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Re: [Freeswitch-users] UK ring tone
by Michael S Collins 13 Mar '10

13 Mar '10

You can set it in the dialplan prior to transferring or bridging the incoming call. Sent from my iPhone On Mar 12, 2010, at 2:22 AM, Gavin Henry <gavin.henry(a)gmail.com> wrote: > Hi, > > I've checked the internal and external sip profiles and think this > needs to be set in autoload switch file as per: > > http://wiki.freeswitch.org/wiki/Variable_transfer_ringback > > I want by default a UK ringtone to be heard when I call a DDI that is > configured on the system. At the moment it sounds like I'm calling the > US. > > Thanks. > > -- http://www.suretecsystems.com/services/openldap/ > http://www.suretectelecom.com > > _______________________________________________ > 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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Gateway context for incoming calls
by Tim St. Pierre 13 Mar '10

13 Mar '10

Hello, Is there a way for incoming calls from a gateway to go to a context other than the default one set in the sofia profile? I need a way to separate calls from providers vs. calls from the public at large. Here's the scenario: Our external sofia profile maps to the public IP address and listens on port 5060. Our providers send calls to this address, but we also need to have anonymous sip calls come in to the public address, on the default port. We want the calls from our SIP trunks to get billed appropriately, and increment our mod_limit counters, etc. If a call comes in from somewhere other than a SIP trunk, we want it to have access to extensions, but be rated as a free call. All our SIP trunks are mapped using static addresses - there isn't any registration or authentication happening in either direction. Each one has a gateway entry. Thanks! -Tim

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voicemail_greeting_number stickiness
by Tim St. Pierre 13 Mar '10

13 Mar '10

Hello all, I'm trying to set up our voice mail system so that greetings can be mapped to failure conditions, ie. "I'm in my office, but not taking calls" when a phone is busy or in DND mode, but "Sorry I'm not here" if no_answer. I programmed things to set voicemail_greeting_number in the dialplan based on the bridge failure cause set when we rang the phone, but when I execute the voicemail application, it stores that greeting number in the database. I would much rather override for this call only, so that the user can still take advantage of the voicemail application's option to select a greeting. (which would be used as the no answer greeting under normal circumstances). Any thoughts? -Tim

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OpenZap/Sangoma drivers
by Mark Sobkow 12 Mar '10

12 Mar '10

I filed a bug report this morning because of a Freeswitch build problem, but the resolution was to update the OpenZap/Sangoma drivers. Updating the Sangoma drivers was a little hairy because of our project build process needing to be updated, but I got it figured out and got the latest build of Freeswitch installed (svn checkout this morning.) Next I need to re-test the call recordings.

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DTMF problem
by Sean Holt 12 Mar '10

12 Mar '10

Question: I have FS 1.0.5 installed on a Centos 5.4 server box. I hope there¹s a simple answer to this question. I¹m having problems in particular when connecting to a outside conference line on a completely different phone system entering numbers that the receiver should accept. For example, call into a conference line, auto-attendent ask for the pin number, I then enter pin number, but The receiving ivr tells me I¹ve entered the wrong pin number. I¹m assuming that I¹ve reached a timeout and the ivr is letting me know that I¹ve entered the incorrect pin number. My thinking is that this is a DTMF issue. I¹m new to FS so not sure if I need to enable something in my dialplan or not. Thoughts? Thanks in advance Sean

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Multiple gateways dial string and user busy
by Max Ivanov 12 Mar '10

12 Mar '10

Hi all! when I do test call from fs_cli: originate sofia/gateway/panas110/223|sofia/gateway/panas111/223 &playaback(local_stream://moh) If firest attempt returns USER_BUSY it tries to call via second one. Is it normal? How can I stop calling attempts after first USER_BUSY?

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UK ring tone
by Gavin Henry 12 Mar '10

12 Mar '10

Hi, I've checked the internal and external sip profiles and think this needs to be set in autoload switch file as per: http://wiki.freeswitch.org/wiki/Variable_transfer_ringback I want by default a UK ringtone to be heard when I call a DDI that is configured on the system. At the moment it sounds like I'm calling the US. Thanks. -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com

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Thangappan Mohan sent you a little gift
by Thangappan Mohan 12 Mar '10

12 Mar '10

Thangappan Mohan is on Skoost and sent you a little gift. To collect your gift, follow the link below: http://www.skoost.com/?id=114961367_5284543 P.S. This is a safe and innocent gift that Thangappan Mohan sent from Skoost. Skoost

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The demo xml dialplan extension mad_boss_intercom problem
by Larry Vatland 11 Mar '10

11 Mar '10

I'm new to freeswitch and just exploring the default dialplan. I have not been able to get the mad_boss_intercom example to work correctly, it pages fine but you should be able to press *2 to start a two-way conversation. Whats happening is the * toggles the phones mute. How do I set up the conference to pass the *2 off to the bind_meta_app to get a page to intercom system working? Thanks for any help. Larry Vatland Computer World 3015 W. Wisconsin Ave. Appleton, WI 54914 (920) 733-9547 lvatland(a)gmail.com

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more than one condition default.xml
by srinivasula reddy 11 Mar '10

11 Mar '10

Hi, can i have more than one conditions in externsion tag in dialplan\default.xml. -- Srinivasula Reddy K

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