session.steamFile misses DTMF event for first few seconds
I recently upgraded from FS 12790M to svn 17188. When I did, I noticed that session.streamFile behaved differently and I started having problems with my IVR app. With the upgraded FS, I have a problem with streamFile no firing on the DTMF and calling the callback function for the first few seconds of the wav file playback. It behaves as though it does not hear the DTMFs. If I wait for 2 seconds or so of the wav file and then DTMF, streamFile catches the DTMF and all is well. If I key as soon as I hear the wav file start, streamFile just keeps playing the wav and does not call the callback function. When I revert back to the previous version of FS, streamFile always fires the callback right away no matter how quickly I press the first DTMF as the wav file starts to stream out. The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz The snippet of js code I am using is as follows. if(session.ready()) { session.answer(); session.sleep(750); while(session.ready()) { session.sleep(500); session.flushDigits(); // clear out input buffers if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi ts_cb,""))===false) { pin=session.getDigits(pinmax,pinterm,pinwait); } else { pin+=session.getDigits(pinmax-1,pinterm,pinwait); } // more code here.. } Do I need to change the way I use streamFile in the later release?
How are you receiving those DTMFs, inbound or outbound? Are those calls coming from a mobile network (GSM). We have experienced lots of problem with DTMF detection in noisy lines. /aep -- Stopping junk mailers is good for the environment
I recently upgraded from FS 12790M to svn 17188. When I did, I noticed that session.streamFile behaved differently and I started having problems with my IVR app.
With the upgraded FS, I have a problem with streamFile no firing on the DTMF and calling the callback function for the first few seconds of the wav file playback. It behaves as though it does not hear the DTMFs. If I wait for 2 seconds or so of the wav file and then DTMF, streamFile catches the DTMF and all is well. If I key as soon as I hear the wav file start, streamFile just keeps playing the wav and does not call the callback function.
When I revert back to the previous version of FS, streamFile always fires the callback right away no matter how quickly I press the first DTMF as the wav file starts to stream out.
The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
The snippet of js code I am using is as follows.
if(session.ready()) { session.answer(); session.sleep(750); while(session.ready()) { session.sleep(500); session.flushDigits(); // clear out input buffers
if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi ts_cb,""))===false) { pin=session.getDigits(pinmax,pinterm,pinwait); } else { pin+=session.getDigits(pinmax-1,pinterm,pinwait); } // more code here.. }
Do I need to change the way I use streamFile in the later release? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
The calls are coming from land based lines. Traditional POTS (not cable company). Caller is calling into FS and providing those DTMF. I can reproduce on my POTS line (3000' from CO) and there is no discernable noise on the line. The call comes into the media gateway and then goes sip to FS. I can reproduce the problem just by starting FS version from the latest trunk. And then I can eliminate the problem by restarting the FS version 12790. In both test cases, the media gateway remains constant. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Alberto Escudero Sent: Tuesday, April 27, 2010 9:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] session.steamFile misses DTMF event for first few seconds How are you receiving those DTMFs, inbound or outbound? Are those calls coming from a mobile network (GSM). We have experienced lots of problem with DTMF detection in noisy lines. /aep -- Stopping junk mailers is good for the environment
I recently upgraded from FS 12790M to svn 17188. When I did, I noticed that session.streamFile behaved differently and I started having problems with my IVR app.
With the upgraded FS, I have a problem with streamFile no firing on the DTMF and calling the callback function for the first few seconds of the wav file playback. It behaves as though it does not hear the DTMFs. If I wait for 2 seconds or so of the wav file and then DTMF, streamFile catches the DTMF and all is well. If I key as soon as I hear the wav file start, streamFile just keeps playing the wav and does not call the callback function.
When I revert back to the previous version of FS, streamFile always fires the callback right away no matter how quickly I press the first DTMF as the wav file starts to stream out.
The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
The snippet of js code I am using is as follows.
if(session.ready()) { session.answer(); session.sleep(750); while(session.ready()) { session.sleep(500); session.flushDigits(); // clear out input buffers
if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi
ts_cb,""))===false) { pin=session.getDigits(pinmax,pinterm,pinwait); } else { pin+=session.getDigits(pinmax-1,pinterm,pinwait); } // more code here.. }
Do I need to change the way I use streamFile in the later release? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
12XXX is so many years old, i wish users who want free help would at least stay up to date with the code. FYI, our repo is on git now and the svn mirror is not updating at the moment. produce a complete minimal script that reproduces your problem and can be run on git HEAD (see download instructions to learn how to build with git) Use existing sound files from the FS install so we can just run it in our lab to reproduce the issue. Open an issue on http://jira.freeswitch.org and attach the script. On Tue, Apr 27, 2010 at 10:15 AM, Frank @ Impact <frank@impactfax.com>wrote:
The calls are coming from land based lines. Traditional POTS (not cable company). Caller is calling into FS and providing those DTMF. I can reproduce on my POTS line (3000' from CO) and there is no discernable noise on the line.
The call comes into the media gateway and then goes sip to FS.
I can reproduce the problem just by starting FS version from the latest trunk. And then I can eliminate the problem by restarting the FS version 12790. In both test cases, the media gateway remains constant.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Alberto Escudero Sent: Tuesday, April 27, 2010 9:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] session.steamFile misses DTMF event for first few seconds
How are you receiving those DTMFs, inbound or outbound? Are those calls coming from a mobile network (GSM). We have experienced lots of problem with DTMF detection in noisy lines.
/aep -- Stopping junk mailers is good for the environment
I recently upgraded from FS 12790M to svn 17188. When I did, I noticed that session.streamFile behaved differently and I started having problems with my IVR app.
With the upgraded FS, I have a problem with streamFile no firing on the DTMF and calling the callback function for the first few seconds of the wav file playback. It behaves as though it does not hear the DTMFs. If I wait for 2 seconds or so of the wav file and then DTMF, streamFile catches the DTMF and all is well. If I key as soon as I hear the wav file start, streamFile just keeps playing the wav and does not call the callback function.
When I revert back to the previous version of FS, streamFile always fires the callback right away no matter how quickly I press the first DTMF as the wav file starts to stream out.
The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
The snippet of js code I am using is as follows.
if(session.ready()) { session.answer(); session.sleep(750); while(session.ready()) { session.sleep(500); session.flushDigits(); // clear out input buffers
if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi
ts_cb,""))===false) { pin=session.getDigits(pinmax,pinterm,pinwait); } else { pin+=session.getDigits(pinmax-1,pinterm,pinwait); } // more code here.. }
Do I need to change the way I use streamFile in the later release? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@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@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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
The svn repo is updated again now, but it will tend to lag (about a day) behind git going forward. Mike On Apr 27, 2010, at 11:40 AM, Anthony Minessale wrote:
12XXX is so many years old, i wish users who want free help would at least stay up to date with the code.
FYI, our repo is on git now and the svn mirror is not updating at the moment.
produce a complete minimal script that reproduces your problem and can be run on git HEAD (see download instructions to learn how to build with git) Use existing sound files from the FS install so we can just run it in our lab to reproduce the issue.
Open an issue on http://jira.freeswitch.org and attach the script.
On Tue, Apr 27, 2010 at 10:15 AM, Frank @ Impact <frank@impactfax.com> wrote: The calls are coming from land based lines. Traditional POTS (not cable company). Caller is calling into FS and providing those DTMF. I can reproduce on my POTS line (3000' from CO) and there is no discernable noise on the line.
The call comes into the media gateway and then goes sip to FS.
I can reproduce the problem just by starting FS version from the latest trunk. And then I can eliminate the problem by restarting the FS version 12790. In both test cases, the media gateway remains constant.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Alberto Escudero Sent: Tuesday, April 27, 2010 9:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] session.steamFile misses DTMF event for first few seconds
How are you receiving those DTMFs, inbound or outbound? Are those calls coming from a mobile network (GSM). We have experienced lots of problem with DTMF detection in noisy lines.
/aep -- Stopping junk mailers is good for the environment
I recently upgraded from FS 12790M to svn 17188. When I did, I noticed that session.streamFile behaved differently and I started having problems with my IVR app.
With the upgraded FS, I have a problem with streamFile no firing on the DTMF and calling the callback function for the first few seconds of the wav file playback. It behaves as though it does not hear the DTMFs. If I wait for 2 seconds or so of the wav file and then DTMF, streamFile catches the DTMF and all is well. If I key as soon as I hear the wav file start, streamFile just keeps playing the wav and does not call the callback function.
When I revert back to the previous version of FS, streamFile always fires the callback right away no matter how quickly I press the first DTMF as the wav file starts to stream out.
The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
The snippet of js code I am using is as follows.
if(session.ready()) { session.answer(); session.sleep(750); while(session.ready()) { session.sleep(500); session.flushDigits(); // clear out input buffers
if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi
ts_cb,""))===false) { pin=session.getDigits(pinmax,pinterm,pinwait); } else { pin+=session.getDigits(pinmax-1,pinterm,pinwait); } // more code here.. }
Do I need to change the way I use streamFile in the later release? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@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@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 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Yes. I understand your position of trying to help people using old code. That is why I tried to download the latest code to try to stay current. But that is also when I found this problem with the newer code. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Tuesday, April 27, 2010 11:40 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] session.steamFile misses DTMF event forfirst few seconds 12XXX is so many years old, i wish users who want free help would at least stay up to date with the code. FYI, our repo is on git now and the svn mirror is not updating at the moment. produce a complete minimal script that reproduces your problem and can be run on git HEAD (see download instructions to learn how to build with git) Use existing sound files from the FS install so we can just run it in our lab to reproduce the issue. Open an issue on http://jira.freeswitch.org and attach the script. On Tue, Apr 27, 2010 at 10:15 AM, Frank @ Impact <frank@impactfax.com> wrote: The calls are coming from land based lines. Traditional POTS (not cable company). Caller is calling into FS and providing those DTMF. I can reproduce on my POTS line (3000' from CO) and there is no discernable noise on the line. The call comes into the media gateway and then goes sip to FS. I can reproduce the problem just by starting FS version from the latest trunk. And then I can eliminate the problem by restarting the FS version 12790. In both test cases, the media gateway remains constant. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Alberto Escudero Sent: Tuesday, April 27, 2010 9:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] session.steamFile misses DTMF event for first few seconds How are you receiving those DTMFs, inbound or outbound? Are those calls coming from a mobile network (GSM). We have experienced lots of problem with DTMF detection in noisy lines. /aep -- Stopping junk mailers is good for the environment
I recently upgraded from FS 12790M to svn 17188. When I did, I noticed that session.streamFile behaved differently and I started having problems with my IVR app.
With the upgraded FS, I have a problem with streamFile no firing on the DTMF and calling the callback function for the first few seconds of the wav file playback. It behaves as though it does not hear the DTMFs. If I wait for 2 seconds or so of the wav file and then DTMF, streamFile catches the DTMF and all is well. If I key as soon as I hear the wav file start, streamFile just keeps playing the wav and does not call the callback function.
When I revert back to the previous version of FS, streamFile always fires the callback right away no matter how quickly I press the first DTMF as the wav file starts to stream out.
The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
The snippet of js code I am using is as follows.
if(session.ready()) { session.answer(); session.sleep(750); while(session.ready()) { session.sleep(500); session.flushDigits(); // clear out input buffers
if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi
ts_cb,""))===false) { pin=session.getDigits(pinmax,pinterm,pinwait); } else { pin+=session.getDigits(pinmax-1,pinterm,pinwait); } // more code here.. }
Do I need to change the way I use streamFile in the later release? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@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@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 <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
I told you what to do if you want to have the answer. do I need to repost it or did you get the last email? On Wed, Apr 28, 2010 at 5:08 AM, Frank @ Impact <frank@impactfax.com> wrote:
Yes. I understand your position of trying to help people using old code. That is why I tried to download the latest code to try to stay current.
But that is also when I found this problem with the newer code.
-----Original Message----- *From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Tuesday, April 27, 2010 11:40 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] session.steamFile misses DTMF event forfirst few seconds
12XXX is so many years old, i wish users who want free help would at least stay up to date with the code.
FYI, our repo is on git now and the svn mirror is not updating at the moment.
produce a complete minimal script that reproduces your problem and can be run on git HEAD (see download instructions to learn how to build with git) Use existing sound files from the FS install so we can just run it in our lab to reproduce the issue.
Open an issue on http://jira.freeswitch.org and attach the script.
On Tue, Apr 27, 2010 at 10:15 AM, Frank @ Impact <frank@impactfax.com> wrote:
The calls are coming from land based lines. Traditional POTS (not cable company). Caller is calling into FS and providing those DTMF. I can reproduce on my POTS line (3000' from CO) and there is no discernable noise on the line.
The call comes into the media gateway and then goes sip to FS.
I can reproduce the problem just by starting FS version from the latest trunk. And then I can eliminate the problem by restarting the FS version 12790. In both test cases, the media gateway remains constant.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Alberto Escudero Sent: Tuesday, April 27, 2010 9:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] session.steamFile misses DTMF event for first few seconds
How are you receiving those DTMFs, inbound or outbound? Are those calls coming from a mobile network (GSM). We have experienced lots of problem with DTMF detection in noisy lines.
/aep -- Stopping junk mailers is good for the environment
I recently upgraded from FS 12790M to svn 17188. When I did, I noticed that session.streamFile behaved differently and I started having problems with my IVR app.
With the upgraded FS, I have a problem with streamFile no firing on the DTMF and calling the callback function for the first few seconds of the wav file playback. It behaves as though it does not hear the DTMFs. If I wait for 2 seconds or so of the wav file and then DTMF, streamFile catches the DTMF and all is well. If I key as soon as I hear the wav file start, streamFile just keeps playing the wav and does not call the callback function.
When I revert back to the previous version of FS, streamFile always fires the callback right away no matter how quickly I press the first DTMF as the wav file starts to stream out.
The wave file is RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
The snippet of js code I am using is as follows.
if(session.ready()) { session.answer(); session.sleep(750); while(session.ready()) { session.sleep(500); session.flushDigits(); // clear out input buffers
if((pin=session.streamFile(snd_prefix+"/enter-acct-numbers.wav",onlyDigi
ts_cb,""))===false) { pin=session.getDigits(pinmax,pinterm,pinwait); } else { pin+=session.getDigits(pinmax-1,pinterm,pinwait); } // more code here.. }
Do I need to change the way I use streamFile in the later release? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@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@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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@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 <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
participants (4)
-
Alberto Escudero
-
Anthony Minessale
-
Frank ＠ Impact
-
Michael Jerris