- FreeSWITCH-users - FreeSWITCH Mailing Lists

MESSAGE Body Encoding
by John Rose 25 Jan '12

25 Jan '12

Does anyone know if the MESSAGE payload can have two encodings in one string when multi-byte text is present? Example: UTF-8 and CP-1252 ? Or is this dependent on the UA or API that populates the body? John

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Noobie question
by Craig Smith 25 Jan '12

25 Jan '12

How do I do a "dialplan show" in FreeSWITCH? I'm trying to troubleshoot why calls aren't going out through a gateway. Thanks in advance.

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Dialplan help
by Elliott Vogel 25 Jan '12

25 Jan '12

Hello, I'm trying to create a dial plan where a user is not registered (online) or an error sip message (480,486,500) is received; I want the call be sent to another sip address. I want to use within this context: <extension name="15618351100"> <condition field="destination_number" expression="^\+(15618351100)$" /> <condition field="caller_id_number" expression="^\+(\d+)$"> <action application="ring_ready" /> <action application="set" data="ignore_early_media=true"/> <action application="bridge" data="user/elliott(a)zoogmedia.com"/<mailto:user/elliott@zoogmedia.com"/>> </condition> </extension>

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Opus codec passthru (or terminate)
by Frank Rosengart 25 Jan '12

25 Jan '12

Hi, how do I have to configure my global codec prefs to pass thru opus codec calls? I guess with mod_opus it's already possible to terminate these calls, right? But anyway, pass thru would be ok. I'm now using baresip as useragent on both ends. Thanks! Frank

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FreeSWITCH ICE Support
by Kristian Kielhofner 25 Jan '12

25 Jan '12

Hello, Has anyone ever investigated adding ICE functionality to FreeSWITCH? While FreeSWITCH happily ignores ICE candidates in the SDP now it would be "really neat" if it could participate in ICE negotiation to find optimal media paths with other ICE implementations. -- Kristian Kielhofner

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Sendmail segfaulting before second message is sent
by Rob Morin 25 Jan '12

25 Jan '12

For some extensions, I've got FreeSWITCH configured to send both the voicemail to an email address and a notification to a cell phone that there is a message. Sendmail is segfaulting (error 6) before the second email message is sent. Sometimes the notification message is going out, sometimes the email message goes. If I am just sending an email, it is *generally* successful. Although sometimes it fails, so I really need to fix this. Does anyone have any suggestions as to how I can troubleshoot this? (Running CentOS 5.7, sendmail 8.13). Thank you, Rob

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Determine what extension picked up after Bridging to a Group Call?
by Frank Busalacchi Jr 25 Jan '12

25 Jan '12

Probably not the right way to do this, but currently I implement "RingGroups" by bridging to a group call (as opposed to FIFO). Is there any way to determine after the bridge is successful which extension in the group call actually answered the call? Example: <extension name="Group-Office"> <condition field="destination_number" expression="^(1200)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="call_timeout=25"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="called_party_callgroup=office"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="${group_call(office@${domain_name}+A)}"/> <action application="transfer" data="1200 XML default"/> <action application="hangup"/> </condition> </extension> I ask as a followup to a question long ago about how to determine the actual call path each call takes. I want to be able to determine that a call came in, was transferred to ext 1200 (The Call Group), and then know which actual extension picked it up. I am doing that in my "Local_Transfer" extension by appending/exporting to a channel variable I have created called ${call_path}. I then have my cdr_csv include ${call_path} in its output. Works nicely except with the "group call". I get 7145551212:1200 Inbound Call_id : transferred to 1200 What I would like to somehow accomplish is: 7145551212:1200:1250 Inbound Call_ID : transferred to 1200 (RingGroup) : Transferred to 1250 (Actual extension in RingGroup that answered). Hope this doesn't sound like gibberish... Thanks for any ideas -Frank

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sofia: disable support for UPDATE message in SIP Accept header
by Michal Zubáč 25 Jan '12

25 Jan '12

Hello. I'd like to remove UPDATE value from SIP Accept header when creating SIP calls. We're sending it in INVITE message and our provider uses that for in-call keep-alive checks every 10 minutes. FreeSwitch doesn't respond to that, so our provider disconnects RTP and call is dropped. According to RFC3311 we can indicate that we don't support this by not sending UPDATE in Accept header. Is this gonna help? Is there any way to drop that from SIP headers from dialplan? Or do I have to change source code? Or better, is there any other (cleaner) way to resolve this? Regards Michal Zubac ComGate Interactive s.r.o.

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Running FreeSWITCH on softmodem
by Sebastian Tarach 25 Jan '12

25 Jan '12

Hello, I'm pretty new to the whole idea of using Linux as telephone central . I tried to setup Asterisk at first but failed. Although I have managed to install it I was unable to "call my computer". That's why I was hoping to deal with FreeSWITCH better. Please tell a noob like myself how to make this finally work. Keep in mind that something that is clear to you to me is probably a dark magic. For instance in installation guide there are mentioned few different ways to connect to a modem. Which method should I use if it comes to my modem : $ lspci -d 8086:24c6 -vvv 00:1f.6 Modem: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Modem Controller (rev 03) (prog-if 00 [Generic]) Subsystem: Toshiba America Info Systems Device 0001 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 4 Region 0: I/O ports at 2400 [size=256] Region 1: I/O ports at 2000 [size=128] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: Intel ICH Modem Thanks in advance

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Re: [Freeswitch-users] announce count mod conference
by Rodney 24 Jan '12

24 Jan '12

yes i would love to implement the announce conference count to user before entering but cant figure out how to "calculate" that. also. the announce count should work because i have tts installed. so i am still concerned why the whole room announce does not work. thanks. ________________________________ From: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Tuesday, January 17, 2012 8:27 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 159 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. FreeSWITCH Community Conference Call Tomorrow (Michael Collins) 2. announce count mod conference (Rodney) 3. Re: announce count mod conference (Michael Collins) 4. Freeswitch & double NAT Configuration (Was: USER_NOT_REGISTERED with external profile) (Alexis de BRUYN [Mailinglists]) 5. Re: Play media dialplan (dan(a)subformat.net) 6. Re: sip_auto_answer auto copy to bleg (Seven Du) Hello all! Just a reminder that we have our FreeSWITCH community conference call tomorrow. We have Moc scheduled to discuss some of the new things he's been cooking up. However, he might have to go play with his cable modem for a bit so we'll see how the schedule pans out. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_01_18 Make a note of the upcoming conference calls. Normally I could say that we are booked solid through February, but since 2012 is a leap year we actually have a call scheduled for February 29. I'd love to hear some suggestions for a special leap-year edition of the FS community conference call! Let me know what you think. :) Thanks, Michael the following section of mod conference does not seem to work for me. is this meant to be played for the whole room or just the user entering the room. I would like it only to work for the person entering the room to cut down on the amount of them pressing the dtmf to execute the announce count extension. it seems this is the first thing they do and I want them to not have to do that. I have set this to 0 and 1 and still no caller count announced. how may I trouble shoot this problem? thanks. when i go into an empty room it says my zerocallers.wav but nothing else if i enter a room with 1 or more callers. for announce-count Requires TTS. The system will speak the total number of callers in the conference when a new person joins, but only once the threshold specified in this parameter is reached. 5 I'm pretty sure that this is for the whole room. You are probably better off calculating the number of users in the conference just prior to sending them in: http://wiki.freeswitch.org/wiki/Conference_Announce_Count_Inline HTH, MC On Tue, Jan 17, 2012 at 11:32 AM, Rodney <notlikeme75(a)yahoo.com> wrote: the following section of mod conference does not seem to work for me. is this meant to be played for the whole room or just the user entering the room. I would like it only to work for the person entering the room to cut down on the amount of them pressing the dtmf to execute the announce count extension. it seems this is the first thing they do and I want them to not have to do that. > > >I have set this to 0 and 1 and still no caller count announced. how may I trouble shoot this problem? thanks. when i go into an empty room it says my zerocallers.wav but nothing else if i enter a room with 1 or more callers. > > > > >for >announce-count Requires TTS. The system will speak the total number of callers in the conference when a new person joins, but only once the threshold specified in this parameter is reached. 5 >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.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 > > Hi Everybody, I am trying to get a conversation between two natted clients and a natted freeswitch server. I am playing with a fresh install, the external profile and default directory (1000 & 1001 users). I can see that my clients are registered on the server logs (on the 5080 port). All others needed ports are opened too. I have followed the instructions from the wiki to get a double nat configuration, but my client phone are not ringing (and I cannot hear anything). All is working fine with defaults in a non nat configuration (with internal profile). Is there anything special to set up in the directory / user configuration ? Can anyone give me some hints with his natted configuration ? I am stuck. Thanks for your help. Regards, -------- Original Message -------- Subject: [Freeswitch-users] USER_NOT_REGISTERED with external profile Date: Mon, 16 Jan 2012 23:37:20 +0100 From: Alexis de BRUYN [Mailinglists] <alexis.mailinglist(a)de-bruyn.fr> Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Hi Everybody, I am trying to use FreeSwitch in a (double) NAT Configuration from a fresh snapshot install (Debian Squeeze, server outside from my client LAN) with default directory for users. When I call 1001 from 1000 (or 1000 from 1001) , I cannot hear the callee, the phone doesn't ring and automatically hangup, I see in the console : 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] However, 1000 and 1001 are registered (from the same LAN) : sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 192.168.1.6 SIP-IP 192.168.1.6 URL sip:mod_sofia@192.168.1.6:5080 BIND-URL sip:mod_sofia@192.168.1.6:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 3 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: aU.1ZlEZkW7nlHpwp9ig1lt7A3weNnwm User: 1001(a)X.Y.Z.T Contact: "Freeswitch" <sip:1001@A.B.C.D:50193> Agent: Bria iOS 2.0.0 Status: Registered(UDP)(unknown) EXP(2012-01-16 23:16:00) EXPSECS(934) Host: phone IP: A.B.C.D Port: 50193 Auth-User: 1001 Auth-Realm: X.Y.Z.T MWI-Account: 1001(a)X.Y.Z.T Call-ID: 3c27b7424837-bsfi877ujb2n User: 1000(a)X.Y.Z.T Contact: "freeswitch" <sip:1000@192.168.0.69:2048;line=skgk4hug> Agent: snom300/8.4.32 Status: Registered(UDP)(unknown) EXP(2012-01-17 00:01:24) EXPSECS(3658) Host: phone IP: A.B.C.D Port: 62061 Auth-User: 1000 Auth-Realm: X.Y.Z.T MWI-Account: 1000(a)X.Y.Z.T Total items returned: 2 ================================================================================================= All necessary ports are opened/forwarded on the server. I See on the 1000 configuration that this is the local ip address which is set as contact. Is there any other setups to do in the directory ? Or other parameters in the external profile ? Thanks for your help ! Regards, -- Alexis de BRUYN Mail : alexis.mailinglist(a)de-bruyn.fr _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 -- -- Alexis de BRUYN Mail : alexis.mailinglist(a)de-bruyn.fr Aaah sorry! That shows just how new I am to FS. Heres the rather large log: http://pastebin.freeswitch.org/18151 Thanks again. Dan. > Oops, you forgot to enable debug. It looks like you're capturing right > from > the console instead of using fs_cli. If you're at the console then type > this before doing a test call: > > console loglevel debug > > Note: if you use fs_cli to connect to FS you'll automatically be at > loglevel debug. Anyway, turn on debug level output and pastebin a new log. > > -MC > > On Tue, Jan 17, 2012 at 2:34 AM, <dan(a)subformat.net> wrote: > >> Thanks MC, Tried that but sleep doesn't appear to be having any effect? >> >> I have pasted my log here: http://pastebin.freeswitch.org/18143 >> >> As you can see in the log the ext is not registered, but the dial plan >> should continue on fail? to the media file etc. >> >> Thanks >> >> Dan. >> >> >> >> > On Mon, Jan 16, 2012 at 11:21 AM, Paul Cupis <paul(a)cupis.co.uk> wrote: >> > >> >> On 16/01/12 14:29, dan(a)subformat.net wrote: >> >> > If you could point me in the right direction, that would be great. >> >> >> >> Can you provide a log of the call from FreeSWITCH, on >> >> http://pastebin.freeswitch.org/ for us to look at, please? >> >> >> >> >> > Also, you might want to add an action right after the answer: >> > <action application="sleep" data="1500"/> >> > >> > Sometimes it takes a moment for media to "come up" and your playback >> could >> > occur before media is happily flowing. If you sleep for a short period >> of >> > time usually that helps. Try making the sleep duration longer or >> shorter >> > to >> > see what happens. >> > >> > -MC >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting(a)freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> > http://www.cudatel.com >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.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 >> > >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.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 >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.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 > I use 7.60.0.110.rom btw. On Wednesday, January 18, 2012 at 1:30 AM, Anthony Minessale wrote: I heard on the beta firmware for yealink that they support the multicast paging as done in esf_page_group from mod_esf > > > >On Tue, Jan 17, 2012 at 12:45 AM, Seven Du <dujinfang(a)gmail.com> wrote: > >seems yealink phone has problem, see >> >> >>http://pastebin.freeswitch.org/18138 >> >> >>there's no intercom in b-leg, but maybe because Call-Info present ? >> >> >> Call-Info: <sip:192.168.7.143>;answer-after=0 >> >> >> >> >> >> >> >> >> >> >>The following confirmed work >> >> >> originate {sip_auto_answer=true}user/1000 &bridge({sip_auto_answer=false,sip_h_Call-info='x'}user/1002) >> >> >> INVITE sip:1002@192.168.7.105:5062 SIP/2.0 >> Call-Info: <x> >> >> >> >> >>On Tuesday, January 17, 2012 at 8:44 AM, Anthony Minessale wrote: >> >>>originate [sip_auto_answer=true]user/1000 &bridge([sip_auto_answer=false]user/1001) >>> >>> >>> >>> >>>On Mon, Jan 16, 2012 at 1:15 PM, Seven Du <dujinfang(a)gmail.com> wrote: >>> >>>Hi, >>>> >>>> >>>>I use originate {sip_auto_answer}user/1000 &bridge(user/1001) and want 1000 to be auto answered but not 1001, but it seems FS automatically copy to b-leg. I manually comment the following lines and it seems to work. >>>> >>>> >>>> >>>> >>>> >>>> >>>>// switch_channel_set_variable(nchannel, "sip_invite_params", "intercom=true"); >>>>} >>>> >>>> >>>>switch_ivr_transfer_variable(session, nsession, SOFIA_REPLACES_HEADER); >>>>// switch_ivr_transfer_variable(session, nsession, "sip_auto_answer"); >>>> >>>> >>>>would it be good to default to not copy but leave the "export" or equivalent to do that? Or should I patch to add a var to disable that? >>>> >>>> >>>>Thanks. >>>> >>>>-- >>>>About: http://about.me/dujinfang >>>>Blog: http://www.dujinfang.com >>>>Proj: http://www.freeswitch.org.cn >>>> >>>>Sent with Sparrow >>>> >>>> >>>> >>>>_________________________________________________________________________ >>>>Professional FreeSWITCH Consulting Services: >>>>consulting(a)freeswitch.org >>>>http://www.freeswitchsolutions.com >>>> >>>>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>http://www.cudatel.com >>>> >>>>Official FreeSWITCH Sites >>>>http://www.freeswitch.org >>>>http://wiki.freeswitch.org >>>>http://www.cluecon.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 >>>> >>>> >>> >>> >>> >>>-- >>>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 >>>googletalk:conf+888@conference.freeswitch.org >>>pstn:+19193869900 >>> >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting(a)freeswitch.org >>>http://www.freeswitchsolutions.com >>> >>> >>>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>http://www.cudatel.com >>> >>> >>>Official FreeSWITCH Sites >>>http://www.freeswitch.org >>>http://wiki.freeswitch.org >>>http://www.cluecon.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 >> >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting(a)freeswitch.org >>http://www.freeswitchsolutions.com >> >>FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>http://www.cudatel.com >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.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 >> >> > > > >-- >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 >googletalk:conf+888@conference.freeswitch.org >pstn:+19193869900 > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting(a)freeswitch.org >http://www.freeswitchsolutions.com > > >FreeSWITCH-powered IP PBX: The CudaTel Communication Server >http://www.cudatel.com > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.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 _______________________________________________ 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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