opensips & Freeswitch calling issue
Hi, I have follow below steps to implement load balancing of FS servers using opensips. http://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeSWI... Here opensips is forwarding registration request to FS servers. But I need users to register only in opensips and when they do calls, FS should work as media server. So in simple, opensips should work as user registration server and Freeswitch should work as media server. I did some modifications in opensips.cfg to do registration in opensips only. Here is the modifications, Removed below code from script if (!add_path_received()) { xlog("L_ERR", "$ci|log|unable to add path"); sl_send_reply("503", "Internal path befuddlement"); #remove the association between the call-id and the media server (if one) #but leave the contact user and server to support transfers cache_remove("local", "$ci"); xlog("L_INFO", "$ci|end|cleaned up call id from cache"); exit; } Added code in place of above code if (!zz_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if (!db_check_to()) { sl_send_reply("403","Forbidden auth ID"); exit; } if ( proto==TCP || 0 ) setflag(7); if (!save("location")) sl_reply_error(); exit; Using above code registration is working fine. But now the problem is with DID number calls. I have two extensions registered in opensips 12345 and 123456. I have DID number 0123456789 in FS.
From Freeswitch I have pointed DID to 123456 extension using sofia/profile/123456@192.168.1.10.
Here, 192.168.1.10 is opensips ip 192.168.1.31 is fs ip But when i am calling that DID number, I am getting error in opensips like, http://pastebin.com/AhuDP6hC Freeswitch is also showing error like "SIP/2.0 503 Maximum Calls In Progress" I am trying trial & error and searching for solution but not getting anything. Please guys help me to fix the issue. Thanks in Advance. Thanks, Sam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/opensips-Freeswitch-calling-is... Sent from the freeswitch-users mailing list archive at Nabble.com.
Hi, you should take a look with wireshark on the sip flow. It seems to be an endless loop. Your selfmade opensips xlog error messages didnt shows up whats going on. 2013/5/17 smrdoshi <smrdoshi@gmail.com>
Hi,
I have follow below steps to implement load balancing of FS servers using opensips.
http://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeSWI...
Here opensips is forwarding registration request to FS servers. But I need users to register only in opensips and when they do calls, FS should work as media server. So in simple, opensips should work as user registration server and Freeswitch should work as media server.
I did some modifications in opensips.cfg to do registration in opensips only. Here is the modifications,
Removed below code from script if (!add_path_received()) { xlog("L_ERR", "$ci|log|unable to add path"); sl_send_reply("503", "Internal path befuddlement");
#remove the association between the call-id and the media server (if one) #but leave the contact user and server to support transfers cache_remove("local", "$ci"); xlog("L_INFO", "$ci|end|cleaned up call id from cache"); exit; } Added code in place of above code if (!zz_authorize("", "subscriber")) { www_challenge("", "0"); exit; }
if (!db_check_to()) { sl_send_reply("403","Forbidden auth ID"); exit; }
if ( proto==TCP || 0 ) setflag(7);
if (!save("location")) sl_reply_error();
exit;
Using above code registration is working fine.
But now the problem is with DID number calls. I have two extensions registered in opensips 12345 and 123456. I have DID number 0123456789 in FS.
From Freeswitch I have pointed DID to 123456 extension using sofia/profile/123456@192.168.1.10.
Here, 192.168.1.10 is opensips ip 192.168.1.31 is fs ip
But when i am calling that DID number, I am getting error in opensips like, http://pastebin.com/AhuDP6hC Freeswitch is also showing error like "SIP/2.0 503 Maximum Calls In Progress"
I am trying trial & error and searching for solution but not getting anything.
Please guys help me to fix the issue. Thanks in Advance.
Thanks, Sam
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/opensips-Freeswitch-calling-is... Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Mit freundlichen Grüßen *Karsten Horsmann*
Hi Karsten, Thanks for the reply. I took tcpdump of FS sip log. Its showing 408 Request timeout error now. I am not sure where's the problem? I am attaching tcpdump in case if you get idea about the issue. tcpdump.pcap <http://freeswitch-users.2379917.n2.nabble.com/file/n7590807/tcpdump.pcap> Please let me know. Thanks again for your help. Thanks, Sam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/opensips-Freeswitch-calling-is... Sent from the freeswitch-users mailing list archive at Nabble.com.
Hi, sorry but i only see the Invites and the register Messages and after then FS returns 408 cause. Maybe you should take an working example of opensips + freeswitch as blueprint for your. < http://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegratio...
2013/5/17 smrdoshi <smrdoshi@gmail.com>
Hi Karsten,
Thanks for the reply.
I took tcpdump of FS sip log. Its showing 408 Request timeout error now. I am not sure where's the problem?
I am attaching tcpdump in case if you get idea about the issue. tcpdump.pcap <http://freeswitch-users.2379917.n2.nabble.com/file/n7590807/tcpdump.pcap> Please let me know.
Thanks again for your help.
Thanks, Sam
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/opensips-Freeswitch-calling-is... Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Mit freundlichen Grüßen *Karsten Horsmann*
Hi Karsten, Thanks for the link. I got the issue. I missed the code to make feature working. I search and kept it. Now its working fine. Thanks, Samir Doshi -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/opensips-Freeswitch-calling-is... Sent from the freeswitch-users mailing list archive at Nabble.com.
participants (2)
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Karsten Horsmann
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smrdoshi