- Uri Shacked *Subject:* Re: dialplan question (was: [SR-Users] ACC with MySqsl problem after installing Siremis) Hello, if you want to discuss about another topic, it is better to start a new email thread with subject reflecting the new topic -- it will be noticed properly, since various components are more familiar to different people.– Thanks, I will next time. On 8/22/11 1:34 PM, אורי שקד- חטיבת הנדסה ורשת- Uri Shacked wrote: Thanks! [...] I am trying to use the Dialplan module in order to translate numbers. For example: When a call arrives and designated to sip:999@localhost (there is no such subscriber) the dialplan should find it (it does) and replace the 999 to 102. My problem is that it does not translate the ruri and does not initiate the call to sip:102@localhost (it is a subscriber). Any ideas? The log is attached, see the line – “registrar [lookup.c:84]: '999' Not found in usrloc” Why is it looking for 999? It was supposed to replace it with 102 no? Seems you don't call dp_translate() to update the r-uri user part, can you paste here the piece of config you use for doing the dialplan translation? The cfg file is attached (line 474 is the dp_translate). I use database configuration, in the Kamailio.log I definitely see it recognize the “999” and var to change is “102” but as I mentioned, it doesn’t change the ruri and search for 999 in the location table…

Thanks a lot! I have to say I have been working on Kamailio for 10 days now. It seems hard to understand..(maybe for me J) But it has a feeling of a very powerful and flexible Sip server… which is very very good! Indeed, it may take a bit to get into it, but once you are there, you will discover how easy is to implement amazing ideas or fix broken signaling sent by devices. Cheers, Daniel BR, Uri ------------------------------------------------------------------------ *From:*Elena-Ramona Modroiu [mailto:ramona@asipto.com] *Sent:* Monday, August 22, 2011 11:57 AM *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List *Cc:* אורי שקד- חטיבת הנדסה ורשת- Uri Shacked *Subject:* Re: [SR-Users] ACC with MySqsl problem after installing Siremis Hi, if you installed Siremis 2.0 via the wizard, the structures of acc and missed_calls tables were updated, so you have to change the parameters of acc module in kamailio.cfg, like: http://kb.asipto.com/siremis:install20:accounting#config_file If you want to see the sql used to create the new acc and missed_calls tables, see: http://siremis.git.sourceforge.net/git/gitweb.cgi?p=siremis/siremis;a=blob;… Regards, Ramona On 8/21/11 10:58 AM, אורי שקד- חטיבת הנדסה ורשת- Uri Shacked wrote: Hi, I managed to setup the ACC module with MySql. Now, after installing Siremis, the ACC table and the Missed_calls table are empty and do not populae the data from the call I make. Any ideas why? BR, Uri