Look at the dialplan module for matching a prefix and maybe use the attr field to set a trunk ID, which you can useHi Olle!Yes, the Carrier Route is complex right, my problem was on routing Logic where I've replaced standard route(PSTN) with the lines I found into the Carrier Route module documentation.# dispatch destinations to PSTN#route(PSTN);# route calls based on hash over callid# choose route domain default of the default carrierif(!cr_route("default", "default", "$rU", "$rU", "call_id")){sl_send_reply("403", "Not allowed");} else {# In case of failure, re-route the requestt_on_failure("1");# Relay the request to the gatewayt_relay();}But actually, I never thinked about the alternatives, maybe there is a better/faster way to deal with my requirements.As internal (registered users) calls are already working It's only missing to place outbound calls to different IPs depending on the number dialled and sometimes I need to apply a prefix before the called party number.
_______________________________________________I'm not requiring any billing/rating process as one of the platform to which I will send calls will bill calls based on CLI/IP auth.Thank you so much for your hard work!CheersMaxIl giorno 01/dic/2014, alle ore 15:37, Olle E. Johansson ha scritto:
On 01 Dec 2014, at 15:20, Massimo Varriale (IPZeta) <m.varriale@ipzeta.it> wrote:As I'm a new Kamailio user I agree that Kamailio could be difficult at the beginning, but I totally do not agree to the fact that reading documentation is a waste of time!What I found online is that it's true that there are so many tutorials out there, but sometimes I found that they refer to old version of OpenSER and sometimes syntax is little different, but as everything in the IT world there is nothing anyone will install/configure and it will work out of the box, at least you have to study the architecture and learn how the software will going to act. Maybe that the software works out of the box will use a standard behaviour, and no special needs.. You talked about Asterisk, but for me it's so easy to install a FreePBX, open the webpage, make some basic configuration and that's it..In my past experience in IT I can say that I really hate the "5 minutes man" that suppose everything is up and running in 5 minutes, everything is so easy that any monkey could do that..Thank you for your feedback. I have been working hard trying to make the documentation files we do have easier to read and understand.BTW, I would love to see some updated real world tutorials on Kamailio as in my case it's 15 days I'm fighting with no success with Carrier Route module and I'm not able to send outbound calls....in my case, if I was on Freeswitch those calls were already sent out..but this is another story.The carrier route module is a very large module that includes a lot of business logic. It's complex. Is it really the one you need or could you try another module that implements least cost routing? We do have several.
/OThanksMaxIl giorno 01/dic/2014, alle ore 15:08, Fred Posner ha scritto:On 12/01/2014 08:35 AM, Aria Mill wrote:Hello,Kamailio is difficult. Many of us want to use it because is open sourceand it's flexible. but to tell you the truth after 3-4 hours of playingwith it I am frustrated, and I am starting to hate it!I hate to hear that you are starting to hate the software... many of us truly love it. I find one correction to your main statement.SIP is difficult. Kamailio is flexible. The more you understand and know SIP, the easier using Kamailio becomes.Why has nobody made a kamailio video series on youtube?... snip ...This is 2014, people don’t waste time readingdocumentation, youtube is much more efficient...... snip ....I for one have almost zero interest in making videos of configuring software. Plus, I have a face for radio. ;)Some of us make tutorials... I have made two of them, and they get a lot of readers; so I do think it's not a waste of time.If you're good at making videos, this would be a great way for you to contribute to the software.The vast majority of asterisk videos I see aren't made by Digium or the Asterisk devs... but instead by either hardware companies selling asterisk based hardware, enthusiasts, or support vendors.Again, this might be a great way for you to contribute.Fred PosnerThe Palner Group, Inc.http://www.palner.com (web)+1-503-914-0999 (direct)+1-954-472-2896 (fax)_______________________________________________SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users_______________________________________________SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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