[OpenSIPS-Users] Tutorials and Help

[Bogdan-Andrei Iancu]

Hi GuruTux ;)


CAW wrote:
> Greetings-
>
> I am new to OpenSIPS and am looking for some additional help/resource 
> materials to go along with the opensips.org <http://opensips.org> 
> documentation.
>
> I have currently set up an OpenSIPS 1.5.0 box using a mySQL backend. 
> I've been able to register three (3) devices to it: X-Lite Softphone, 
> Linksys SPA-942, & a SIP trunk off of my Asterisk 1.4.23 server.
>
> I'll go ahead and explain a few of my issues so you will have a better 
> understanding of my level of understanding and how much help I'm 
> really needing.
>
> With some routing magic off of the Ast box, I am able to call to 
> either one of the other devices and, I believe, connect the calls. 
> However, RTP does not seem to be passing as I get no audio. Could this 
> be an RTP issue, codec issue, or just a result of registering a phone 
> directly to the OpenSIPS?
OpenSIPS is doing just signalling, it does not get into the media path, 
so RTP is bypassing it. If there is not NAT involved, it should not be 
any issues with the RTP flow.
>
> I've not quite got my head around how number routing is supposed to 
> work. I'm currently trying to use the LCR, as that's going to be an 
> important piece in the near future, but I don't know that I'm going 
> about it right.
>
> lcr routes
> +----+--------+----------+--------+----------+
> | id | prefix | from_uri | grp_id | priority |
> +----+--------+----------+--------+----------+
> |  1 | 555    |          |      1 |        1 |
> +----+--------+----------+--------+----------+
> lcr gateways
> +---------------+----------------+------+------------+-----------+--------+-------+------+-------+
> | gw_name       | ip_addr        | port | uri_scheme | transport | 
> grp_id | strip | tag  | flags |
> +---------------+----------------+------+------------+-----------+--------+-------+------+-------+
> | AstSvr | a.b.c.d | 5060 |          1 |         3 |      1 |     0 
> |      |     0 |
> +---------------+----------------+------+------------+-----------+--------+-------+------+-------+
>
> If I'm understanding this correctly, anything I dial from one of the 
> devices should go out the gateway to my AstSvr unless the number 
> begins with 555. Correct?
not really - you need a default rule (with no prefix) to match all other 
destinations.
> And anything I dial that begins with 555 should be routed out group_id 
> 1 (in this case is also my AstSvr). 
this is correct.
> Unfortunately, any number I dial, whether it begins with 555 or not, I 
> cannot complete the call. And to answer the question before it is 
> asked, it didn't work with only the gateway and no second route either.
do you get any errors (script or sip level) ? you need to look for what 
is the reason for call failing. You can try using full debug mode 
(debug=6 in your cfg) to get more info about what is going on.
>
> Another small item I'm running into is that I receive the following in 
> my error log whenever I do anything with the lcr command:
>
> ERROR:mi_fifo:mi_fifo_server: command lcr_reload is not available
the command is made available by the lcr module, so, even if it sounds 
as a stupid question, do you load the lcr module?
>
> I've not been able to find any information about this error either on 
> the opensips.org <http://opensips.org> site or through Google.
>
> Any help you might be able to give would be greatly appreciated.

My personal advice is not to invest time in the LCR module, but to start 
using Drouting module - Drouting will replace LCR (which will be 
obsoleted) in the next release. Drouting module has some really nice 
add-on and better performance.

Sooner, a DR tutorial will be available.

Regards,
Bogdan