I said practically. If they're both broken, then they're practically the same :)

I have the ext-sip-ip and ext-rtp-ip set to my public address in internal.xml, external.xml and the mynetfone external profile.

Where else do I need to set the public IP to get the SDP to report correctly?

Regards,

Richard Adams

Technical Manager

STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA

PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798

Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU

WWW.ZENITEL.COM ● WWW.DNH.NO

cid:image001.jpg@01CD7927.5758B850

From: jay binks [mailto:jaybinks@gmail.com]
Sent: Monday, January 05, 2015 6:09 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Help required

Not identical.

the FROM's differ
one has X-Serialnumber & P-Key-Flags headers.

Both of them have bad IP's in the SDP ( 192.168.34.2 )
this will be key here.. fix this and it may fix all your problems.

that NEEDS To have your external, REAL IP Address.

Do you know what that IP does in the SDP ??

Thats what tells My Net Fone where to send the audio.
how can they route to that IP address, its not globally routable.
so its either going to fail 100% of the time, or they are going to rely on hacks and magic to make a guess at where to send the audio. and sometimes that magic might fail ( like your seeing )

Make sure your RTP IP address is your EXTERNAL IP ! XXX.243.120.79
( XXX Added to protect you from more attacks :P )

On 5 January 2015 at 16:47, Richard Adams <Richard.Adams@stentofon.com.au> wrote:

First invite is a successful call from my desk to mobile, and the second is from external incoming call to my mobile.

The packets are practically identical.

send 1224 bytes to udp/[125.213.160.83]:5060 at 17:41:21.029475:

------------------------------------------------------------------------

INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0

Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKS3jUUDpSeDpXr

Max-Forwards: 33

From: "0397296600" <sip:0397296600@sip20.mynetfone.com.au>;tag=D7U82j7Qj3pKQ

To: <sip:0404058798@sip20.mynetfone.com.au>

Call-ID: b36af627-0f48-1233-04b8-00188b436d1c

CSeq: 69883976 INVITE

Contact: <sip:0397296600@123.243.120.79:5080;transport=udp;gw=MyNetFone>

User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY

Supported: timer, path, replaces

Allow-Events: talk, hold, conference, refer

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 270

Diversion: <sip:86447200@125.213.160.83>;reason=unconditional

X-FS-Support: update_display,send_info

Remote-Party-ID: "0397296600" <sip:0397296600@sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off

v=0

o=FreeSWITCH 1420422599 1420422600 IN IP4 192.168.34.2

s=FreeSWITCH

c=IN IP4 192.168.34.2

t=0 0

m=audio 17482 RTP/AVP 0 8 3 101 13

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

send 1208 bytes to udp/[125.213.160.83]:5060 at 17:42:26.665905:

------------------------------------------------------------------------

INVITE sip:0404058798@sip20.mynetfone.com.au SIP/2.0

Via: SIP/2.0/UDP 123.243.120.79:5080;rport;branch=z9hG4bKX7Qy2SS72gF8p

Max-Forwards: 69

From: "Richard" <sip:0397296600@sip20.mynetfone.com.au>;tag=egN14DrUFcD6j

To: <sip:0404058798@sip20.mynetfone.com.au>

Call-ID: da8a496a-0f48-1233-04b8-00188b436d1c

CSeq: 69884009 INVITE

Contact: <sip:0397296600@123.243.120.79:5080;transport=udp;gw=MyNetFone>

User-Agent: FreeSWITCH-mod_sofia/1.4.15+git~20141229T185951Z~507a0f22c5~64bit

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY

Supported: timer, path, replaces

Allow-Events: talk, hold, conference, refer

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 270

X-Serialnumber: 0004133887C7

P-Key-Flags: keys="3"

X-FS-Support: update_display,send_info

Remote-Party-ID: "Richard" <sip:0397296600@sip20.mynetfone.com.au>;party=calling;screen=yes;privacy=off

v=0

o=FreeSWITCH 1420422380 1420422381 IN IP4 192.168.34.2

s=FreeSWITCH

c=IN IP4 192.168.34.2

t=0 0

m=audio 17766 RTP/AVP 0 8 3 101 13

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

Regards,

Richard Adams

Technical Manager

STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA

PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798

Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU

WWW.ZENITEL.COM ● WWW.DNH.NO

cid:image001.jpg@01CD7927.5758B850

From: jay binks [mailto:jaybinks@gmail.com]
Sent: Monday, January 05, 2015 5:35 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Help required

it might just be easier to pastebin your config files , specifically the SIP profiles for a start.

On 5 January 2015 at 16:16, Richard Adams <Richard.Adams@stentofon.com.au> wrote:

.2 is the PBX.

Bypass media is not on.

I'll look at the RTP-IP.

Regards,

Richard Adams

Technical Manager

STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA

PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798

Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU

WWW.ZENITEL.COM ● WWW.DNH.NO

cid:image001.jpg@01CD7927.5758B850

From: jay binks [mailto:jaybinks@gmail.com]
Sent: Monday, January 05, 2015 5:09 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Help required

That is kinda what happens when you post IP Address in a mailing list or online :(

You can rest assured MyNetFone will also be getting them to their IP address.

OK so the SDP imn your 200 OK definitaly tells MNF to send the media stream to 192.168.34.2 so either your RTP IP is not set, your router is re-writing it with an ALG, or your telling freeswitch to bypass media or something.

What IS 192.168.34.2 ?? if that is your PBX, then that will be option #1 or #2

if .34.2 is a phone handset then id guess its closer to option #3

Jay

On 5 January 2015 at 15:56, Richard Adams <Richard.Adams@stentofon.com.au> wrote:

I have attached a packet trace from a failed call.

We use TPG. The service is exceptional, and we've had only one downtime event in 5 years.

We don't have any call issues on a standard ADSL2+ line with all office internet and phones on the same line.

Some nefarious people are on this mailing list, as I'm now under attack. Hooray!

Regards,

Richard Adams

Technical Manager

STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA

PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798

Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU

WWW.ZENITEL.COM ● WWW.DNH.NO

cid:image001.jpg@01CD7927.5758B850

From: jay binks [mailto:jaybinks@gmail.com]
Sent: Monday, January 05, 2015 4:48 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Help required

Im coming in late to the party here.

Do you have any packet captures of this ???

It sounds like your trying to bypass the media and get out of the RTP stream.

Nobody is going to do that for you :)

Also, from your logs I can see your on a TPG service.

A quick test here shows 8% packet loss and average of 130ms latency with 40ms std dev.

I hope that is just the service your using for your testing ??

Jay Binks

On 5 January 2015 at 15:25, Richard Adams <Richard.Adams@stentofon.com.au> wrote:

UPNP is now off. Same result.

The short version of the call progress is below. For testing, I'm calling from an internal extension 127 to the DID for the Help Line. It fails the same with an external first caller.

2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/127@192.168.34.2 [253bea7c-949a-11e4-9703-192279674221]

2015-01-05 16:17:29.846341 [INFO] mod_dialplan_xml.c:635 Processing Theo <127>->86447200 in context default

2015-01-05 16:17:29.846341 [NOTICE] switch_channel.c:1055 New Channel sofia/external/86447200 [253cc730-949a-11e4-970c-192279674221]

2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/external/61397296600@125.213.160.83 [25584398-949a-11e4-9710-192279674221]

2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context public

2015-01-05 16:17:30.026327 [NOTICE] switch_ivr.c:1854 Transfer sofia/external/61397296600@125.213.160.83 to XML[0386447200@default]

2015-01-05 16:17:30.026327 [INFO] mod_dialplan_xml.c:635 Processing 61397296600 <61397296600>->0386447200 in context default

2015-01-05 16:17:30.026327 [NOTICE] switch_channel.c:1055 New Channel sofia/internal/sip:122@192.168.34.20:2100 [25594e96-949a-11e4-9727-192279674221]

2015-01-05 16:17:30.086342 [NOTICE] sofia.c:6716 Ring-Ready sofia/internal/sip:122@192.168.34.20:2100!

2015-01-05 16:17:30.086342 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/external/61397296600@125.213.160.83!

2015-01-05 16:17:30.106356 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/external/61397296600@125.213.160.83!

2015-01-05 16:17:30.266329 [NOTICE] sofia.c:6716 Ring-Ready sofia/external/86447200!

2015-01-05 16:17:30.286303 [NOTICE] mod_sofia.c:2098 Ring-Ready sofia/internal/127@192.168.34.2!

2015-01-05 16:17:30.286303 [NOTICE] switch_ivr_originate.c:527 Ring Ready sofia/internal/127@192.168.34.2!

2015-01-05 16:17:40.006322 [NOTICE] switch_ivr_originate.c:3495 Hangup sofia/internal/sip:122@192.168.34.20:2100 [CS_CONSUME_MEDIA] [NO_ANSWER]

2015-01-05 16:17:40.026337 [NOTICE] switch_ivr_originate.c:2735 Cannot create outgoing channel of type [USER] cause: [NO_ANSWER]

2015-01-05 16:17:40.026337 [INFO] mod_dptools.c:3234 Originate Failed. Cause: NO_ANSWER

2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1633 Session 8 (sofia/internal/sip:122@192.168.34.20:2100) Ended

2015-01-05 16:17:40.026337 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/sip:122@192.168.34.20:2100 [CS_DESTROY]

2015-01-05 16:17:40.026337 [NOTICE] switch_channel.c:1055 New Channel sofia/external/0404058798 [2b4cddb8-949a-11e4-9730-192279674221]

2015-01-05 16:17:44.646340 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/0404058798!

2015-01-05 16:17:47.586301 [NOTICE] sofia.c:7416 Channel [sofia/external/0404058798] has been answered

2015-01-05 16:17:47.586301 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/external/61397296600@125.213.160.83] has been answered

2015-01-05 16:17:47.846326 [NOTICE] sofia.c:7475 Channel [sofia/external/86447200] has been answered

2015-01-05 16:17:47.866327 [NOTICE] switch_ivr_originate.c:3522 Channel [sofia/internal/127@192.168.34.2] has been answered

2015-01-05 16:17:48.846328 [NOTICE] switch_channel.c:1055 New Channel sofia/external/2057@123.243.120.79 [30909a08-949a-11e4-9736-192279674221]

2015-01-05 16:17:48.846328 [INFO] mod_dialplan_xml.c:635 Processing 2057 <2057>->000972595404572 in context public

2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:315 sofia/external/2057@123.243.120.79 has executed the last dialplan instruction, hanging up.

2015-01-05 16:17:48.846328 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/2057@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING]

2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1633 Session 10 (sofia/external/2057@123.243.120.79) Ended

2015-01-05 16:17:48.846328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/2057@123.243.120.79 [CS_DESTROY]

2015-01-05 16:17:56.026325 [NOTICE] sofia.c:952 Hangup sofia/internal/127@192.168.34.2 [CS_EXECUTE] [NORMAL_CLEARING]

2015-01-05 16:17:56.046305 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/86447200 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]

2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 6 (sofia/external/86447200) Ended

2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/86447200 [CS_DESTROY]

2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1633 Session 5 (sofia/internal/127@192.168.34.2) Ended

2015-01-05 16:17:56.046305 [NOTICE] switch_core_session.c:1637 Close Channel sofia/internal/127@192.168.34.2 [CS_DESTROY]

2015-01-05 16:17:56.246327 [NOTICE] sofia.c:952 Hangup sofia/external/61397296600@125.213.160.83 [CS_EXECUTE] [NORMAL_CLEARING]

2015-01-05 16:17:56.266328 [NOTICE] switch_ivr_bridge.c:754 Hangup sofia/external/0404058798 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]

2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 9 (sofia/external/0404058798) Ended

2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/0404058798 [CS_DESTROY]

2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1633 Session 7 (sofia/external/61397296600@125.213.160.83) Ended

2015-01-05 16:17:56.266328 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/61397296600@125.213.160.83 [CS_DESTROY]

2015-01-05 16:18:03.866325 [NOTICE] switch_channel.c:1055 New Channel sofia/external/8363@123.243.120.79 [39838eb8-949a-11e4-9739-192279674221]

2015-01-05 16:18:03.866325 [INFO] mod_dialplan_xml.c:635 Processing 8363 <8363>->00972595391982 in context public

2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:315 sofia/external/8363@123.243.120.79 has executed the last dialplan instruction, hanging up.

2015-01-05 16:18:03.866325 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/external/8363@123.243.120.79 [CS_EXECUTE] [NORMAL_CLEARING]

2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1633 Session 11 (sofia/external/8363@123.243.120.79) Ended

2015-01-05 16:18:03.866325 [NOTICE] switch_core_session.c:1637 Close Channel sofia/external/8363@123.243.120.79 [CS_DESTROY]

Regards,

Richard Adams

Technical Manager

STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA

PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798

Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU

WWW.ZENITEL.COM ● WWW.DNH.NO

cid:image001.jpg@01CD7927.5758B850

From: Francis [mailto:sms@icefire.qza.net.au]
Sent: Monday, January 05, 2015 4:01 PM

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Help required

WRT autostart, that's built in to the stock Debian Freeswitch packages that you'd get with apt, no need to use git or compile from source. You can choose to install the default vanilla packages, or the whole lot, which is convenient if you want to tinker.

If you're forwarding ports manually, switch off UPNP (Universal Pain 'N' Punishment) since they are mutually exclusive in most cases.

*However*, I suspect that, if normal inbound/outbound calls are working fine and the issue is only when a call is being forwarded out, then this may be something other than NAT/FW, at least in the typical sense. Hop on to FS_CLI and capture some logs of the call progress, then past them here. What you're trying to do should work, I've had success doing similar things bouncing calls between Brisbane and Melbourne.

On 5/01/2015 12:35 PM, Richard Adams wrote:

Ext-rtp-ip and exp-sip-ip are set to the public IP Address of our internet connection.

I pass all incoming calls to the internal profile for routing, and as before, everything works beautifully.

We're using UPNP, and the ports are forwarded correctly.

Regards,

Richard Adams

Technical Manager

STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA

PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798

Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU

WWW.ZENITEL.COM ● WWW.DNH.NO

From: Moishe Grunstein [mailto:max@nysolutions.com]
Sent: Monday, January 05, 2015 1:23 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Help required

Well if internal IP’s are in the SDP you will have audio NAT issues, we need a bit more info about your config to start troubleshooting. What profiles are you using, what do you have set for ext-rtp-ip and ext-sip-ip. Are you using UPNP, STUN? Do you have ports forwarded?

Thanks,

Moishe Grunstein

Tornado Computer Systems, Inc.

212.400.7650 888.IPPBX.US
Service Request Email: support@nysolutions.com

Polycom Certified VAR
Microsoft Small Business Specialist, Cisco SMB Select Certified

Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams
Sent: Sunday, January 4, 2015 8:58 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Help required

Hello,

1. Everything is fine across the NAT for normal incoming and outgoing calls. I can't see any difference in the traces between an outgoing call and a incoming to outgoing call.

2. Compile from source.

Regards,

Richard Adams

Technical Manager

STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA

PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798

Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU

WWW.ZENITEL.COM ● WWW.DNH.NO

From: Moishe Grunstein [mailto:max@nysolutions.com]
Sent: Monday, January 05, 2015 12:34 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Help required

1)      Typical NAT issue, https://freeswitch.org/confluence/display/FREESWITCH/NAT and many other NAT pages in the wiki

2)      Which install method did you use? did you compile from source? Did you create a init.d?

Thanks,

Moishe Grunstein

Tornado Computer Systems, Inc.

212.400.7650 888.IPPBX.US
Service Request Email: support@nysolutions.com

Polycom Certified VAR
Microsoft Small Business Specialist, Cisco SMB Select Certified

Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Adams
Sent: Sunday, January 4, 2015 8:19 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Help required

Trying the mailing list for the first time. Being in Australia puts us out of step with timezones for the IRC.

3 issues;

                1. No audio on calls when an external call is bridged to another external location.

                                We have a help desk number that we'd like to have dial an external call group after hours. Calls are routed first to an internal extension, and then after a 20 second timer, to an external mobile phone number. When the call is answered internally, everything is fine. However, if the call diverts to the mobile phone, the call can be answered but there is no audio in either direction. My SIP Trunk provider is blaming us for sending an internal IP address in the SDP of the invite, but the invite packet is the same for a normal outgoing call.

                2. No autostart Freeswitch on system boot for Debian 7 and current build.

                                To attempt to correct #1, I rebuilt the server with a fresh Debian 7 installation and a Git based build and installation of FS, following the directions for setting the autostart. However, the system does not autostart, and requires the FS to be booted manually after power up.



                3. The auto Makefile did not work. It kept failing on missing dependancies that should have been installed by the auto makefile. I ended up running it manually.

If I can get some guidance please, and I'm happy to provide whatever logging is required.

Regards,

Richard Adams

Technical Manager

STENTOFON AUSTRALIA ● UNIT 2 / 670 MOUNTAIN HIGHWAY ● BAYSWATER VIC 3153 ● AUSTRALIA

PHONE: +61 3 9729 6600 ● FAX: +61 3 9729 0099 ● MOBILE: +61 404 058 798

Richard.Adams@STENTOFON.COM.AU ● WWW.STENTOFON.COM.AU

WWW.ZENITEL.COM ● WWW.DNH.NO
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: 
consulting@freeswitch.org
http://www.freeswitchsolutions.com
 
Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com
 
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

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Sincerely

Jay