December 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Zombie Records in core db
by DJB 08 Dec '09

08 Dec '09

We have FreeSWITCH Version 1.0.4 (exported) running at a high volume traffic. I normally check the concurrent calls by looking at the number of sessions from status command. However, the number of concurrent calls in FS is normally higher than it's supposed to be after we ran traffic for about a week. Thus, I routed the traffic away from the FS and found out from "show calls" that there were so many old calls from previous days. We are running a pass-thru traffic in signaling only. I wonder whether there is a way to have those "zombied" records clean up automatically. Also, what should I do to prevent this problem? Thank you, Dorn B.

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Trapping dtmf on bridged call
by Nik Middleton 08 Dec '09

08 Dec '09

Hi Is it possible to trap on DTMF on a bridged call within an LUA script? I've tried setting the gateway to use inband, but no joy. It looks like I could use start_dtmf, but I can't see how to launch this within LUA Regards,

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No NOTIFY MWI when registering via proxy.
by mayamatakeshi 07 Dec '09

07 Dec '09

Hello, I'm testing FS support for the header Path (FS is behind opensips). It pretty much works: I tested calling from one user to the other and calls work perfectly. However, I've noticed that when I register my terminal directly with FS without going thru the proxy, I receive an unsolicited NOTIFY containing Message-Waiting information. But when I register via proxy, FS doesn't send this NOTIFY. What could be causing this difference of behavior? (enabling debug (F8) doesn't show anything for registration handling). I have just updated to trunk Revision 14729 and this behavior persists. Also, is it possible to disable this unsolicited MWI notification? (I want to send it only to subscribed terminals). regards, takeshi

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Database suggestions/pointers/?
by Steve Klein 07 Dec '09

07 Dec '09

Greetings. We need to add database access to an IVR application we are prototyping. Based on FS "best practice" suggestions, we are using Lua for the scripts. Since FS uses SQLite internally, we presumed that Lua + SQLite would be a recommended approach. However, we can't find any examples of this combo anywhere. So, what is the "best practice" scripting + database recommendation for a high-volume database-driven FS app? Any help/pointers greatly appreciated! --Steve Klein

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Access to users variables
by Mark Campbell-Smith 07 Dec '09

07 Dec '09

Hi! How can I access the variables that are defined in a users xml file? For example, say user 1000 has a variable called smsnumber, as defined below: <include> <user id="1000" mailbox="1000"> <params> <param name="password" value="1000"/> </params> <variables> <variable name="smsnumber" value="12345"/> </variables> </user> </include> How can I use variable ${smsnumber} in a dialplan to run a perl script using <action application="system" data="sms.pl"/> ? Thanks

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Sporadic call drops
by Luis F Urrea 07 Dec '09

07 Dec '09

Hi all, Guys I know the question could be too vague, but I have a customer that just reported frequent failure to place outbound calls though a PSTN gateway on the LAN. I looked at the logs and I seem to be able to confirm that FS fails to place the call through the gateway and that the issue resides on the FS side since the first channel that s killed is tht of the internal extension registered to FS and then FS send the BYE to gw and kills the channel. What are possible causes of this? I know you always like to look at complete logs but here's a snip that could shed some light on the disconnection. (I can provide full logs if required and worthed) 2009-12-04 11:21:56 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/internal/200(a)172.16.3.5 entering state [ready][200] 2009-12-04 11:21:59 [DEBUG] sofia.c:3037 sofia_handle_sip_i_state() Channel sofia/internal/200(a)172.16.3.5 entering state [terminated][200] 2009-12-04 11:21:59 [NOTICE] sofia.c:3597 sofia_handle_sip_i_state() Hangup sofia/internal/200(a)172.16.3.5 [CS_EXECUTE] [NORMAL_CLEARING] 2009-12-04 11:21:59 [DEBUG] switch_channel.c:1660 switch_channel_perform_hangup() Send signal sofia/internal/200(a)172.16.3.5[KILL] 2009-12-04 11:21:59 [DEBUG] switch_core_session.c:933 switch_core_session_signal_state_change() Send signal sofia/internal/ 200(a)172.16.3.5 [BREAK] 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:371 audio_bridge_thread() sofia/internal/200(a)172.16.3.5 ending bridge by request from write function 2009-12-04 11:22:00 [DEBUG] switch_ivr_bridge.c:426 audio_bridge_thread() sofia/pstn/22909980(a)172.16.3.46 receive message [UNBRIDGE] Is the 6th line normal behavior for ending the channel? FreeSWITCH Version 1.0.trunk (13484M) TIA

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Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
by Jerry Richards 07 Dec '09

07 Dec '09

I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _____ From: Michael Jerris [mailto:mike@jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org <http://pastebin.freeswitch.org/> On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards <jerry.richards(a)teotech.com> wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing <enable-3pcc> to "true" and also "proxy", but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org <http://www.freeswitch.org/> -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net/> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Need Conference design help
by shehzad p 07 Dec '09

07 Dec '09

Hello Every one, I have to design conference, and I need community guidance to efficiently accomplish that. I need to create Conference which will have three kind of users: 1. Moderator (may be only one per conference) 2. User who can participate in conference without moderator interaction. 3. User who can only participate when Moderator allow them to get in. Also besides above setup I have to perform other things like Record the conference, Multicast the conference to other freeswitch server. I saw the conference Record CLI command but wondering where to setup when conference starts. I am also wondering how Multicast Conference is possible in Freeswitch and how the receiver Freeswitch configuration will look like. Thanks. msp -- View this message in context: http://old.nabble.com/Need-Conference-design-help-tp26653473p26653473.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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att_xfer origination_cancel not working and b has no chance to talk with c in an A-B-C scenario
by Seven Du 07 Dec '09

07 Dec '09

Hi, I know there's some chang on att_xfer, and after upgrade(re-bootstrap) to trunk code, no sound after att_xfer. Then I rebuild FS 15807 with a fresh checkout, but still using the old conf/ settings, sound is ok, but there are other problems: A call B, and B att_xfer C 1) origination_cancel_key not working. no even no DTMF log in FS when I press # or any other key, I tried with Zoiper and Snom(on the B leg) 2) when C answers, B immediately hangup, so B has no chance talk to C Could this be a problem? I pasted logs: http://pastebin.freeswitch.org/11417 Thanks.

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Audiocodes PRI Gateway
by jay binks 07 Dec '09

07 Dec '09

Guys, im after info from people with experience with AudioCodes Mediant 2k PRI Gateways. specifically how well they inter-op with Freeswitch, and how compliant their SIP stack is. I guess the bottom line is, would you recommend these gateways or would you suggest something else ? -- Sincerely Jay

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