Greetings-<br><br>I am new to OpenSIPS and am looking for some additional help/resource materials to go along with the <a href="http://opensips.org">opensips.org</a> documentation.<br><br>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.<br>
<br>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.<br><br>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?<br>
<br>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.<br>
<br>lcr routes<br>+----+--------+----------+--------+----------+<br>| id | prefix | from_uri | grp_id | priority |<br>+----+--------+----------+--------+----------+<br>| 1 | 555 | | 1 | 1 |<br>+----+--------+----------+--------+----------+<br>
lcr gateways<br>+---------------+----------------+------+------------+-----------+--------+-------+------+-------+<br>| gw_name | ip_addr | port | uri_scheme | transport | grp_id | strip | tag | flags |<br>+---------------+----------------+------+------------+-----------+--------+-------+------+-------+<br>
| AstSvr | a.b.c.d | 5060 | 1 | 3 | 1 | 0 | | 0 |<br>+---------------+----------------+------+------------+-----------+--------+-------+------+-------+<br><br>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? And anything I dial that begins with 555 should be routed out group_id 1 (in this case is also my AstSvr). 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.<br>
<br>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:<br><br>ERROR:mi_fifo:mi_fifo_server: command lcr_reload is not available<br><br>I've not been able to find any information about this error either on the <a href="http://opensips.org">opensips.org</a> site or through Google.<br>
<br>Any help you might be able to give would be greatly appreciated.<br><br>tia,<br><br>CAW<br>