June 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Recovering from database fault
by Clive Lansink 24 Jun '11

24 Jun '11

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. 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: Thu, 23 Jun 2011 16:39:01 -0700 Which files in the db/ directory did you actually delete? There are several files in there and generally it is "safe" to delete them all. Note that if you delete the voicemail_xxx.db for your domain then you will lose user voicemail information, like new vs. saved msgs. Of course, the recordings themselves reside in storage/voicemail/xxx/. (xxx = the name of your domain, so if it's the default configs then your domain is "default") -MC On Thu, Jun 23, 2011 at 4:21 PM, Clive Lansink <clive(a)lansink.co.nz> wrote: > Hello list > > After a computer crash yesterday I noticed FreeSwitch reporting that the > database disk image was malformed. After googling on this it seemed that the > thing to do was to either delete or rename the db folder and get FreeSwitch > to recreate it. After I did that and restarted Freeswitch, I found that all > my phones seemed to register ok in that their web interfaces reported they > were registered and I could make calls on them, but none of them could ring. > When I rang another extension, it seemed from the log that the > sofia/internal calling phone would get to the point where it would bridge to > the sofia/internal for the extension, and then there would be an error > something like could not create the channel, cause, user not registered. So > all the phones could make calls but not receive them. > > I needed to get things working quickly and fortunately I achieved this by > brute force - simply uninstalling Freeswitch, getting rid of its folder, > installing again, and copying a saved copy of the conf directory back. Our > installation is simple and we don't rely on the internal voicemail etc, so > that was a quick way to get going again. > > But does anyone know what was going on here? Are there other data files > apart from the db folder we should be aware of? > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > _______________________________________________ > 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 > _______________________________________________ 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

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Trunk Issue
by William Moore 24 Jun '11

24 Jun '11

I can't get any incoming nor outgoing calls. The trunk is through Junction Networks and registers fine. I though I had the dial plan correct but there is obviously something I am missing. I am running Blue.Box GUI with FreeSwitch. It is in CentOS 5.5. I am using Verizon FiOS internet and the firewall is properly configured. This is a test environment so I would be happy to let someone in to take a look. I have SSH setup and web access via phone.statirasystems.com. Other wise it is a lot of configuration info to post. I have checked with Junction Networks and everything is fine on there end. My server is not responding to the requests. I am fairly green with freeswitch and would appreciate any help. Please note: I have sought assistance via the 2600hz community to no avail. Thank you, -- William J. Moore Owner Statira Systems 611 Caroline St Fredericksburg, VA 22401 540.693.0579 www.statirasystems.com

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CRM Integration or Softphone with CRM integration
by William Moore 24 Jun '11

24 Jun '11

I tried searching through the archives. It turn up nothing. Were can I find information on integrating freeswitch in to SugarCRM? And/Or does anyone know of a soft phone that could help with this? What I am looking for is something that will pop up or give a choice for popping up a SugarCRM account based on the caller ID. Thanks, -- William J. Moore Statira Systems 611 Caroline St Fredericksburg, VA http://www.statirasystems.com

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how to use session:playAndGetDigits with the "say" command in lua script?
by Wes 24 Jun '11

24 Jun '11

I'm trying to convert an example from a dialplan: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits#Ex… <extension name="play_and_get_digits with say"> <condition field="destination_number" expression="^(6500)$"> <action application="play_and_get_digits" data="1 1 1 3000 # say:'press one for technical support' silence_stream://250 res \d+" /> </condition> </extension> into a lua statement like this: digits = session:playAndGetDigits(1, 1, 1, 3000, "#", "say:'press one for technicial support' ", "say:'invalid entry'", "\\d+") but I'm having trouble with the quoting. The log says: 2011-06-22 14:40:52.614654 [ERR] switch_ivr_play_say.c:1144 Invalid Args I can't find any lua examples that use both playAndGetDigits and "say" any suggestions?

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Re: [Freeswitch-users] mod_rtmp client API
by Madovsky 24 Jun '11

24 Jun '11

if I type "actionscript mod_rtmp freeswitch" on google nothing that helps a common developer to work on mod_rtmp .... ----- Original Message ----- From: Madovsky To: FreeSWITCH Users Help Sent: Wednesday, June 22, 2011 11:20 AM Subject: Re: [Freeswitch-users] mod_rtmp client API Mike, I didn't ask tools, but to know at least which function it needs to call to the server to make it work ! :) ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Wednesday, June 22, 2011 11:11 AM Subject: Re: [Freeswitch-users] mod_rtmp client API No, it's a DIY project. We provided the server, you provide the client. :D Google around and you will find various tools and pieces to start building your own. -MC On Wed, Jun 22, 2011 at 7:52 AM, Madovsky <infos(a)madovsky.org> wrote: Is there any client side API to use mod_rtmp ? Thanks _______________________________________________ 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 ---------------------------------------------------------------------------- _______________________________________________ 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

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FreeSWITCH-users@lists.freeswitch.org
by Henk Oegema 24 Jun '11

24 Jun '11

How come I don't see my mails in the FreeSWITCH-Users Archives? Rgds. Henk

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(no subject)
by Chris Cureau 24 Jun '11

24 Jun '11

I'm guessing this is just a pipe dream, but I'll throw it out anyway... Is there any database available that can determine the carrier for an outbound number? It would be really nice to be able to route cellular calls via a cell phone automatically instead of setting up specific numbers in the profile.

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Issue with format value 19 in SDP
by Suneel Papineni 24 Jun '11

24 Jun '11

Hi, I am facing an issue with SDP. FS is getting an INVITE with media parameters in SDP as "m=image 23456 udptl t38". For this FS is responding with "m=image 0 udptl 19" in its 183 response. This is correct as per RFC's. Unfortunately my ITSP is expecting the same format what it is sent (i.e. t38 instead of 19). Is there any way in Freeswitch I can specify to the send same format values in response. I tried to look into the code, it is mentioned that 19 is used as it is not used by anybody it is used. (sdp_print.c in libsofia_sip_ua_static project). Could someone please help me how to send all format values that were send in INVITE. Of course according to SDP standards it is enough to send one value. I want to know how to send at least one value which is same as sent in INVITE. Thanks & Regards Suneel

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Loading Pocketsphinx
by Bryan Lemon 24 Jun '11

24 Jun '11

Whenever I try to load the pocketsphinx module, either on startup, or with the load mod_pocketsphinx command, I get the following error: freeswitch@bryanlemon-laptop> load mod_pocketsphinx 2011-06-19 15:17:04.471275 [INFO] mod_enum.c:775 ENUM Reloaded 2011-06-19 15:17:04.471275 [INFO] switch_time.c:1020 Timezone reloaded 530 definitions +OK Reloading XML -ERR [module load file routine returned an error] 2011-06-19 15:17:04.471275 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_pocketsphinx.so **/usr/local/freeswitch/mod/mod_pocketsphinx.so: undefined symbol: feat_init** I am running Ubuntu 10.04, and installed freeswitch using the git method. I tried a git pull && make current, and it runs through with no problem. Any Ideas? Thank you, Bryan Lemon

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Any RECORD_STOP event sample available?
by Frankie Yiu 24 Jun '11

24 Jun '11

Hi there, Could someone please point me where I can find a sample of the RECORD_STOP event look like? I look at this page but it does not have any information about it. ( http://wiki.freeswitch.org/wiki/Event_list#RECORD_STOP) Thanks, Frankie

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