April 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

uuid_displace & FIFO help
by Adnan Barakat 01 May '09

01 May '09

Hi All, I'm trying to use uuid_displace to play a message to a caller who is in the fifo queue, but uuid_displace doesn't play until the destination answers. I've tried uuid_broadcast, but the quality of the call becomes *really* bad, then returns to normal quality after the file has finished playing. I have also tried to use streamfile() from lua, but this mixes the audio file with the music on hold of the fifo (also very poor quality). What I am trying to achieve is the ability to play a file (and stop it on demand) to the caller, and/or also be able to use the phrase function to announce the caller's position. At the moment the caller is put into the fifo queue, then a separate script is run via luarun which dials the destination and bridges. Any ideas or suggestions on how to do this would be highly appreciated. Thanks Adnan

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Phones become unreachable after some time
by paul.degt 30 Apr '09

30 Apr '09

I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds to Mysql DB for SIP registrations, presence etc. I noticed that after some time probably >30 min. phones which have been registered but without making calls become unreachable. Meaning that any call to such extension gets forwarded to VM as if it was offline, until I reload such phone. I did try to make the phones to register every 5 min. but it does not help. I also see valid registration information in sip_registrations table. X-Lite has r-port and keep alive settings on. Would appreciate any hints on what can be the issue here.

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Ask for name in conferencing?
by Ross McKillop 30 Apr '09

30 Apr '09

As a former user of the app_confcall (http://www.freeswitch.org/node/ 100) Asterisk module produced by FreeSWITCH I'm in the process of moving a number of Asterisk-based services to FreeSWITCH and am trying to find the equivalent of the "record name before enter" feature of app_confcall. It seems strange that the FreeSWITCH conference module doesn't include this when the one built by the FreeSWITCH developers for Asterisk does... I know I could amend conference.js to do it, however there's no point in re-inventing the wheel if there's already another way. Anyone done this with FreeSWITCH, if so, how? Regards, Ross p.s. if it appears that i've posted this twice please accept my apologies - I tried through Nabble and it failed, so i've used a proper mail client this time...

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Re: [Freeswitch-users] RFC compliance list, FS Road map and RFC.5411
by Richard Lamkin 30 Apr '09

30 Apr '09

Sorry in my eagerness I seem to have sent my original email with a few words missing. Too much cut and paste! Q1 - I have looked on the wiki and was unable to find a list of RFC's that FS is intended to comply with. The page http://wiki.freeswitch.org/wiki/Specsheet#Protocols has a list of SIP protocols by name but these have no RFC number against them. Have I just missed the page?, if not is there any plan to put such a page together? Is there a SIP compliance matrix ? Q2 -Are there any plans to publish an FS road map ? Even a wish list of features which users of FS vote on would be helpful. I know FS is OSS and it does fall to all and not just the core team to implement/extend the product but a road map would be helpful. Q3- I have recently been looking at RFC.5411 which is a basically list of SIP RFC's. A developer or system designer like me is in the future are likely to use an RFC like it as a compliance list for SIP stack selection. Ultimately, it will be the likes of marketing men who will be looking for the one stop shop for a SIP spec who use RFC5411 as the SIP part of a product spec. Are there any plans to use RFC 5411 as a goal? Best Regards Richard Lamkin Mettoni Group UK richard.lamkin(a)mettonigroup.com ************************************************************************* 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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RFC compliance list, FS Road map and RFC.5411
by Richard Lamkin 30 Apr '09

30 Apr '09

Q1 - I have looked on the wiki and was unable to find a list of RFC's that FS is intended to comply with. The page http://wiki.freeswitch.org/wiki/Specsheet#Protocols has a list of SIP protocols by name but these have no RFC number against them. Have I just missed the page?, if not is there any plan to put such a page together? Is there a SIP compliance matrix ? Q2 -This finally brings me on to my question; Are there any plans to publish an FS road map ? Even a wish list of features which users of FS vote on would be helpful. I know FS is OSS and it does fall to all and not just the core team to implement/extend the product but a road map would Q3- I have recently been looking at RFC.5411 which is a basically list of SIP RFC's. A developer or system designer like me is in the future are likely to use an RFC like it as a compliance list for SIP stack selection. Ultimately, it will be the likes of marketing men who will be looking for the one stop shop for a SIP spec who use RFC5411 as the SIP part of a product spec. Are there any plans to use RFC 5411 as a goal? Best Regards Richard Lamkin Mettoni Group UK richard.lamkin(a)mettonigroup.com ************************************************************************* 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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Ask for name in conferencing?
by Ross McKillop 30 Apr '09

30 Apr '09

As a former user of the app_confcall (http://www.freeswitch.org/node/100) Asterisk module produced by FreeSWITCH I'm in the process of moving a number of Asterisk-based services to FreeSWITCH and am trying to find the equivalent of the "record name before enter" feature of app_confcall. It seems strange that the FreeSWITCH conference module doesn't include this when the one built by the FreeSWITCH developers for Asterisk does... I know I could amend conference.js to do it, however there's no point in re-inventing the wheel if there's already another way. Anyone done this with FreeSWITCH, if so, how? Regards, Ross -- View this message in context: http://n2.nabble.com/Ask-for-name-in-conferencing--tp2746159p2746159.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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First Linux soft phone running TLS/SRTP on Linux (Zoipe Bizz)
by Peter P GMX 30 Apr '09

30 Apr '09

After 6 months of discussions with Attractel, today we finally got a new version of Zoiper Bizz, which works with TLS and SRTP (previous versions only supported TLS). I have added the info, how to set it up, in the wiki http://wiki.freeswitch.org/wiki/Interop_List#Zoiper_Bizz_2.10_and_TLS.2FSRTP We've been searching for a long time to have a working secure VoIP client under Linux. So far Zoiper seems to be the only VoIP soft phone capable of managing TLS/SRTP with Freeswitch under Linux. BTW: The free version does not support encryption. The Zoiper "Bizz" version does, but is not for free. Best regards Peter

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dynamically change the presence status
by Antonio Gallo 29 Apr '09

29 Apr '09

Hello, i'm a bit confused about how to port a special function i've on some asterisk system to Freeswitch. In my old asterisk scenario i've all the phones subscribe to extensions <sip:999@PBXIP> this exentions is configured not to monitor a device but a "DeviceState" variable, i.e.: 999,hint,DS/faxserver using dialplan i turn on the blinking status using DevState(DS/FaxServer,${RINGING}) and i turn it off when the call is hanged down. The same applies to other functions: queues, night service, etc. Under freeswitch the solution is http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_presence ? I tried but it didn't worked: presence in|999@PBXIP|dnd|'Do not disturb.' presence out|999@PBXIP|dnd|'Do not disturb.' The LED on the phones (Grandstream GXP 2010 or Snom 320) does not turn RED/ON. Any suggestion?

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Re-2: Re: Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL
by Guido Kuth 29 Apr '09

29 Apr '09

Thank you again Brian. The reason why I want to test with inband dtmf is that in the real environment FS will be behind a conventional ISDN PBX which will work as a gateway to the ISDN Network. So I do not know if the PBX will do something like a translation between DTMF Tones to rfc and backwars. If you have experience with this any help will be very welcome....Guido Original Message processed by David InfoCenter Subject: Re: [Freeswitch-users] Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL (29-Apr-2009 19:59) From: Brian West <brian(a)freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Well the best option is to NOT use inband at all if possible. And use RFC2833 which eyebeam/xlite support as do most providers out there... You do not HAVE to start_dtmf on sip channels unless they only send the DTMF inband. set the dtmf-type back to rfc2833 and restart FS. /b On Apr 29, 2009, at 12:48 PM, Guido Kuth wrote: First thanks for your reply. I have subscribed to all Events, so this can't be the mistake. I sent start_ dtmf app to FreeSwitch in caller channel and the wiki says that you have to do this on sip channels to enable inband dtmf. I checked sofia.conf and I have found that param dtmf-type is commented out. Would it be helpful to set this to "info"? I think setting it to "rfc2833" would not be very meanigfull. I will try to update to svn trunk tomorrow. Again thanks for first help...Guido Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com

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Re-2: Serious Problem detecting DTMF in bridged SIP call using ESL
by Guido Kuth 29 Apr '09

29 Apr '09

First thanks for your reply. I have subscribed to all Events, so this can't be the mistake. I sent start_dtmf app to FreeSwitch in caller channel and the wiki says that you have to do this on sip channels to enable inband dtmf. I checked sofia.conf and I have found that param dtmf-type is commented out. Would it be helpful to set this to "info"? I think setting it to "rfc2833" would not be very meanigfull. I will try to update to svn trunk tomorrow. Again thanks for first help...Guido Original Message processed by David.InfoCenter Subject: Re: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL (29-Apr-2009 19:34) From: Brian West <brian(a)freeswitch.org> To: gk(a)exram.de If you subscribe to the event you will receive one on every DTMF press if FreeSWITCH gets it... if you happen to be getting them via inband you won't receive an event unless you enable the inband detection app. http://wiki.freeswitch.org/wiki/Event_list#DTMF http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf I also highly recommend you update to SVN trunk. /b On Apr 29, 2009, at 12:21 PM, Guido Kuth wrote: What I need is that I get an Event on DTMF Entry on the bridged call. Please I have to resolve this, cause this is the reason why I came from Asterisk to FreeSwitch. Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com

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