<div dir="ltr"><div><div>Hi,<br><br></div>you should take a look with wireshark on the sip flow. <br>It seems to be an endless loop.<br><br></div>Your selfmade opensips xlog error messages didnt shows up whats going on.<br>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/5/17 smrdoshi <span dir="ltr"><<a href="mailto:smrdoshi@gmail.com" target="_blank">smrdoshi@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi,<br>
<br>
I have follow below steps to implement load balancing of FS servers using<br>
opensips.<br>
<a href="http://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeSWITCH_installs" target="_blank">http://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeSWITCH_installs</a><br>
<br>
Here opensips is forwarding registration request to FS servers. But I need<br>
users to register only in opensips and when they do calls, FS should work as<br>
media server. So in simple, opensips should work as user registration server<br>
and Freeswitch should work as media server.<br>
<br>
I did some modifications in opensips.cfg to do registration in opensips<br>
only. Here is the modifications,<br>
<br>
Removed below code from script<br>
if (!add_path_received())<br>
{<br>
xlog("L_ERR", "$ci|log|unable to add path");<br>
sl_send_reply("503", "Internal path befuddlement");<br>
<br>
#remove the association between the call-id and the media server<br>
(if one)<br>
#but leave the contact user and server to support transfers<br>
cache_remove("local", "$ci");<br>
xlog("L_INFO", "$ci|end|cleaned up call id from cache");<br>
exit;<br>
}<br>
Added code in place of above code<br>
if (!zz_authorize("", "subscriber"))<br>
{<br>
www_challenge("", "0");<br>
exit;<br>
}<br>
<br>
if (!db_check_to())<br>
{<br>
sl_send_reply("403","Forbidden auth ID");<br>
exit;<br>
}<br>
<br>
if ( proto==TCP || 0 ) setflag(7);<br>
<br>
if (!save("location"))<br>
sl_reply_error();<br>
<br>
exit;<br>
<br>
Using above code registration is working fine.<br>
<br>
But now the problem is with DID number calls. I have two extensions<br>
registered in opensips 12345 and 123456. I have DID number 0123456789 in FS.<br>
>From Freeswitch I have pointed DID to 123456 extension using<br>
sofia/profile/<a href="mailto:123456@192.168.1.10">123456@192.168.1.10</a>.<br>
<br>
Here,<br>
192.168.1.10 is opensips ip<br>
192.168.1.31 is fs ip<br>
<br>
But when i am calling that DID number, I am getting error in opensips like,<br>
<a href="http://pastebin.com/AhuDP6hC" target="_blank">http://pastebin.com/AhuDP6hC</a><br>
Freeswitch is also showing error like "SIP/2.0 503 Maximum Calls In<br>
Progress"<br>
<br>
I am trying trial & error and searching for solution but not getting<br>
anything.<br>
<br>
Please guys help me to fix the issue.<br>
Thanks in Advance.<br>
<br>
Thanks,<br>
Sam<br>
<br>
<br>
<br>
--<br>
View this message in context: <a href="http://freeswitch-users.2379917.n2.nabble.com/opensips-Freeswitch-calling-issue-tp7590795.html" target="_blank">http://freeswitch-users.2379917.n2.nabble.com/opensips-Freeswitch-calling-issue-tp7590795.html</a><br>
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</blockquote></div><br><br clear="all"><br>-- <br>Mit freundlichen Grüßen<br>*Karsten Horsmann*<br>
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