Greetings and a couple of questions
First, to all, greetings. I am just beginning a quest in trying to setup a simple FS box to route my incoming skype account to a SIP ATA (SPA-1001). I have this installed on Ubuntu 8.04 (since I read that it came with a stock tickless kernel with 100HZ tick). I have (as far as I can tell) compiled the alsa 1.0.20 drivers and included the mod_skypiax dummy file. I followed install instructions in the FS wiki for Hardy & FS & skypiax. 1) I am able to see a call come in, and it gets routed to the sample IVR to start. The first thing off is that when I dial from PSTN -> Skype-In, it does not let me push any buttons. If I launch a second skype client, and dial the skype user on FS directly, it works fine. Any ideas? 2) No matter how Skype works, when I hang up, it throws 3 or 4 errors (2010-02-16 20:41:45.384633 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1), and when I try subsequent dials, I get pages and pages of this error, and no audio. Eventually it crashes FS. Any ideas? 3) When dialing extension to extension, or even testing out the IVR, it all works fine - no errors. The only thing it does make me do, and I haven't tracked it down yet is it waits like 30 seconds before responding (unless I press the # after I type the extension number). Any ideas? 4) When I run startskype.sh, I see these errors (and are they worth concerning?) : expected keysym, got XF86KbdLightOnOff: line 70 of pc expected keysym, got XF86KbdBrightnessDown: line 71 of pc expected keysym, got XF86KbdBrightnessUp: line 72 of pc Could not init font path element /usr/share/fonts/X11/cyrillic, removing from list! Could not init font path element /var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list! ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null Thanks for the patience in letting me email to the group. I used Asterisk in the past, and even though I am finding a bit of a learning curve using FS, I am enjoying it. Hopefully someone will have some insight at least into where or how I can settle the above items down, and I can start building a proper dialplan. Thanks, Doc
On Tue, 16 Feb 2010 20:53 -0600, "Doc" <ledoktre@meanie.us> wrote:
1) I am able to see a call come in, and it gets routed to the sample IVR to start. The first thing off is that when I dial from PSTN -> Skype-In, it does not let me push any buttons. If I launch a second skype client, and dial the skype user on FS directly, it works fine. Any ideas?
Hi Doc, When you dial in from the PSTN the 'push button' events are present as "in band" audio tones. By default the sample IVR only works on "out of band" DTMF events. This is why when you call directly from another skype client the 'push button' events are detected because they are passed as "out of band" signaling. Now, In the dial plan you can tell FS to listen for "in band" audio DTMF tones using <action application="start_dtmf" /> However, This currently does not work during a skype call for some reason? http://jira.freeswitch.org/browse/MODSKYPIAX-66 A work around, is to sign up for the "Skype SIP Beta" product where the 'push button' events are sent to FS "out of band". This conversion is done at the PSTN --> Skype gateway by dedicated DTMF tone detection hardware. Skype has either made a business decision, or a technical over sight to pass DTMF events 'out of band' for only addition fee products. It has also been reported in New Zealand that even the 'in band' tones where present one day and actually filtered out the next. This seems extreme, but either through deliberate action or a technology change this is what was reported on one blog. It is unclear to me if this was a technical limitation or a blunt business decisions, but a audio sample showed the audio tones missing? In any case you would not want to rely on 'In band' DTMF' tones when passed through 'lossy' codecs anyway. Best to stick with 'out of band' signaling for reliability. regards, Scott Torr
Once you mentioned the in-band versus out-of-band DTMF, it made sense. I've been reading on it since your post, and I'm going to try a few things. Wonder if there is any app for voice controlled IVR (so many systems seem to support it these days, it would be a nice way to circumvent the DTMF issue... :-) ). Any ideas on the rest of my points? -Thanks, Scott Torr wrote:
On Tue, 16 Feb 2010 20:53 -0600, "Doc" <ledoktre@meanie.us> wrote:
1) I am able to see a call come in, and it gets routed to the sample IVR to start. The first thing off is that when I dial from PSTN -> Skype-In, it does not let me push any buttons. If I launch a second skype client, and dial the skype user on FS directly, it works fine. Any ideas?
Hi Doc,
When you dial in from the PSTN the 'push button' events are present as "in band" audio tones. By default the sample IVR only works on "out of band" DTMF events.
This is why when you call directly from another skype client the 'push button' events are detected because they are passed as "out of band" signaling.
Now, In the dial plan you can tell FS to listen for "in band" audio DTMF tones using <action application="start_dtmf" />
However, This currently does not work during a skype call for some reason? http://jira.freeswitch.org/browse/MODSKYPIAX-66
A work around, is to sign up for the "Skype SIP Beta" product where the 'push button' events are sent to FS "out of band".
This conversion is done at the PSTN --> Skype gateway by dedicated DTMF tone detection hardware.
Skype has either made a business decision, or a technical over sight to pass DTMF events 'out of band' for only addition fee products.
It has also been reported in New Zealand that even the 'in band' tones where present one day and actually filtered out the next. This seems extreme, but either through deliberate action or a technology change this is what was reported on one blog.
It is unclear to me if this was a technical limitation or a blunt business decisions, but a audio sample showed the audio tones missing?
In any case you would not want to rely on 'In band' DTMF' tones when passed through 'lossy' codecs anyway.
Best to stick with 'out of band' signaling for reliability.
regards, Scott Torr
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Hello! For ASR menus check this out: http://wiki.freeswitch.org/wiki/Mod_pocketsphinx <http://wiki.freeswitch.org/wiki/Mod_pocketsphinx>I wouldn't say it is perfect but it is nice, on the other hand there is this dialplan tool that makes fs detect inband dtmf: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf>hope it helps. -Milena 2010/2/17 Doc <ledoktre@meanie.us>
Once you mentioned the in-band versus out-of-band DTMF, it made sense. I've been reading on it since your post, and I'm going to try a few things. Wonder if there is any app for voice controlled IVR (so many systems seem to support it these days, it would be a nice way to circumvent the DTMF issue... :-) ).
Any ideas on the rest of my points? -Thanks,
Scott Torr wrote:
On Tue, 16 Feb 2010 20:53 -0600, "Doc" <ledoktre@meanie.us> wrote:
1) I am able to see a call come in, and it gets routed to the sample IVR to start. The first thing off is that when I dial from PSTN -> Skype-In, it does not let me push any buttons. If I launch a second skype client, and dial the skype user on FS directly, it works fine. Any ideas?
Hi Doc,
When you dial in from the PSTN the 'push button' events are present as "in band" audio tones. By default the sample IVR only works on "out of band" DTMF events.
This is why when you call directly from another skype client the 'push button' events are detected because they are passed as "out of band" signaling.
Now, In the dial plan you can tell FS to listen for "in band" audio DTMF tones using <action application="start_dtmf" />
However, This currently does not work during a skype call for some reason? http://jira.freeswitch.org/browse/MODSKYPIAX-66
A work around, is to sign up for the "Skype SIP Beta" product where the 'push button' events are sent to FS "out of band".
This conversion is done at the PSTN --> Skype gateway by dedicated DTMF tone detection hardware.
Skype has either made a business decision, or a technical over sight to pass DTMF events 'out of band' for only addition fee products.
It has also been reported in New Zealand that even the 'in band' tones where present one day and actually filtered out the next. This seems extreme, but either through deliberate action or a technology change this is what was reported on one blog.
It is unclear to me if this was a technical limitation or a blunt business decisions, but a audio sample showed the audio tones missing?
In any case you would not want to rely on 'In band' DTMF' tones when passed through 'lossy' codecs anyway.
Best to stick with 'out of band' signaling for reliability.
regards, Scott Torr
_______________________________________________ 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
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participants (3)
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Doc
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Milena
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Scott Torr