June 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP Trace
by Eric Beard 28 Jun '11

28 Jun '11

Hello, In fs_cli, I can issue this command: sofia global siptrace on And I see all SIP traffic. But the SIP does not get logged to the actual log file. How do I get the same output to be sent to the log? I've searched through the Wiki but I'm not seeing a clear way to do it. Thanks! ----------------------- Eric Z. Beard, CTO Loop LLC w (877) 850-2010 x9249 m (727) 776-2768 eric(a)loopfx.com<mailto:eric@loopfx.com>

8 14

getting rid of hanging sessions?
by Wes 27 Jun '11

27 Jun '11

in my testing, I'm getting some sessions that are, hung, I guess. Status shows: UP 0 years, 0 days, 2 hours, 7 minutes, 37 seconds, 752 milliseconds, 720 microseconds 23 session(s) since startup 5 session(s) 0/30 1000 session(s) max min idle cpu 0.00/0.00 if I do a shutdown, it waits on them for a while, then eventually shuts down. Is there any way to "clean" these up periodically? Or, what am I doing to leave them out there? thanks!

3 4

How to deal with user pressing extra keys? flush_dtmf? session::flushDigits?
by Wes 27 Jun '11

27 Jun '11

I'm writing an ivr script in lua, and using playAndGetDigits in a couple different places. Both places just take a single digit to continue on. If the user presses 2 digits, it seems as if the second digit is being queued and it's being processed in the next call to playAndGetDigits. Often the extra digit is invalid, so when it tells the user to press a certain key, it follows immediately with the message "invalid key", due to the extra key being in the "queue"... is this anything I can protect against? I found a flush_dtmf, but it is not very documented: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_flush_dtmf and I also found this: http://wiki.freeswitch.org/wiki/Session_flushDigits but calling it before and after playAndGetDigits didn't *seem* to help... should it? script here: http://pastebin.freeswitch.org/16584

3 4

second member to dynamic conference hears only hold music
by elijah 27 Jun '11

27 Jun '11

This setup was working for me until updating against git yesterday. Via a bind statement on the b-leg of inbound calls, a user is able to pull both legs into a conference bridge. The problem now is that the a-leg hears only hold music. I'm having this and another problem with my dialplan after updating, w/o changing my own configuration - could there be updates to source that would be affecting me? Here's the setup: <extension name="telifi_sales_queue"> <condition field="destination_number" expression="^telifi_sales_queue$"> <action application="bridge_export" data="telifi_call_id=${telifi_call_id}"/> <action application="set" data="bind_meta_key=#"/> <action application="bind_meta_app" data="1 b s execute_extension::dx1 XML callsdirect"/> <action application="bind_meta_app" data="2 b s execute_extension::dx2 XML callsdirect"/> <action application="callcenter" data="${ telifi_queue_name}(a)live001.voice.telifi.com"/> </condition> </extension> <extension name="bind_receiver_2"> <condition field="destination_number" expression="^dx2$"> <action application="answer"/> <action application="transfer" data="-both dynamic_conference XML callsdirect"/> </condition> </extension> <extension name="dynamic_conference"> <condition field="destination_number" expression="^dynamic_conference"/> <condition field="rdnis" expression="^(dx2)$"> <action application="answer"/> <action application="bind_digit_action" data="moderator,*1,exec:execute_extension,ASK_FOR_NUMBER__${telifi_call_id} XML callsdirect"/> <action application="bind_digit_action" data="moderator,*2,exec:execute_extension,CANCEL_LAST_CALL__${telifi_call_id} XML callsdirect"/> <action application="digit_action_set_realm" data="moderator"/> <action application="conference" data="${telifi_call_id}@simple"/> <anti-action application="answer"/> <anti-action application="conference" data="${telifi_call_id}@simple "/> </condition> </extension>

3 5

Please let's protect the wiki from spammers
by Diego Viola 27 Jun '11

27 Jun '11

Hi, As most of us may alright know, the FreeSWITCH wiki has been abused and attacked by spammers, as seen here: http://wiki.freeswitch.org/wiki/Special:RecentChanges Some FreeSWITCH users have been deleting and blocking these spammers, but they are increasing their spamming attacks. I went into #mediawiki in freenode and asked them if there is any solution to this and this is what they suggest: 15:18 < bawolff> !spam | diegoviola 15:18 < mw-bot> diegoviola: For information about combating and handling spam in MediaWiki, see <http://www.mediawiki.org/wiki/Manual:Combating_spam> and <http://www.mediawiki.org/wiki/Anti-spam_features>. I've seen most wikis who have implemented this solution and they no longer get any spamming after applying said solution. If somebody here have administration access to http://wiki.freeswitch.org/ I offer my help to apply this solution. Thanks, Diego Viola

3 2

Removing SDP from 183/180
by Artur Cichocki 26 Jun '11

26 Jun '11

Hi All. I didn't get any response from fs-dev moderator for one month so it goes here. I just wrote this patch. It ads 'sip_remove_183sdp' flag, which removes SDP from 183 and returns 180 (it also removes SDP from 180, because FS changes 180 with SDP to 183). To turn it on add {sip_remove_183sdp=true} to the dial string. Why? Because in some situations we don't want to transfer voice between clients before a real call establishment (and yes, I know that early media in this stage is allowed by RFC). Real case: Try to setup two clients, one of them using Sipdroid. Turn on bypass mode. Try to call to Sipdroid client, you will hear caller voice together with the ring (because Sipdroid sends 180 with SDP, and puts voice on the speaker). Patch attached. -- Artur Cichocki

2 1

SpiderMonkey 1.8.5
by Jan Berger 25 Jun '11

25 Jun '11

Does anyone know about an existing Visual Studio build for SpiderMonkey 1.8.5? I am interested in testing it out, but I was hoping to find a pre-build or project for VS 2005, 2008 or 2010. It's only a few months old (march 2011), so might be a bit early yet??? Jan

1 1

Lan redundancy
by Alessandro 25 Jun '11

25 Jun '11

Hi, I'm going to install freeswitch in a system with LAN redundancy (duplicated). All the pc have double LAN interfaces. How can I configure Freeswitch to work with this configuration? Actually in vars, the variable Domains has this value <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/> I have to set one ip of the two network interface? I need to set the sub-net mask? (example domain=192.168.2.101/255.255.255.0) Second question: All the PC with softphone will be connect at two LAN and the two LAN are on different Network. (Example one LAN is on network 192.168.2.0 and the other in the LAN 192.168.1.0). I bind the address of one of the two network to freeswitch. I will add the extension in the internal profile. How does freeswitch understand that an extension is in the local network? All the softphone should stay on the same network, right? What happens if an extension configured in the internal profile, try to contact FS from a different network? Best Regards Alessandro Luppi -- Ing. Alessandro Luppi Software development Seletech srl Via Collodi 8, 20052 Monza (MI) - Italy Tel: +39.039.5962000 - Fax: +39.039.9716905 email: a.luppi(a)seletech.com - Web: www.seletech.com or www.seletech.eu

3 5

Re: [Freeswitch-users] Recovering from database fault
by Clive Lansink 25 Jun '11

25 Jun '11

Yes that's exactly right. You understand the situation correctly. I guess this situation is not going to occur often so I'll just watch to see if it ever happens again. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150 -----Original message----- From: Michael Collins <msc(a)freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Recovering from database fault Reply-to: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Fri, 24 Jun 2011 16:36:42 -0700 On Fri, Jun 24, 2011 at 4:26 PM, Clive Lansink <clive(a)lansink.co.nz> wrote: > Hello Michael and others. > > I deleted the whold db folder which in our case was ok because we don't use > voice mail. Then when I ran Freeswitchconsole, the db folder was recreated. > But that was when we had the problem. The phones could make calls but none > could ring. > Okay, so you removed the whole db folder then restarted FS and everything worked except for the registrations? Then when you did a complete re-install of FS then everything, including registrations, seemed to work? I'd recommend that you update to the latest git and watch and wait. If the symptom returns then maybe you could collect a gcore and open a jira, unless you're on Windows. (We have Windows guys who can guide you on what to do for troubleshooting - I just don't happen to be one of them...) -MC _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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

3 2

mm,m has invited you to join!
by mm,m 25 Jun '11

25 Jun '11

no text body

1 0