- FreeSWITCH-users - FreeSWITCH Mailing Lists

Increase volume of "say" phrases to match play-file?
by Robert Hadley 19 Mar '10

19 Mar '10

Hello, Is there a way to increase the volume of the "say" phrases to match the "play-file" phrases? I am using mod_directory, the sound files from http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/, and the built-in phrases from /usr/local/freeswitch/conf/lang/en/dir/sounds.xml. Thanks, Robert

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dial by name directory // mod_directory?
by William Traenkle 19 Mar '10

19 Mar '10

Has freeswitch implemented the "Dial by Name" feature where a caller can enter the first x number of characters of an extensions first or last name and have the operator either connect the user to the result and/or state the result's extension number? I heard this was a feature that was implemented or going to be implemented in freeswitch. If it has been implemented, would you send me info on how to configure it? If it has not been implemented, would you state when it is planned to be implemented? I believe the module name is mod_directory, is that correct? Thanks, -Will

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registering with auth id
by srinivasula reddy 19 Mar '10

19 Mar '10

HI, i have a user 1002 in freeswitch, i am trying to register 2001 with help of 1002, mean with auth id is 1002 for me, is it possible to register in this way -- Srinivasula Reddy K

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mod_voicemail modifications
by Tim St. Pierre 19 Mar '10

19 Mar '10

Anybody out there know the inner workings of the voice mail module? It looks like I may have to modify the source code to make it do what I want. It's so close, but I can't quite get it to work. I need to change it so that the voicemail_greeting_number channel variable is used to set the greeting path, but is NOT stored back in the sql database. I would also like to change it so that choosing the option to e-mail the message does not mark it as deleted. I imagine I will get many angry phone calls if I put it into production this way. The prompt doesn't tell you that the message will be deleted if you e-mail it, and not all of my users have e-mail set up correctly. I would prefer messages not to be deleted unless delete is explicitly pressed. I'm pretty new to programming, but I would like to give it a try (unless anyone else out there knows a better way). Can someone point me in the right direction? I mostly want to know where the sql gets updated when the variable is set. That wasn't obvious when I looked through the code. In terms of the e-mail causing the deletion, it looks like the e-mail routine is run in-line with the delete routine, so I'm not sure if they can be separated easily. Thanks for any help anyone can provide. -Tim

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Originate timeout with multiple destinations
by Nicolas Brenner 19 Mar '10

19 Mar '10

Hi, I need to do an originate to multiple destinations, but I need that the first call has a timeout of 17 seconds, and if it's not answered, then call the other destinations. I was thinking something like: originate {ignore_early_media=true}[timeout=17]sofia/gateway/mygateway/xxxxxxx, sofia/gateway/mygateway/yyyyyyyy, sofia/gateway/mygateway/zzzzzz I tried using the call_timeout, timeout and originate_timeout variables in the square brackets for the first dialstring, but it didn't work. Is there a way of doing this with 1 originate command? Thanks!

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Conference with FSComm
by Suneel Papineni 19 Mar '10

19 Mar '10

Hi, I am trying to achieve 3-way (or 3-party) conference scenario using FSComm. Scenario is as follows: An incoming call comes to FSComm from an extension 'A' and is answered. Now FSComm makes a call to another extension 'B' and is answered. Then I want to bridge all three calls and be in conference. So all three parties A, B and FSComm will be in conference. Could someone please give me an idea how can I achieve this scenario using Event Sockets. (want to test with fs_cli using commands). Thanks & Regards Suneel ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

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How does mod_directory get users information?
by Robert Hadley 19 Mar '10

19 Mar '10

Hello, Where does mod_directory get the user information it uses to put into the database in creates? Thanks, Robert

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Re: [Freeswitch-users] rtp timeout issue
by Madovsky 19 Mar '10

19 Mar '10

Found the iptables lines that caused problem : #### #Prevent external packets from using loopback -A INPUT -i br0 -s 127.0.0.1 -j DROP -A FORWARD -i br0 -s 127.0.0.1 -j DROP -A INPUT -i br0 -d 127.0.0.1 -j DROP -A FORWARD -i br0 -d 127.0.0.1 -j DROP ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, March 19, 2010 1:53 PM Subject: Re: [Freeswitch-users] rtp timeout issue bingo.... :), grrrr, now I need to read all my iptables :( ----- Original Message ----- From: Anthony Minessale To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, March 19, 2010 1:44 PM Subject: Re: [Freeswitch-users] rtp timeout issue try stopping iptables for 1 call to make sure it is not related to that. On Fri, Mar 19, 2010 at 12:29 PM, Madovsky <infos(a)madovsky.org> wrote: yes I did it also. the issue is only noticeable if I use 2 softphones on the same public IP where there is FS. I tried from another PC (even through nat) and it works fine... Arrgh, I'm lost ----- Original Message ----- From: Anthony Minessale To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, March 19, 2010 1:17 PM Subject: Re: [Freeswitch-users] rtp timeout issue you should also unblock the rtp port range 16384-32768 udp On Fri, Mar 19, 2010 at 12:05 PM, Madovsky <infos(a)madovsky.org> wrote: with bypass-media=true everything is ok, but I need to pass media through FS, I missed some settings but don't know where... I use default configs, inverted the internal and external port and commented out all nat params. Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, March 19, 2010 12:29 PM Subject: rtp timeout issue I set FS internal on port 5080 and external 5060. 2 softphones on public IP, so no nat. the sip invite is ok, but there's a media timeout.. port 5080 and 5060 are opened on my firewall must I set any nat params anyway ? Thanks Franck _______________________________________________ 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 -- 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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 ------------------------------------------------------------------------ _______________________________________________ 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 _______________________________________________ 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 -- 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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 ---------------------------------------------------------------------------- _______________________________________________ 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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Re: [Freeswitch-users] rtp timeout issue
by Madovsky 19 Mar '10

19 Mar '10

with bypass-media=true everything is ok, but I need to pass media through FS, I missed some settings but don't know where... I use default configs, inverted the internal and external port and commented out all nat params. Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, March 19, 2010 12:29 PM Subject: rtp timeout issue I set FS internal on port 5080 and external 5060. 2 softphones on public IP, so no nat. the sip invite is ok, but there's a media timeout.. port 5080 and 5060 are opened on my firewall must I set any nat params anyway ? Thanks Franck

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rtp timeout issue
by Madovsky 19 Mar '10

19 Mar '10

I set FS internal on port 5080 and external 5060. 2 softphones on public IP, so no nat. the sip invite is ok, but there's a media timeout.. port 5080 and 5060 are opened on my firewall must I set any nat params anyway ? Thanks Franck

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