- FreeSWITCH-users - FreeSWITCH Mailing Lists

Very confusing startup error
by Gerry Hull 29 Apr '09

29 Apr '09

All of a sudden I'm getting this startup error when I start FreeSwitch: C:\DVLP\FreeSwitch>freeswitch Error including C:\DVLP\FreeSwitch\conf\autoload_configs\..\sip_profiles\internal/*.xml (Invalid argument) Cannot Initialize [[error near line 2423]: unexpected closing tag </context>] However, none of the files in conf have a tag called </context>. All files are conforming xml. I can't seem to find what's changed. Any ideas?

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Condition and custom variables
by Alex Gusak 29 Apr '09

29 Apr '09

Hello. Can I use custom variables in the condition field? For example if i set my custom var ${forward_all} and continue search extensions <condition field="${forward_all}" .../> not work even if the condition is met and call does not go in the uplink. Or, in the condition I can use only variables freeswitch like ${sip_from_user}? Is it possible to use custom variables? <extension name="set-vars" continue="true"> <condition> <action application="set" data="forward_all=123456789"/> </condition> </extension> <extension name="call"> <condition field="${forward_all}" expression="^(\d{8,9,10})$" break="on-true"> ... <action application="bridge" data="sofia/gateway/uplink/${forward_all}"/> <action application="hangup"/> </condition> <condition field="destination_number" expression="^(.*)$"> ... <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <action application="hangup"/> </condition> </extension> -- Alex Gusak

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SNMP or Cacti to graph perrformance?
by Ron McCarthy 29 Apr '09

29 Apr '09

Hi List, I have searched and searched and have found nothing. Id like to be able to keep track of current calls and the CPS and then chart these via Cacti. Is their some a SNMP plugin that I have not seen for FS that can do some of these things? Also, if I have to make some custom plugin that montiors output the of show channels /show calls via one of the API's, anyway we can get a realtime stream so I don't have to query it every second to get correct data? Thanks!

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ANNOUNCEMENT: FreeSWITCH 1.0.4pre6 Now Available
by Michael Collins 28 Apr '09

28 Apr '09

The latest pre-release of 1.0.4 is available. More information here: http://www.freeswitch.org/node/177 The FreeSWITCH development team would like to thank everyone for their help in reporting bugs and testing features. Please keep the feedback coming and spread the word about what a cool project FreeSWITCH really is! -Michael See you at ClueCon - http://www.cluecon.com

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Voice mail: skip announcement
by paul.degt 28 Apr '09

28 Apr '09

Is there a way to skip "the person at extension..." announcement when forwarding a call to voice mail, so that it starts recording immediately, or just says something like "start recording..."? Also, is it possible to turn off voice mail play back feature, meaning when somebody tries to leave a voice mail one needs to press a button to end message and then VM suddenly plays it back? I see there's sounds.xml but not sure it help me to do what I need , is there any write up on how to customize it? Would appreciate some directions.

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Re: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre6 Now Available
by Brian West 28 Apr '09

28 Apr '09

Fixed. /b On Apr 28, 2009, at 3:38 PM, Alex Rambau wrote: > The link in the post refers to pre5, not pre6. Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com

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DID Provider
by Saeed Ahmed 28 Apr '09

28 Apr '09

Hi All, Could you please refer me to a best DID provider which also works perfectly with FreeSWITCH. I would need 50 channels per DID number. Thanks.

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FreeSWITCH In Blogosphere
by Michael Collins 28 Apr '09

28 Apr '09

I just wanted to let everyone know that there are a few blog posts out there that you FreeSWITCHers might find interesting. Please check out these two stories on the main FreeSWITCH page: Another Interesting Use For FreeSWITCH: OpenSim<http://www.freeswitch.org/node/175> DIDX Interview With Anthony Minessale and Brian West<http://DIDX%20Interview%20With%20Anthony%20Minessale%20and%20Brian%20West> Be sure to leave comments where appropriate. I will followup with digg links shortly. -MC

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Low rtp-timeout-sec hangs up call in ringing state - expected behaviour?
by Mikael Aleksander Bjerkeland 28 Apr '09

28 Apr '09

Hi, I have been testing inbound calls to a Nokia phone with handover to a cellphone number if I get MEDIA_TIMEOUT on the B leg of the call, and had to set rtp-timeout to a very low 6 seconds in order to get "fast" handover. This introduces an interesting side-effect that hangs up calls even in the ringing state after 6 seconds. Is this the desired behaviour of rtp-timeout-sec? My initial guess was that rtp-timeout-sec should only be valid for established calls where the two endpoints have exchanged rtp at some point but have stopped exchanging media. As far as I know a phone call in ringing state has not shared any RTP with the other endpoint until it gets early media or is answered. Should rtp-timeout-sec really be valid even when ringing? It seems to me that setting rtp-timeout-sec to 60 seconds would add an absolute time limit on ringing phone calls to 60 seconds, which I believe is not the actual purpose of this limit. Could anyone please share their thoughts on this matter? Thanks, Mikael

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Anybody tried Nokia E71 (symbian S60 3rd) with Pjsip and TLS/SRTP?
by Peter P GMX 28 Apr '09

28 Apr '09

Hallo, has anybody tried successfully to setup a Nokia E71 (or similar symbian S60 3rd phone) with Pjsip and TLS/SRTP? TLS seems to work but what about the SRTP part? Best regards Peter

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