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

29 Apr '09

Hello Anthony, sorry, but I forgot to tell you that I have an inbound ESL connection not an outbound one. So I connect to FS and then wait for Events. I know that I can set async flag in outbound socket, but is this also possible for inbound socket, and when, is it the same as in outbound socket behind the IP-Address? Thank you very much...Guido Original Message processed by David.InfoCenter Subject: Re: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL (29-Apr-2009 19:47) From: Anthony Minessale <anthony.minessale(a)gmail.com> To: gk(a)exram.de set the async flag on the socket app call that triggers your ESL connection On Wed, Apr 29, 2009 at 12:21 PM, Guido Kuth <gk(a)exram.de> wrote: I have a problem I am trying to solve for several days now. I have FS 1.3.0 installed. I have the default configuration except that I have edited event_socket.conf to match my configuration. I have two computers with x-Lite SIP phone 1000 and 1001. Both started and registered. I call in from 1000 and my esl app answers the call plays back a greeting and after that sends a record_session command and a start_dtmf command. Now I send the bridge command with sofia/internal/1001@ip-address. The x-lite 1001 rings and I can take the call the two can talk to each other and both are able to end the call by hanging up the phone, but there is no reaction on any dtmf tone except when I press * and 1-3, cause this is defined by bind-meta-app in default dialplan. 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. Any help or suggestion is welcome. Thanks in advance...Guido _______________________________________________ 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/ 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:213-799-1400

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

29 Apr '09

I have a problem I am trying to solve for several days now. I have FS 1.3.0 installed. I have the default configuration except that I have edited event_socket.conf to match my configuration. I have two computers with x-Lite SIP phone 1000 and 1001. Both started and registered. I call in from 1000 and my esl app answers the call plays back a greeting and after that sends a record_session command and a start_dtmf command. Now I send the bridge command with sofia/internal/1001@ip-address. The x-lite 1001 rings and I can take the call the two can talk to each other and both are able to end the call by hanging up the phone, but there is no reaction on any dtmf tone except when I press * and 1-3, cause this is defined by bind-meta-app in default dialplan. 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. Any help or suggestion is welcome. Thanks in advance...Guido

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Re: [Freeswitch-users] Very confusing startup error
by Guido Kuth 29 Apr '09

29 Apr '09

At least your dialplan should have a tag named <context>. See default dialplan ! Original Message processed by David.InfoCenter Subject: [Freeswitch-users] Very confusing startup error (29-Apr-2009 17:26) From: Gerry Hull <gerry(a)pstn2.net> To: gk(a)exram.de 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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HELP 3-way network access
by Edward Q. 29 Apr '09

29 Apr '09

Hi guys .. I need your help please... I am trying to setup an FS box. It has to be like a 3 way thing since i reside in one network - my FS machine resides on another network - and my provider (gateway) resides on another network. I am going to try to be specific as much as i can. I did a Quick and Dirty install. Here is my testing servers info. Hardware Intel Dual Core 2.6 GHZ Real memory 3.56 GB total, 267.71 MB used Hard Drives 1 SATA 250GB MotherBoard Biostar P4M900-M4 Motherboard - VIA P4M900, Socket 478, MicroATX Software Operating system CentOS Linux 5.2 Kernel and CPU Linux 2.6.18-92.1.22.el5 on i686 Apache 2.2.3 MySQL 5.0.45 SSH OpenSSH 4.3 freeswitch@internal> version FreeSWITCH Version 1.0.trunk (13181M) Ok the FS testing server resides on xxx.9.10.xxx. The gateway resides on xxx.9.9.xxx. And my computer resides on 75.74.xxx.xxx (My computer has X-lite) installed. When i create the SIP profile on X-Lite in my computer and tell X-Lite to register on xxx.9.10.xxx It says discovering network ... Initializing... Registering ... And then it shows up your Your username is: 1000 (looks like it is registered). Now when i try to dial 5000 to listen at least to the IVR demo i get ... The person you are calling is unavailable please try again ... message. and shows on the top of the username ... Call failed: Request Timeout (message) And on the fs_cli console it shows this .. freeswitch@internal> 2009-04-29 11:46:05 [DEBUG] sofia.c:4242 sofia_handle_sip_i_invite() IP 75.74.xxx.xxx Rejected by acl "domains". Falling back to Digest auth. I am a total noob on this. I replaced my original acl.conf.xml with this... <configuration name="acl.conf" description="Network Lists"> <network-lists> <list name="test1" default="deny"> <node type="allow" cidr="75.74.xxx.0/24"/> </list> </network-lists> </configuration> I shutdown FS and then restart FS with the -nc option. And Still the same thing. My gateway is a CANTATA switch which does not require authentication. I am trying to generate a call from my 75.74.xxx.xxx using X-Lite to a PSTN phone on the outside using the CANTATA switch on xxx.9.9.xxx through my FS box on xxx.9.10.xxx But as for now I can't even get the IVR to work for now ... Since i don't know anything about FS i would like to know what am i doing wrong.. And what files have to be either created or updated to do this. Thanks to everyone for all the help Edward

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