sr-users April 2008

sr-users@lists.kamailio.org

141 participants
187 discussions

by Bogdan-Andrei Iancu

Hi Norberto, most sure the problem is in routing the CANCEL request. On this topic, please see: http://www.voice-sistem.ro/downloads/2007.08.29-Admin-Course/ There is detailed explanation about CANCEL routing. Regards, Bogdan Norberto Monteiro wrote: > > Hello, > > I’m having a problem that is the following: > > I’m using rewriteport to send traffic to another proxy. Then When I > make a call and if I hung up the other phone still rings… > > What could be the problem? > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

16 years, 1 month

[OpenSER-Users] Openser and Portabilling100

by Jaas

Hi all, I would like to know if there could be same problems with OpenSER and PortaBilling100 because I have to choose a billing system for the ISP I work. I don't see any document with interaction between these two systems... Alternatively, can you give me hints about a good and complete billing system? Thank you in advance

16 years, 1 month

[OpenSER-Users] FreeSWITCH and OpenSER

by Mikael A. Bjerkeland

Hi, I am trying to look up information on the use of FreeSWITCH together with OpenSER. I am doing a configuration on my own without much experience on the configuration of OpenSER. In my setup OpensSER acts as a registrar and proxy. I have multiple FreeSWITCH servers for media, routing and PSTN termination, but I am not sure if my approach is the "right way". All SIP traffic goes to the OpenSER proxy which in turn routes the call with the LCR module to a FreeSWITCH server. This has proven to work fine in my limited testing environment. I am using ENUM on the FreeSWITCH server to look up dialed numbers. If the dialed number was found in our ENUM registry and belongs to the proxy, we call back to OpenSER: SIP UA (subscriber 56195122) -> INVITE 25929511 -> FreeSWITCH -> ENUM lookup -> Found -> Routed back to OpenSER -> SIP UA (subscriber (25929511) If the call is not found: SIP UA (subscriber 56195122) -> INVITE 95018615 -> FreeSWITCH -> ENUM lookup -> Not found -> Routed to PSTN gateway or SIP peer I am not using aliases in OpenSER as it seems that ENUM provides the same functionality, but please correct me if I'm wrong. I also made this decision because it seems easier to implement features like call forwarding (unconditional, busy, no answer and timed) in FreeSWITCH rather than in OpenSER. This brings me back to my current issue: Call transfer Outbound and inbound calling with ENUM lookups in FreeSWITCH to my test phones works fine. So does calls to the PSTN gateway. My problem has to do with transfers. I can do a an attended transfer from my Polycom phone when I'm in a call. The second call is set up and I can talk to the person I want to transfer to. I hit the transfer button. Nothing happens, and FS and OpenSER try to look up number_to_transfer_to@openser-registrar. The call is routed to OpenSER which complains that it cannot find the subscriber/alias, which makes sense as there isn't configured an alias. But this also happens when I transfer to a PSTN number. Do I need to take special actions on REFERs? Are there any examples of a working OpenSER + FreeSWITCH setup out there? I believe FreeSWITCH and OpenSER complement each other great but I found the lack of information on the two applications used together a showstopper. Has anyone implemented phone service based on OpenSER and FreeSWITCH who can explain their setup a bit? Regards, Mikael

16 years, 1 month

[OpenSER-Users] Question about installing openser

by Pete Kay

Hi all, I am trying to get Openser 1.3.1 installed on Debian. It gave me a lot of errors when I try to "make" from the source. Could someone please tell me the list of packages I should install before building Openser? By the way, I haven't been able to find a step-by-step guides on installing Openser from scratch on Debian. If there is one exists somewhere on the Internet, I would greatly appreciate it if you can point me to it. I am currently working through a few of the how-tos trying to put pieces together. Thanks in advance for your help. Regards, Pete

16 years, 1 month

[OpenSER-Users] Problem with redirection (302 Moved temporarly)

by Antonio Reale

Hi all, I have a Linksys SPA2102 with 2 phones configured with the same account/number. If I call that number both phones ring and all is OK. The problem is that if I set a call forwarding on one of the two line on the SPA only one phone rings. Openser receives the 302 message from the SPA but the failure route that contains t_check_status("302") is not called. The forwarding works if only one line is connected. How can I handle the 302 in this situation? Thanks in advance. Regards, Antonio.

16 years, 1 month

[OpenSER-Users] sipwise config error - save location?

by Robert McNaught

Please discard - fix for this was a curly bracket on line 313 - not sure if that was sipwise or me! Robert

16 years, 1 month

[OpenSER-Users] sipwise config error - save location?

by Robert McNaught

Hi, Has anyone ever seen this error before? I created a config file using the sipwise utility: [root@ser1 openser]# openser -c 0(30469) parse error (314,1-3): syntax error 0(30469) parse error (314,1-3): ERROR: bad config file (2 errors) 0(30469) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy [root@ser1 openser]# 314 if(!save("location")) 315 { 316 317 xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); 318 sl_reply_error(); 319 exit; 320 } It complains about line 314 having a syntax error - I have another openser machine running with the same exact line and it is ok? Has anyone seen this before? TIA Robert

16 years, 1 month

[OpenSER-Users] SIP Load Balanser for OpenSER

by Vladimir Romanov

Hi! I want test setup OpenSER in HA & performance cluster. For this I need SIP LoadBalancer. Anyone known were I can download vovida lbproxy? Vovida site not answer last time :( Maybe there exist other software SIP load balancers. -- Vladimir Romanov

16 years, 1 month

[OpenSER-Users] Avoid Loops between Openser and Asterisk

by Cosimo Fadda

Hi everybody, I'm using Opeser as Proxy and one Asterisk as Gateway and Voicemail server. Suppose that an incoming calls arrives to Ser from Asterisk and the user is offline; Then Openser sends back the Invite to Asterisk that should activate the voicemail application but unfortunately it detects a Loop (482) and rejects the invite. I think I have to mangle the SIP message in Openser before the send to Asterisk again. This is what I currently do: if(!lookup("location")) { #send to voicemail if active but not registerd xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if($avp(s:vmail) == "1") { route(11); } else { route(20); } } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(9); } exit; .... route[11] { xlog("L_INFO", "Forwarding request to VM\n"); prefix("vm"); sethostport("asterisk-gw:5060"); #append_branch(); if (!t_relay()) { xlog("L_INFO", "Unable TO Forward the request to VM\n"); route(20); exit; } What can I do to avoid Loop Detection? Thanks in advance, Cosimo Fadda ___________

16 years, 1 month

[Serusers] Asterisk with SER ...great solution

by satish patel

find solution on my website http://linuxbug.org/index_files/projects.html $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Get the freedom to save as many mails as you wish. Click here to know how.

16 years, 1 month

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sr-users April 2008