[Asterisk-Users] packet8 and asterisk
What is the current state on the packet8 network? Would I be able to send/receive calls through asterisk with their (residential) service, or are there some caveats that I need to be aware of? Thanks to anyone that has any info, Matthias -- Matthias Granberry [EMAIL PROTECTED] (469) 371-0596 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is it possible to do this with Asterisk?
On Thu, 2003-06-19 at 20:17, K a z wrote: > Here's what I am trying to do... > > First I'll have a list of 4 digit numbers like: > > Code:OtherCode > 1234:4321 > : > : > > People will call our 800#, Have the number they > are calling from read to them (ANI?) & then enter > in the code (let's say 1234). If the code matches > one on the list, then the OtherCode (4321 for 1234) > will be read/spoken to them. > > With the exception of the usual recorded prompts, > that's all I'm trying to do here. > > I would like to be able to have this system running on a 24 line card. > > Is it possible to do this with Asterisk? These types of questions annot me to no end. This is a very simple thing to get done, and should have taken no immagination to figure out via the documentation. What you want to do can easily done via many options. The SayNumber will say a number as one large sentence, ie. 1234 will be read as one thousand two hundred thirty four. Their is also SayDigits which will split the 1234 into the component numbers and say each one individually, ie. 1234 will be read one two three four. So this covers the repeating of your code. For the code lookup you can either use PGSQL to do postgres queries for lookup of the code from one code the the other. You could use agi to do anything more complex than a quick lookup. From with AGI you can do saynumber or saydigits. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with CID matching
I'm having a problem with Caller ID matching. The call is coming in via IAX2 to our system, the caller id doesn't seem to parse right. I just got the latest CVS version an hour ago or so. Relative extensions are pretty simple: [disaid] ; ; Check caller id for disa access ; exten => s,1,Wait,0 exten => s/7031234567,1,goto,disa|s|1 exten => s,2,congestion [main] exten => _13019876543,1,Goto,disaid|s|1 I've tried a few variations. IAX2 debug shows: VERSION : 2 CALLED NUMBER : 13019876543 CALLING NUMBER : 7031234567 ANI : 7031234567 LANGUAGE: en USERNAME: xx FORMAT : 4 CAPABILITY : 2147483519 ADSICPE : 2 The result is executed congestion on the call even though the caller id matches. This should work I would think even though the caller id is coming across IAX2 instead of a zap interface. Obviously I've changed the numbers to be not real ones, but they all match like the changed values. Thanks, Steve Radich BitShop, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is it possible to do this with Asterisk?
Yes; a simple AGI script can do this. The script will have access to caller id info and can prompt them for numbers, etc. then lookup in a text file or database to find the value to return. There's a text -> speech engine also available (the name slipped my mind). As well as the usual things like say number. Steve Radich BitShop, Inc. -Original Message- From: K a z [mailto:[EMAIL PROTECTED] Sent: Thursday, June 19, 2003 9:17 PM To: [EMAIL PROTECTED] Here's what I am trying to do... First I'll have a list of 4 digit numbers like: Code:OtherCode 1234:4321 : : People will call our 800#, Have the number they are calling from read to them (ANI?) & then enter in the code (let's say 1234). If the code matches one on the list, then the OtherCode (4321 for 1234) will be read/spoken to them. With the exception of the usual recorded prompts, that's all I'm trying to do here. I would like to be able to have this system running on a 24 line card. Is it possible to do this with Asterisk? _ The new MSN 8: advanced junk mail protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is it possible to do this with Asterisk?
Here's what I am trying to do... First I'll have a list of 4 digit numbers like: Code:OtherCode 1234:4321 : : People will call our 800#, Have the number they are calling from read to them (ANI?) & then enter in the code (let's say 1234). If the code matches one on the list, then the OtherCode (4321 for 1234) will be read/spoken to them. With the exception of the usual recorded prompts, that's all I'm trying to do here. I would like to be able to have this system running on a 24 line card. Is it possible to do this with Asterisk? _ The new MSN 8: advanced junk mail protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] festival error
from the /usr/src/festival folder I ran patch -p1
To: <[EMAIL PROTECTED]>
Sent: Thursday, June 19, 2003 3:18 PM
Subject: Re: [Asterisk-Users] festival error
> You didn't apply the patch to festival did you?
>
> On Thu, 2003-06-19 at 13:55, Chad Sawyer wrote:
> > Did it just like that, still got an error:
> > -- Executing Festival("Zap/5-1", "I am talking. Yes linux can talk.")
in
> > new stack
> > == Parsing '/etc/asterisk/festival.conf': == Parsing
> > '/etc/asterisk/festival.conf': Found
> > voip*CLI> SIOD ERROR: unbound variable : tts_textasterisk
> > == Spawn extension (local, 8195, 1) exited non-zero on 'Zap/5-1'
> >
> >
> >
> >
> > >
> > > You have looked at festival.conf right? What's your
> > > exten line line here's one of mine:
> > >
> > > exten => 1021,1,Festival(mary had a little lamb)
> > >
> > > Note the lack of quotes
> > >
> > > hth
> > >
> > > Andy
> > >
> > > *** REPLY SEPARATOR ***
> > >
> > > On 19/06/2003 at 13:57 Chad Sawyer wrote:
> > >
> > > >I followed the directions I found in the list to a tee
> > > >http://www.marko.net/asterisk/archives/0209/0389.html
> > > >
> > > >The server starts fine, but when I call the festival extension it
gives
> > me
> > > >an OID error variable tts_textasterisk
> > > >
> > > >I have RH7.3
> > > >festival 1.4.2
> > > >speech_tools 1.2.2
> > > >patched it with the festival-1.4.2.diff located in the
/usr/src/asterisk/
> > > >folder . When I patched it, the patch was looking for festival to
have
> > > >been extracted in festival-1.4.2 instead of just "festival". So I
did it
> > > >that way and patched it. Same error.
> > > >
> > > >Any ideas, maybe a nudge? I know it has something to do with that
patch
> > > >adding the variable to festival. The patch says it completed
> > > >successfully...
> > > >
> > > >Thanks
> > > >
> > > >Chad Sawyer, Manager, Network Administrator
> > > >Your Total Communications, LLC
> > > >
> > > >
> > > >
> > > >___
> > > >Asterisk-Users mailing list
> > > >[EMAIL PROTECTED]
> > > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> Steven Critchfield <[EMAIL PROTECTED]>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
___
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RE: [Asterisk-Users] VoiceMail recording dialtone
Does the incoming ring debounce have anything to do with Voicemail recording dialtone? > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Martin Pycko > Sent: Thursday, June 19, 2003 5:57 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] VoiceMail recording dialtone > > Well experiment yourself with the code. > > in wcfxo.c > /* Don't accept a ring for another 1000 ms */ > wc->ringdebounce = 1000; > > Try a diffrent value (e.g. 3000 for 3 sec) > and in zaptel.h > > #defineRING_DEBOUNCE_TIME 500 /* 500 ms ring debounce > time */ > > try the same value as in wcfxo.c > > recompile/reload and test > > regards > Martin > > On Thu, 19 Jun 2003, Sam Bingner wrote: > > > Zaptel was the version from about 4 days ago when I sent this message, I > > updated again yesterday night > > > > Sam > > > > Quoting Martin Pycko <[EMAIL PROTECTED]>: > > > > > How old is your zaptel code ? > > > Mark recently increased some timer for that. > > > > > > Martin > > > > > > On Wed, 18 Jun 2003, Sam Bingner wrote: > > > > > > > I have an extension setup with voicemail, for incoming calls on an > X100P > > > > card. It quite often will record about 15 seconds of dialtone... > I'm > > > > guessing that it picks up the line after the outgoing line has been > > > > disconnected. > > > > > > > > Has anybody else run into this problem? Shouldn't chan_zap be > detecting > > > > the hangup and ending the connection? > > > > > > > > Sam > > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > - > > This mail sent through IMP: http://horde.org/imp/ > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billsec on CDR
It has to do with the fact that with analog channels like FXO we don't have a way to tell whether the call has been answered or not. So after the interfaces sends the called number we assume that the call got answered. This happens unless you have callprogress=yes in zapata.conf. But it's designed to be working only in US. Martin On Thu, 19 Jun 2003, Dan Fernandez wrote: > I have an X100P and when I place calls to the PSTN which are not answered, the > Billsec field of the CDR still logs the seconds that the phone rang. > > Can someone please confirm that this has to do with the ringcadance of the > indications.conf file? Is there anything else I need to check ? > > Thanks in advance > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMail recording dialtone
Well experiment yourself with the code. in wcfxo.c /* Don't accept a ring for another 1000 ms */ wc->ringdebounce = 1000; Try a diffrent value (e.g. 3000 for 3 sec) and in zaptel.h #defineRING_DEBOUNCE_TIME 500 /* 500 ms ring debounce time */ try the same value as in wcfxo.c recompile/reload and test regards Martin On Thu, 19 Jun 2003, Sam Bingner wrote: > Zaptel was the version from about 4 days ago when I sent this message, I > updated again yesterday night > > Sam > > Quoting Martin Pycko <[EMAIL PROTECTED]>: > > > How old is your zaptel code ? > > Mark recently increased some timer for that. > > > > Martin > > > > On Wed, 18 Jun 2003, Sam Bingner wrote: > > > > > I have an extension setup with voicemail, for incoming calls on an X100P > > > card. It quite often will record about 15 seconds of dialtone... I'm > > > guessing that it picks up the line after the outgoing line has been > > > disconnected. > > > > > > Has anybody else run into this problem? Shouldn't chan_zap be detecting > > > the hangup and ending the connection? > > > > > > Sam > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > - > This mail sent through IMP: http://horde.org/imp/ > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] festival error
You patched it, then compiled it, right?
Might want to start from fresh:
Untar
Patch
Build
-wade
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Chad Sawyer
> Sent: Thursday, June 19, 2003 4:35 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] festival error
>
> I applied the patch, one weird thing happened, the patch expected
> festival-1.4.2 folder instead of festival. I told it where the files to
> patch where, I have also tried patch -p1 inside of the festival directory,
> which completed without error, but I still get the error in asterisk.
>
> Running RH7.3
> festival 1.4.2
> speech_tools 1.2.2
>
>
>
> Chad Sawyer, Manager, Network Administrator
> - Original Message -
> From: "Witold P. Krecicki" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, June 19, 2003 3:12 PM
> Subject: Re: [Asterisk-Users] festival error
>
>
> > W liście z czw, 19-06-2003, godz. 20:55, Chad Sawyer pisze:
> > > Did it just like that, still got an error:
> > > -- Executing Festival("Zap/5-1", "I am talking. Yes linux can talk.")
> in
> > > new stack
> > > == Parsing '/etc/asterisk/festival.conf': == Parsing
> > > '/etc/asterisk/festival.conf': Found
> > > voip*CLI> SIOD ERROR: unbound variable : tts_textasterisk
> > > == Spawn extension (local, 8195, 1) exited non-zero on 'Zap/5-1'
> > Are you SURE that you are using festival with applied asterisk patch?
> > what system are You using? I have festival with asterisk patch src.rpm
> > if you want to.
> > WK
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
___
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Re: [Asterisk-Users] VoiceMail recording dialtone
Zaptel was the version from about 4 days ago when I sent this message, I updated again yesterday night Sam Quoting Martin Pycko <[EMAIL PROTECTED]>: > How old is your zaptel code ? > Mark recently increased some timer for that. > > Martin > > On Wed, 18 Jun 2003, Sam Bingner wrote: > > > I have an extension setup with voicemail, for incoming calls on an X100P > > card. It quite often will record about 15 seconds of dialtone... I'm > > guessing that it picks up the line after the outgoing line has been > > disconnected. > > > > Has anybody else run into this problem? Shouldn't chan_zap be detecting > > the hangup and ending the connection? > > > > Sam > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Billsec on CDR
I have an X100P and when I place calls to the PSTN which are not answered, the Billsec field of the CDR still logs the seconds that the phone rang. Can someone please confirm that this has to do with the ringcadance of the indications.conf file? Is there anything else I need to check ? Thanks in advance
Re: [Asterisk-Users] festival error
I applied the patch, one weird thing happened, the patch expected
festival-1.4.2 folder instead of festival. I told it where the files to
patch where, I have also tried patch -p1 inside of the festival directory,
which completed without error, but I still get the error in asterisk.
Running RH7.3
festival 1.4.2
speech_tools 1.2.2
Chad Sawyer, Manager, Network Administrator
- Original Message -
From: "Witold P. Krecicki" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, June 19, 2003 3:12 PM
Subject: Re: [Asterisk-Users] festival error
> W liście z czw, 19-06-2003, godz. 20:55, Chad Sawyer pisze:
> > Did it just like that, still got an error:
> > -- Executing Festival("Zap/5-1", "I am talking. Yes linux can talk.")
in
> > new stack
> > == Parsing '/etc/asterisk/festival.conf': == Parsing
> > '/etc/asterisk/festival.conf': Found
> > voip*CLI> SIOD ERROR: unbound variable : tts_textasterisk
> > == Spawn extension (local, 8195, 1) exited non-zero on 'Zap/5-1'
> Are you SURE that you are using festival with applied asterisk patch?
> what system are You using? I have festival with asterisk patch src.rpm
> if you want to.
> WK
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
___
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Re: [Asterisk-Users] festival error
W liście z czw, 19-06-2003, godz. 20:55, Chad Sawyer pisze:
> Did it just like that, still got an error:
> -- Executing Festival("Zap/5-1", "I am talking. Yes linux can talk.") in
> new stack
> == Parsing '/etc/asterisk/festival.conf': == Parsing
> '/etc/asterisk/festival.conf': Found
> voip*CLI> SIOD ERROR: unbound variable : tts_textasterisk
> == Spawn extension (local, 8195, 1) exited non-zero on 'Zap/5-1'
Are you SURE that you are using festival with applied asterisk patch?
what system are You using? I have festival with asterisk patch src.rpm
if you want to.
WK
___
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Re: [Asterisk-Users] festival error
You didn't apply the patch to festival did you?
On Thu, 2003-06-19 at 13:55, Chad Sawyer wrote:
> Did it just like that, still got an error:
> -- Executing Festival("Zap/5-1", "I am talking. Yes linux can talk.") in
> new stack
> == Parsing '/etc/asterisk/festival.conf': == Parsing
> '/etc/asterisk/festival.conf': Found
> voip*CLI> SIOD ERROR: unbound variable : tts_textasterisk
> == Spawn extension (local, 8195, 1) exited non-zero on 'Zap/5-1'
>
>
>
>
> >
> > You have looked at festival.conf right? What's your
> > exten line line here's one of mine:
> >
> > exten => 1021,1,Festival(mary had a little lamb)
> >
> > Note the lack of quotes
> >
> > hth
> >
> > Andy
> >
> > *** REPLY SEPARATOR ***
> >
> > On 19/06/2003 at 13:57 Chad Sawyer wrote:
> >
> > >I followed the directions I found in the list to a tee
> > >http://www.marko.net/asterisk/archives/0209/0389.html
> > >
> > >The server starts fine, but when I call the festival extension it gives
> me
> > >an OID error variable tts_textasterisk
> > >
> > >I have RH7.3
> > >festival 1.4.2
> > >speech_tools 1.2.2
> > >patched it with the festival-1.4.2.diff located in the /usr/src/asterisk/
> > >folder . When I patched it, the patch was looking for festival to have
> > >been extracted in festival-1.4.2 instead of just "festival". So I did it
> > >that way and patched it. Same error.
> > >
> > >Any ideas, maybe a nudge? I know it has something to do with that patch
> > >adding the variable to festival. The patch says it completed
> > >successfully...
> > >
> > >Thanks
> > >
> > >Chad Sawyer, Manager, Network Administrator
> > >Your Total Communications, LLC
> > >
> > >
> > >
> > >___
> > >Asterisk-Users mailing list
> > >[EMAIL PROTECTED]
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
Steven Critchfield <[EMAIL PROTECTED]>
___
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RE: Re[4]: [Asterisk-Users] == Everyone is busy at this timeproblem
> Plus one priority is for unavailable, specifically for when dial has > timed out. 100 plus priority is for busy, like when the line is already > in use. It's actually 101 plus priority for busy. I got caught by this more than once. :) -wade ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] festival error
Did it just like that, still got an error:
-- Executing Festival("Zap/5-1", "I am talking. Yes linux can talk.") in
new stack
== Parsing '/etc/asterisk/festival.conf': == Parsing
'/etc/asterisk/festival.conf': Found
voip*CLI> SIOD ERROR: unbound variable : tts_textasterisk
== Spawn extension (local, 8195, 1) exited non-zero on 'Zap/5-1'
>
> You have looked at festival.conf right? What's your
> exten line line here's one of mine:
>
> exten => 1021,1,Festival(mary had a little lamb)
>
> Note the lack of quotes
>
> hth
>
> Andy
>
> *** REPLY SEPARATOR ***
>
> On 19/06/2003 at 13:57 Chad Sawyer wrote:
>
> >I followed the directions I found in the list to a tee
> >http://www.marko.net/asterisk/archives/0209/0389.html
> >
> >The server starts fine, but when I call the festival extension it gives
me
> >an OID error variable tts_textasterisk
> >
> >I have RH7.3
> >festival 1.4.2
> >speech_tools 1.2.2
> >patched it with the festival-1.4.2.diff located in the /usr/src/asterisk/
> >folder . When I patched it, the patch was looking for festival to have
> >been extracted in festival-1.4.2 instead of just "festival". So I did it
> >that way and patched it. Same error.
> >
> >Any ideas, maybe a nudge? I know it has something to do with that patch
> >adding the variable to festival. The patch says it completed
> >successfully...
> >
> >Thanks
> >
> >Chad Sawyer, Manager, Network Administrator
> >Your Total Communications, LLC
> >
> >
> >
> >___
> >Asterisk-Users mailing list
> >[EMAIL PROTECTED]
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
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Re: [Asterisk-Users] asterisk -rx under cron?
> "Christopher" == Christopher Arnold <[EMAIL PROTECTED]> writes: Christopher> But is asterisk -rx "show channels" really not meant to Christopher> work from cron or an noninteractive script? If so i guess Christopher> some documentaion about the limitations would help other Christopher> folks out there. You may need to do some 'creative quoting'. Try backslash escaping the quotes in the script. Or put it in a script and have the crontab call that script. I've generally had more luck calling a script than putting complex commands in the crontab; some versions of cron over the years have been known to just break on whitespace, w/o taking account of quotes et al -JimC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] festival error
You have looked at festival.conf right? What's your exten line line here's one of mine: exten => 1021,1,Festival(mary had a little lamb) Note the lack of quotes hth Andy *** REPLY SEPARATOR *** On 19/06/2003 at 13:57 Chad Sawyer wrote: >I followed the directions I found in the list to a tee >http://www.marko.net/asterisk/archives/0209/0389.html > >The server starts fine, but when I call the festival extension it gives me >an OID error variable tts_textasterisk > >I have RH7.3 >festival 1.4.2 >speech_tools 1.2.2 >patched it with the festival-1.4.2.diff located in the /usr/src/asterisk/ >folder . When I patched it, the patch was looking for festival to have >been extracted in festival-1.4.2 instead of just "festival". So I did it >that way and patched it. Same error. > >Any ideas, maybe a nudge? I know it has something to do with that patch >adding the variable to festival. The patch says it completed >successfully... > >Thanks > >Chad Sawyer, Manager, Network Administrator >Your Total Communications, LLC > > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[4]: [Asterisk-Users] == Everyone is busy at this timeproblem
On Thu, 2003-06-19 at 12:57, Eric Wieling wrote:
> > >
> > > exten => _0.,1,StripMSD,1
> > > exten => _.,2,DigitTimeout,10
> > > exten => _.,3,ResponseTimeout,20
> > > exten => _.,4,Dial,Modem/ttyI0:${EXTEN}
> > > exten => _.,4,Dial,Modem/ttyI1:${EXTEN}
>
> This is wrong. Should be:
>
> exten => _.,1,StripMSD,1
> exten => _.,2,DigitTimeout,10
> exten => _.,3,ResponseTimeout,20
> exten => _.,4,Dial,Modem/ttyI0:${EXTEN}
> exten => _.,5,Dial,Modem/ttyI1:${EXTEN}
Plus one priority is for unavailable, specifically for when dial has
timed out. 100 plus priority is for busy, like when the line is already
in use.
--
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[Asterisk-Users] festival error
I followed the directions I found in the list to a tee http://www.marko.net/asterisk/archives/0209/0389.html The server starts fine, but when I call the festival extension it gives me an OID error variable tts_textasterisk I have RH7.3 festival 1.4.2 speech_tools 1.2.2 patched it with the festival-1.4.2.diff located in the /usr/src/asterisk/ folder . When I patched it, the patch was looking for festival to have been extracted in festival-1.4.2 instead of just "festival". So I did it that way and patched it. Same error. Any ideas, maybe a nudge? I know it has something to do with that patch adding the variable to festival. The patch says it completed successfully... Thanks Chad Sawyer, Manager, Network AdministratorYour Total Communications, LLC
Re: Re[4]: [Asterisk-Users] == Everyone is busy at this timeproblem
> >
> > exten => _0.,1,StripMSD,1
> > exten => _.,2,DigitTimeout,10
> > exten => _.,3,ResponseTimeout,20
> > exten => _.,4,Dial,Modem/ttyI0:${EXTEN}
> > exten => _.,4,Dial,Modem/ttyI1:${EXTEN}
This is wrong. Should be:
exten => _.,1,StripMSD,1
exten => _.,2,DigitTimeout,10
exten => _.,3,ResponseTimeout,20
exten => _.,4,Dial,Modem/ttyI0:${EXTEN}
exten => _.,5,Dial,Modem/ttyI1:${EXTEN}
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[Asterisk-Users] Unable to find a path
Hi! I just installed Asterisk 0.4.0 with all the default options, and the configuration samples it has. When I try to dial from an h323 client (gnomemeeting) I get this message on the messages file: Jun 19 11:48:45 WARNING[15375]: File file.c, Line 410 (ast_openstream): File demo-congrats does not exist in any format Jun 19 11:48:45 WARNING[15375]: File file.c, Line 553 (ast_streamfile): Unable to open demo-congrats (format 8): Success Actually, demo-congrats.gsm is installed on /var/lib/asterisk/sounds. show modules says this: Module Description Use Count chan_alsa.so ALSA Console Channel Driver 0 chan_phone.soLinux Telephony API Support 0 chan_local.soLocal Proxy Channel 0 chan_iax2.so Inter Asterisk eXchange (Ver 2) 0 chan_mgcp.so Media Gateway Control Protocol (MGCP)0 chan_agent.soAgent Proxy Channel 0 chan_modem_i4l.soISDN4Linux Emulated Modem Driver 0 chan_modem_bestdata.so BestData (Conexant V.90 Chipset) VoiceMo 0 chan_oss.so OSS Console Channel Driver 0 chan_sip.so Session Initiation Protocol (SIP)0 chan_iax.so Inter Asterisk eXchange 0 res_monitor.so Call Monitoring Resource 1 res_indications.so Indications Configuration0 res_crypto.soCryptographic Digital Signatures 1 res_parking.so Call Parking Resource1 res_adsi.so Call Parking Resource1 pbx_config.soText Extension Configuration 0 chan_oh323.soOpenH323 Channel Driver 0 res_musiconhold.so Music On Hold Resource 1 chan_modem_aopen.so A/Open (Rockwell Chipset) ITU-2 VoiceMod 0 chan_modem.soGeneric Voice Modem Driver 0 Also, I needed to put chan_oh323.so and pbx_config.so in the modules.conf file (asterisk didn't load them automatically). Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[4]: [Asterisk-Users] == Everyone is busy at this time problem
- Original Message -
From: "Angelo Sampietro" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>; "Steven Critchfield"
<[EMAIL PROTECTED]>
Cc: <[EMAIL PROTECTED]>
Sent: Thursday, June 19, 2003 5:30 PM
Subject: Re[4]: [Asterisk-Users] == Everyone is busy at this time problem
> SC> Next tip is that BYEXTENSION is deprecated, use ${EXTEN}. But this
won't
> SC> make your calls work.
>
> SC> Does modem support groups? I think capi does, but I don't remember
modem
> SC> doing it. So try changing the g1 to the actuall tty device.
>
> hi!
> i tried but the result is still the same:
>
> to 10.8.210.147:5060
> == Accepting call on 'SIP/g.carnero-5366' (g.carnero)
> -- Executing Goto("SIP/g.carnero-5366", "doisdn|00115601992|1") in new
stack
> -- Goto (doisdn,00115601992,1)
> -- Executing StripMSD("SIP/g.carnero-5366", "1") in new stack
> -- Executing DigitTimeout("SIP/g.carnero-5366", "10") in new stack
> -- Set Digit Timeout to 10
> -- Executing ResponseTimeout("SIP/g.carnero-5366", "20") in new stack
> -- Set Response Timeout to 20
> -- Executing Dial("SIP/g.carnero-5366", "Modem/ttyI0:0115601992") in
new stack
> -- Called ttyI0:0115601992
> -- Modem[i4l]/ttyI0 is busy
> == Everyone is busy at this time
> -- Hungup 'Modem[i4l]/ttyI0'
>
> this is my configuration now:
>
> exten => _0.,1,StripMSD,1
> exten => _.,2,DigitTimeout,10
> exten => _.,3,ResponseTimeout,20
> exten => _.,4,Dial,Modem/ttyI0:${EXTEN}
> exten => _.,4,Dial,Modem/ttyI1:${EXTEN}
>
> thanks for the suggestion of the deprecated word :)
> if you have any other suggerstin let me know...
> g1 was a group...
> could be the problem related to the i4l driver?
I don't suppose there is a StripMSD in your modem.conf as well is there?
That hit me when I first tried dialing out on i4l.
Regards,
Iain
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RE: [Asterisk-Users] i4l - summary of patches?
--On Thursday, June 19, 2003 17:24:21 +1000 Adam Goryachev <[EMAIL PROTECTED]> wrote: One problem I had with this problem is when I dial out through asterisk, once I have dialled, the remote end doesn't detect my dtmf key presses. ie, I can diall (eg a bank) but when they ask to press 3 for assistance, I can press 3 many times, but they never realise I have pressed it. Any ideas on how to resolve this? Have you installed the asterisk dsp patch for i4l? It's so long since I ran ISDN that I can't remember for sure whether DTMF was handled correctly but I "think" it was. I was using 2 patches - the one to disable kernel DTMF and silence suppression and Pauline's DSP patch for chan_modemi4l. Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re[4]: [Asterisk-Users] == Everyone is busy at this time problem
SC> Next tip is that BYEXTENSION is deprecated, use ${EXTEN}. But this won't
SC> make your calls work.
SC> Does modem support groups? I think capi does, but I don't remember modem
SC> doing it. So try changing the g1 to the actuall tty device.
hi!
i tried but the result is still the same:
to 10.8.210.147:5060
== Accepting call on 'SIP/g.carnero-5366' (g.carnero)
-- Executing Goto("SIP/g.carnero-5366", "doisdn|00115601992|1") in new stack
-- Goto (doisdn,00115601992,1)
-- Executing StripMSD("SIP/g.carnero-5366", "1") in new stack
-- Executing DigitTimeout("SIP/g.carnero-5366", "10") in new stack
-- Set Digit Timeout to 10
-- Executing ResponseTimeout("SIP/g.carnero-5366", "20") in new stack
-- Set Response Timeout to 20
-- Executing Dial("SIP/g.carnero-5366", "Modem/ttyI0:0115601992") in new stack
-- Called ttyI0:0115601992
-- Modem[i4l]/ttyI0 is busy
== Everyone is busy at this time
-- Hungup 'Modem[i4l]/ttyI0'
this is my configuration now:
exten => _0.,1,StripMSD,1
exten => _.,2,DigitTimeout,10
exten => _.,3,ResponseTimeout,20
exten => _.,4,Dial,Modem/ttyI0:${EXTEN}
exten => _.,4,Dial,Modem/ttyI1:${EXTEN}
thanks for the suggestion of the deprecated word :)
if you have any other suggerstin let me know...
g1 was a group...
could be the problem related to the i4l driver?
thaanks a lot for your help
Angelo
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Re: [Asterisk-Users] asterisk -rx under cron?
On Thu, 19 Jun 2003, Steven Critchfield wrote: > > If you follow what was said above, it works interactively, but not > non-interactive. Place that in crontab and it doesn't work as expected. Oops, guess I need to read more carefully. I'll look into this issue since it's likely happening in code that I wrote. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi syntax
This dials the access number and connects... but then connects the call to the sip
phone.. it doesn't do the Wait or SendDTMF...
> Hi,
>
> welcome to the world of digital signalling :-)
>
> use this:
>
> exten => _90.,1,Dial(CAPI/[msn here]:[access number])
> exten => _90.,2,Wait(3)
> exten => _90.,3,SendDTMF(${EXTEN:1})
>
> regards
> kapejod
>
> P.S. inband signalling s*cks ;)
>
> Am Don, 2003-06-19 um 14.08 schrieb WipeOut .:
> > Hi,
> >
> > What is the correct chan_capi dial syntax??
> >
> > This is what I think it is..
> >
> > exten => _90.,1,Dial(CAPI/[msn here]:${EXTEN:1})
> >
> > This seems to work for local numbers.. but I have an access number for cheap long
> > distance calls.. wich gets dialed and then the number I want to call is sent as
> > DTMF after a few waits (w)..
> >
> > On my X100P I used the following..
> >
> > exten => _9001.,1,Dial(Zap/1/[access number]www${EXTEN:1})
> >
> > With chan_capi I tried..
> >
> > exten => _9001.,1,Dial(CAPI/[msn here]:[access number]www${EXTEN:1})
> >
> > But it doesn't seem to work.. it doesn't even dial the access number..
> >
> > Any ideas??
> > --
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Re: [Asterisk-Users] VoiceMail recording dialtone
How old is your zaptel code ? Mark recently increased some timer for that. Martin On Wed, 18 Jun 2003, Sam Bingner wrote: > I have an extension setup with voicemail, for incoming calls on an X100P > card. It quite often will record about 15 seconds of dialtone... I'm > guessing that it picks up the line after the outgoing line has been > disconnected. > > Has anybody else run into this problem? Shouldn't chan_zap be detecting > the hangup and ending the connection? > > Sam > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk -rx under cron?
On Thu, 2003-06-19 at 08:47, James Golovich wrote: > > On Thu, 19 Jun 2003, Christopher Arnold wrote: > > > And it solwed my immediate need too. > > > > But is asterisk -rx "show channels" really not meant to work from cron or > > an noninteractive script? If so i guess some documentaion about the > > limitations would help other folks out there. > > It works just fine for me, the only time it doesn't work is when I don't > put some quotes around the command. > > [EMAIL PROTECTED]:~/cvs/asterisk# asterisk -rx 'show channels' > Channel (ContextExtensionPri ) State Appl. Data > 0 active channel(s) > > [EMAIL PROTECTED]:~/cvs/asterisk# asterisk -rx "show channels" > Channel (ContextExtensionPri ) State Appl. Data > 0 active channel(s) > > [EMAIL PROTECTED]:~/cvs/asterisk# asterisk -rx show channels > No such command 'show' (type 'help' for help) If you follow what was said above, it works interactively, but not non-interactive. Place that in crontab and it doesn't work as expected. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [Asterisk-Users] == Everyone is busy at this time problem
On Thu, 2003-06-19 at 03:17, Angelo Sampietro wrote:
> >> -- Executing Dial("SIP/a.sampietro-f7be", "Modem/g1:BYEXTENSION") in new stack
> >> -- Called g1:0115601992
> >> -- Modem[i4l]/ttyI1 is busy
> >> -- Hungup 'Modem[i4l]/ttyI1'
> >>
> >> someone could help?
>
> >> Look at your dial string, change the colon to a slash.
>
>
> i tried to put the slash instead the colon, but doesn't work
> this is the debug:
>
> [doisdn]
> exten => _0.,1,StripMSD,1
> exten => _.,2,DigitTimeout,10
> exten => _.,3,ResponseTimeout,20
> exten => _.,4,Dial,Modem/g1/BYEXTENSION
>
> (in my previous configuration the last row was: exten
> =>_.,4,Dial,Modem/g1:BYEXTENSION )
>
> [dialout]
> include => interni
> exten => _0.,1,Goto,doisdn|BYEXTENSION|1
>
> i hope that you can find where the error is...
Next tip is that BYEXTENSION is deprecated, use ${EXTEN}. But this won't
make your calls work.
Does modem support groups? I think capi does, but I don't remember modem
doing it. So try changing the g1 to the actuall tty device.
--
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[Asterisk-Users] Newbie: Looking to setup calling between 2 analog phones with a TDM20B
I have a TDM20B and asterisk compiled fine. The drivers have been loaded (wcfxs and run ztcfg -vv, I see the drivers loaded with lsmod). Asterisk starts up fine. I am using the default configuration files that are made when you do a "make samples". I was wondering if someone had a link or website that stepped someone through this kind of setup. What I want to do right now, is use a pair of analog phones plugged into ports 1 and 2 of the TDM20B. I would like to pick up analog phone 1 and call analog phone 2, having it ring. That's all I'm trying to do right now, but I can't seem to get a dial tone or get asterisk to call the other phone. It doesn't seem to detect the ring tones when I dial either. If you could CC: any replies to [EMAIL PROTECTED] it would be appreciated. I do watch the mailing list, but sometimes I miss stuff. Thanks in advance, -- Leif Madsen - Telecommunications Technology Pager (416) 370- Cell (519) ***- SMS [EMAIL PROTECTED] ICQ 344-5119 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] uclibc enviroment #2
ok i have another problem - howto run asterisk as a daemon ( fork ) in uclibc enviroment ? uClinux can only do vfork() a i think this is problem... does anybody know how to solve this ? regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_oh323 problem
Hello I have the following problem using chan_oh323 I have DialGate 2160 for SystemBas (www.sysbas.com) connected to PSTN (H.323 to FXO/FXS gateway) when i try to make call form one pstn phone to other trough asterisk or when i make call from software h.323 client trough asterisk and this gateway to pstn i have the problem with voice quality. The side that initiated call can be heared clearly, but the side that answered the call can be heared aproximatly at half. E.g. on this side everything is heared with some cuts. I runed asterisk -vvvc and during the call it constantly printning: WARNIGN .. File chan_oh323.c, Line 761 (oh323_read) H323:20016 Requested read buffer size is too long (2560) ! and next line message about truncating buffer. I've changed OH323_MAX_BUF to 2560 - warning message disappeared but it didn't improove the quality. I configures Milliwatt aplication and when i ring from PSTN i hear beeps, not continuees tone and its printning warning message in line 1460 : Only doing 640 bytes (1280 bytes requested) (sometimes prints 2560 bytes requested) (from soft client almost everything is ok, only some seldom clicks) I thought that this is the problem of dialgate, but when i use it w/o asterisk it works w/o this problems. Thanks in advance, Andrey
Re: [Asterisk-Users] Wrap-up
Is there a better place to add this type of logic, 1 draw back here
is that moh stops for a moment for user in the app_queue
>In principle, agent_call is supposed to return almost immediately.
>
>Mark
>
>On Wed, 18 Jun 2003, TC wrote:
>
>> I found a similar issue
>> I change'd the chan_agent so that an agent must "press any key to
continue"
>> b4 the agent is actually connected to the next call
>> its just a couple line patch
>>
>> --- chan_agent.c 2003-06-18 15:26:18.0 -0700
>> +++ chan_agent.std 2003-06-18 07:15:08.0 -0700
>> @@ -220,6 +220,7 @@
>>
>> static int agent_call(struct ast_channel *ast, char *dest, int timeout)
>> {
>> +int d;
>> struct agent_pvt *p = ast->pvt->pvt;
>> int res = -1;
>> ast_pthread_mutex_lock(&p->lock);
>> @@ -250,8 +251,12 @@
>> }
>> if( !res )
>> {
>> - /* Call is immediately up */
>> - ast_setstate(ast, AST_STATE_UP);
>> +d = ast_waitfordigit(p->chan, 5000);
>> +if ( d > 0)
>> + /* %TC Call is immediately up */
>> +ast_setstate(ast, AST_STATE_UP);
>> +else
>> +res = 0; /* nothing timed out */
>> }
>> CLEANUP(ast,p);
>> ast_pthread_mutex_unlock(&p->lock);
>>
>>
>>
>> -Original Message-
>> From: Jim Friedeck <[EMAIL PROTECTED]>
>> To: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
>> Date: June 18, 2003 1:29 PM
>> Subject: [Asterisk-Users] Wrap-up
>>
>>
>> >Is it possible to specify a 'wrap-up' time in a queue so agents will
>> >have a specified amount of time to complete tasks between calls unless
>> >they hit a key on the phone? As it is they can recieve a call moments
>> >after they hang up with no 'down time'. Thanks
>> >
>> >Jim Friedeck
>> >
>> >___
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>>
>> ___
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>>
>
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Re: [Asterisk-Users] soundcore???
I assume you mean phonecore, and not soundcore. For some reason it never got added back to cvs, but I have a tarball of it before it vanished. http://asterisk.gnuinter.net/files/cvsnightly/phonecore-cvsnightly.tar.gz James On Thu, 19 Jun 2003, Roy Sigurd Karlsbakk wrote: > hi all > > does anyone have the soundcore lib? I need it for a slackware installation of > gnophone ... > > perhaps it's time to return it to the new cvs? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk -rx under cron?
On Thu, 19 Jun 2003, Christopher Arnold wrote: > And it solwed my immediate need too. > > But is asterisk -rx "show channels" really not meant to work from cron or > an noninteractive script? If so i guess some documentaion about the > limitations would help other folks out there. It works just fine for me, the only time it doesn't work is when I don't put some quotes around the command. [EMAIL PROTECTED]:~/cvs/asterisk# asterisk -rx 'show channels' Channel (ContextExtensionPri ) State Appl. Data 0 active channel(s) [EMAIL PROTECTED]:~/cvs/asterisk# asterisk -rx "show channels" Channel (ContextExtensionPri ) State Appl. Data 0 active channel(s) [EMAIL PROTECTED]:~/cvs/asterisk# asterisk -rx show channels No such command 'show' (type 'help' for help) James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wrap-up
In principle, agent_call is supposed to return almost immediately.
Mark
On Wed, 18 Jun 2003, TC wrote:
> I found a similar issue
> I change'd the chan_agent so that an agent must "press any key to continue"
> b4 the agent is actually connected to the next call
> its just a couple line patch
>
> --- chan_agent.c 2003-06-18 15:26:18.0 -0700
> +++ chan_agent.std 2003-06-18 07:15:08.0 -0700
> @@ -220,6 +220,7 @@
>
> static int agent_call(struct ast_channel *ast, char *dest, int timeout)
> {
> +int d;
> struct agent_pvt *p = ast->pvt->pvt;
> int res = -1;
> ast_pthread_mutex_lock(&p->lock);
> @@ -250,8 +251,12 @@
> }
> if( !res )
> {
> - /* Call is immediately up */
> - ast_setstate(ast, AST_STATE_UP);
> +d = ast_waitfordigit(p->chan, 5000);
> +if ( d > 0)
> + /* %TC Call is immediately up */
> +ast_setstate(ast, AST_STATE_UP);
> +else
> +res = 0; /* nothing timed out */
> }
> CLEANUP(ast,p);
> ast_pthread_mutex_unlock(&p->lock);
>
>
>
> -Original Message-
> From: Jim Friedeck <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
> Date: June 18, 2003 1:29 PM
> Subject: [Asterisk-Users] Wrap-up
>
>
> >Is it possible to specify a 'wrap-up' time in a queue so agents will
> >have a specified amount of time to complete tasks between calls unless
> >they hit a key on the phone? As it is they can recieve a call moments
> >after they hang up with no 'down time'. Thanks
> >
> >Jim Friedeck
> >
> >___
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>
> ___
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[Asterisk-Users] Modems supported by Asterisk
Which modems are supported by Asterisk for full-duplex (as a VoIP-PSTN gateway) operations? I've tried Asmax Mustang (based on TI RK 56000 chipset) but it's 4-bit only, also tried Zoltrix on Rockwell 56k chipset, hacked a lil' bit chan_modem_aopen (so it chooses 7200Hz instead of 8000), sound was 'understandable' but only one-way :/ .Does anybody has experience in using full-duplex voice modem with Asterisk? Are there any cheap alternatives? WK ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] soundcore???
hi all does anyone have the soundcore lib? I need it for a slackware installation of gnophone ... perhaps it's time to return it to the new cvs? -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] compile in uclibc enviroment
> Hi, > > Here it is, attached. Adds a setting in the Makefile where enum > support can be turned off. > > There will probably be some offset when patching due to other changes > in my sources. > > Steve now * compile without errors... but to start * i made following entry in modules.conf under [modules] section : noload => app_enumlookup.so regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk -rx under cron?
On Wed, 18 Jun 2003, Martin Pycko wrote:
> This works for me.
>
> #!/usr/bin/perl -w
> use Socket;
> use IO::Handle;
>
> socket(SOCK, AF_INET, SOCK_STREAM, getprotobyname('tcp'))
> or die "Cannot create a socket: $!\n";
> connect(SOCK, sockaddr_in(5038, inet_aton('localhost')))
> or die "Cannot connect to the manager port\n";
> SOCK->autoflush(1);
> $text = "Action: Login\r\n";
> $text .= "Username: mark\r\n";
> $text .= "Secret: pass\r\n\r\n";
> $text .= "Action: Command\r\nCommand: show channels\r\n\r\n";
> print SOCK $text;
> while () {
> print if not /Message:/ and not /Response:/ and not /END COMMAND/;
> exit 0 if /END COMMAND/
> }
>
> exit 0;
>
And it solwed my immediate need too.
But is asterisk -rx "show channels" really not meant to work from cron or
an noninteractive script? If so i guess some documentaion about the
limitations would help other folks out there.
/Chris
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Re: [Asterisk-Users] chan_capi syntax
Hi,
welcome to the world of digital signalling :-)
use this:
exten => _90.,1,Dial(CAPI/[msn here]:[access number])
exten => _90.,2,Wait(3)
exten => _90.,3,SendDTMF(${EXTEN:1})
regards
kapejod
P.S. inband signalling s*cks ;)
Am Don, 2003-06-19 um 14.08 schrieb WipeOut .:
> Hi,
>
> What is the correct chan_capi dial syntax??
>
> This is what I think it is..
>
> exten => _90.,1,Dial(CAPI/[msn here]:${EXTEN:1})
>
> This seems to work for local numbers.. but I have an access number for cheap long
> distance calls.. wich gets dialed and then the number I want to call is sent as DTMF
> after a few waits (w)..
>
> On my X100P I used the following..
>
> exten => _9001.,1,Dial(Zap/1/[access number]www${EXTEN:1})
>
> With chan_capi I tried..
>
> exten => _9001.,1,Dial(CAPI/[msn here]:[access number]www${EXTEN:1})
>
> But it doesn't seem to work.. it doesn't even dial the access number..
>
> Any ideas??
> --
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>
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[Asterisk-Users] Asterisk support for Voicetronix
Hello Everyone I see in the archives talk of supporting Voicetronix OpenSwitch 6 and 12 cards on Asterisk. How far has this come? TIA /* Tielman Koekemoer Unix and Network Administrator at Vista University Tel: 012-352 4093 Cel: 083-445 0019 */ _ Content and Virus scanned by Inflex and Mcafee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi syntax
Hi,
What is the correct chan_capi dial syntax??
This is what I think it is..
exten => _90.,1,Dial(CAPI/[msn here]:${EXTEN:1})
This seems to work for local numbers.. but I have an access number for cheap long
distance calls.. wich gets dialed and then the number I want to call is sent as DTMF
after a few waits (w)..
On my X100P I used the following..
exten => _9001.,1,Dial(Zap/1/[access number]www${EXTEN:1})
With chan_capi I tried..
exten => _9001.,1,Dial(CAPI/[msn here]:[access number]www${EXTEN:1})
But it doesn't seem to work.. it doesn't even dial the access number..
Any ideas??
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[Asterisk-Users] number of digits from incoming msn on i4l modem
Hi all, is it possible to define the number of digits in the modem.conf ? I think now it is so that i can set a * for any number and concrete numbers for defined numbers. But, i want to define the modem so that it goes into the extensions if the incoming number have a defind number of digits. Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] compile in uclibc enviroment
On Thu, 19 Jun 2003, Holger von Ameln wrote:
> Hi,
>
> Stephen Davis offered to send me a patch that leaves out enum support.
> That would at least solve the undefined references to res_ninit,
> res_nsearch and res_nclose in enum.c.
>
> Cheers,
> Holger
Hi,
Here it is, attached. Adds a setting in the Makefile where enum
support can be turned off.
There will probably be some offset when patching due to other changes
in my sources.
Steve
Index: Makefile
===
RCS file: /usr/cvsroot/asterisk/Makefile,v
retrieving revision 1.17
diff -u -r1.17 Makefile
--- Makefile17 Jun 2003 22:30:25 - 1.17
+++ Makefile19 Jun 2003 10:50:00 -
@@ -51,6 +51,9 @@
#
MALLOC_DEBUG = #-include $(PWD)/include/asterisk/astmm.h
+# Do you want ENUM support?
+ENUM_SUPPORT = #-DENUM_SUPPORT
+
# Where to install asterisk after compiling
# Default -> leave empty
INSTALL_PREFIX=
@@ -85,12 +88,14 @@
INCLUDE=-Iinclude -I../include
CFLAGS=-pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations
$(DEBUG) $(INCLUDE) -D_REENTRANT -D_GNU_SOURCE #-DMAKE_VALGRIND_HAPPY
CFLAGS+=$(OPTIMIZE)
+CFLAGS+=$(ENUM_SUPPORT)
CFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null
2>&1; then echo "-march=$(PROC)"; fi)
CFLAGS+=$(shell if uname -m | grep -q ppc; then echo "-fsigned-char"; fi)
ifeq (${OSARCH},OpenBSD)
CFLAGS+=-pthread
endif
+#CFLAGS+=-DSLD
#CFLAGS+=$(shell if [ -f /usr/include/linux/zaptel.h ]; then echo
"-DZAPTEL_OPTIMIZATIONS"; fi)
LIBEDIT=editline/libedit.a
@@ -125,7 +130,8 @@
ulaw.o alaw.o callerid.o fskmodem.o image.o app.o \
cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o \
dsp.o chanvars.o indications.o autoservice.o db.o privacy.o \
- astmm.o enum.o srv.o
+ astmm.o
+OBJS+=enum.o srv.o
CC=gcc
INSTALL=install
Index: asterisk.c
===
RCS file: /usr/cvsroot/asterisk/asterisk.c,v
retrieving revision 1.11
diff -u -r1.11 asterisk.c
--- asterisk.c 22 May 2003 14:24:06 - 1.11
+++ asterisk.c 19 Jun 2003 10:50:03 -
@@ -1339,10 +1339,12 @@
printf(term_quit());
exit(1);
}
+#ifdef ENUM_SUPPORT
if (ast_enum_init()) {
printf(term_quit());
exit(1);
}
+#endif
/* We might have the option of showing a console, but for now just
do nothing... */
if (option_console && !option_verbose)
Index: enum.c
===
RCS file: /usr/cvsroot/asterisk/enum.c,v
retrieving revision 1.5
diff -u -r1.5 enum.c
--- enum.c 12 Jun 2003 12:48:57 - 1.5
+++ enum.c 19 Jun 2003 10:50:08 -
@@ -11,6 +11,8 @@
*
*/
+#ifdef ENUM_SUPPORT
+
#include
#include
#include
@@ -382,3 +384,5 @@
{
return ast_enum_init();
}
+
+#endif /* -DENUM_SUPPORT */
Index: loader.c
===
RCS file: /usr/cvsroot/asterisk/loader.c,v
retrieving revision 1.5
diff -u -r1.5 loader.c
--- loader.c16 May 2003 02:50:46 - 1.5
+++ loader.c19 Jun 2003 10:50:10 -
@@ -146,7 +146,9 @@
/* We'll do the logger and manager the favor of calling its reload here first
*/
reload_manager();
+#ifdef ENUM_SUPPORT
ast_enum_reload();
+#endif
ast_rtp_reload();
time(&ast_lastreloadtime);
Index: srv.c
===
RCS file: /usr/cvsroot/asterisk/srv.c,v
retrieving revision 1.1
diff -u -r1.1 srv.c
--- srv.c 12 Jun 2003 22:14:03 - 1.1
+++ srv.c 19 Jun 2003 10:50:23 -
@@ -11,6 +11,8 @@
*
*/
+#ifdef ENUM_SUPPORT
+
#include
#include
#include
@@ -297,3 +299,5 @@
res_nclose(&srvstate);
return ret;
}
+
+#endif /* ifdef ENUM_SUPPORT */
Index: channels/chan_sip.c
===
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.125
diff -u -r1.125 chan_sip.c
--- channels/chan_sip.c 18 Jun 2003 22:34:55 - 1.125
+++ channels/chan_sip.c 19 Jun 2003 10:50:50 -
@@ -664,6 +668,7 @@
portno = atoi(port);
else
portno = DEFAULT_SIP_PORT;
+#ifdef ENUM_SUPPORT
if (srvlookup) {
char service[256];
int tportno;
@@ -675,6 +680,7 @@
portno = tportno;
}
}
+#endif
hp = gethostbyname(hostn);
if (hp) {
strncpy(r->tohost, peer, sizeof(r->tohost) - 1);
Re: [Asterisk-Users] compile in uclibc enviroment
Marian Danisek wrote: hello, i try to compile * in uclibc enviroment (uclibc 0.9.19 ), but still getting following error does anyone know how to solve it ? regards Marian - gcc -g -o asterisk -rdynamic io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o -ldl -lpthread -lncurses -lm -lresolv editline/libedit.a db1-ast/libdb1.a enum.o: In function `ast_get_enum': /usr/src/asterisk-cvs/enum.c:279: undefined reference to `__res_ninit' /usr/src/asterisk-cvs/enum.c:307: undefined reference to `__res_nsearch' /usr/src/asterisk-cvs/enum.c:325: undefined reference to `__res_nclose' enum.o: In function `parse_naptr': /usr/src/asterisk-cvs/enum.c:157: undefined reference to `__dn_expand' srv.o: In function `ast_get_srv': /usr/src/asterisk-cvs/srv.c:279: undefined reference to `__res_ninit' /usr/src/asterisk-cvs/srv.c:282: undefined reference to `__res_nsearch' /usr/src/asterisk-cvs/srv.c:297: undefined reference to `__res_nclose' srv.o: In function `parse_srv': /usr/src/asterisk-cvs/srv.c:136: undefined reference to `__dn_expand' collect2: ld returned 1 exit status make: *** [asterisk] Error 1 Hi, Stephen Davis offered to send me a patch that leaves out enum support. That would at least solve the undefined references to res_ninit, res_nsearch and res_nclose in enum.c. Cheers, Holger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] compile in uclibc enviroment
hello, i try to compile * in uclibc enviroment (uclibc 0.9.19 ), but still getting following error does anyone know how to solve it ? regards Marian - gcc -g -o asterisk -rdynamic io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o -ldl -lpthread -lncurses -lm -lresolv editline/libedit.a db1-ast/libdb1.a enum.o: In function `ast_get_enum': /usr/src/asterisk-cvs/enum.c:279: undefined reference to `__res_ninit' /usr/src/asterisk-cvs/enum.c:307: undefined reference to `__res_nsearch' /usr/src/asterisk-cvs/enum.c:325: undefined reference to `__res_nclose' enum.o: In function `parse_naptr': /usr/src/asterisk-cvs/enum.c:157: undefined reference to `__dn_expand' srv.o: In function `ast_get_srv': /usr/src/asterisk-cvs/srv.c:279: undefined reference to `__res_ninit' /usr/src/asterisk-cvs/srv.c:282: undefined reference to `__res_nsearch' /usr/src/asterisk-cvs/srv.c:297: undefined reference to `__res_nclose' srv.o: In function `parse_srv': /usr/src/asterisk-cvs/srv.c:136: undefined reference to `__dn_expand' collect2: ld returned 1 exit status make: *** [asterisk] Error 1 -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream and GSM..
They are thinking of ilbc and said something like 7 months to market ... On Thursday 19 Jun 2003 9:03 am, WipeOut . wrote: > Anyone know if Grandstream are thinking about incorporating the GSM codec > into their phone? > > Would really save the hassle of sorting out G.729 for low bandwidth > requirements.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] == Everyone is busy at this time problem
>> hi,
>> i installed asterisk and works very well, the only problem is that
>> when i try to call a direct number of a company that has a normal PBX
>> i got this error:
>>
>> to 10.8.210.153:5060
>> == Accepting call on 'SIP/a.sampietro-f7be' (a.sampietro)
>> -- Executing Goto("SIP/a.sampietro-f7be", "doisdn|BYEXTENSION|1") in new stack
>> -- Goto (doisdn,00115601992,1)
>> -- Executing StripMSD("SIP/a.sampietro-f7be", "1") in new stack
>> -- Executing DigitTimeout("SIP/a.sampietro-f7be", "10") in new stack
>> -- Set Digit Timeout to 10
>> -- Executing ResponseTimeout("SIP/a.sampietro-f7be", "20") in new stack
>> -- Set Response Timeout to 20
>> -- Executing Dial("SIP/a.sampietro-f7be", "Modem/g1:BYEXTENSION") in new stack
>> -- Called g1:0115601992
>> -- Modem[i4l]/ttyI1 is busy
>> -- Hungup 'Modem[i4l]/ttyI1'
>>
>> someone could help?
>> Look at your dial string, change the colon to a slash.
i tried to put the slash instead the colon, but doesn't work
this is the debug:
to 10.8.210.147:5060
== Accepting call on 'SIP/g.carnero-65eb' (g.carnero)
-- Executing Goto("SIP/g.carnero-65eb", "doisdn|BYEXTENSION|1") in new stack
-- Goto (doisdn,00115601992,1)
-- Executing StripMSD("SIP/g.carnero-65eb", "1") in new stack
-- Executing DigitTimeout("SIP/g.carnero-65eb", "10") in new stack
-- Set Digit Timeout to 10
-- Executing ResponseTimeout("SIP/g.carnero-65eb", "20") in new stack
-- Set Response Timeout to 20
-- Executing Dial("SIP/g.carnero-65eb", "Modem/g1/BYEXTENSION") in new stack
-- Couldn't call g1/0115601992
-- Hungup 'Modem[i4l]/ttyI1'
== Everyone is busy at this time
this is what i did in the extentions.comf when i changed the colon in
slash:
[doisdn]
exten => _0.,1,StripMSD,1
exten => _.,2,DigitTimeout,10
exten => _.,3,ResponseTimeout,20
exten => _.,4,Dial,Modem/g1/BYEXTENSION
(in my previous configuration the last row was: exten =>_.,4,Dial,Modem/g1:BYEXTENSION
)
[dialout]
include => interni
exten => _0.,1,Goto,doisdn|BYEXTENSION|1
i hope that you can find where the error is...
thanks a lot for your help!
best regards
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[Asterisk-Users] Grandstream and GSM..
Anyone know if Grandstream are thinking about incorporating the GSM codec into their phone? Would really save the hassle of sorting out G.729 for low bandwidth requirements.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialogic pricing & Natural Micro-systems support
Hi All, I've had a great success these last couple of days on bringing up Asterisk with a load of Cisco 7940 phones and a couple of soft phones as well. I'd really like to try and link this in to my companies PBX so I can call in from the PSTN as well. I understand there is a charge for the module that would allow me to use my old Dialogic board, can anyway provide some more information and ideally a sales contact ? I was also wondering if NMS E1 boards are/would be supported ? Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Leave one call to pick up another?
Maybe I'm an idiot and this is elementary to everyone but me. Maybe
it's not a function of asterisk. It's driving me nuts.
When I'm on a call and another one comes in for me, in general I *don't*
want to put the current call on hold and pick up; I almost always want
to drop the current call ("Talk to you later. I've got another call.")
and pick up the other one.
But if I flash, asterisk puts the call on hold. If I hang up and let it
ring, oftentimes by the time a ring or two has gone by the second caller
has been sent to voicemail, and I don't think I can barge on voicemail.
One ring doesn't appear to behave any differently than a flash.
What is the way to do this? Is it an RTFM that I have missed?
Thanks.
B.
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RE: [Asterisk-Users] i4l - summary of patches?
One problem I had with this problem is when I dial out through asterisk, once I have dialled, the remote end doesn't detect my dtmf key presses. ie, I can diall (eg a bank) but when they ask to press 3 for assistance, I can press 3 many times, but they never realise I have pressed it. Any ideas on how to resolve this? (I just use 0 for an outside line that doesn't have this problem because it will 'prefer' the analog line on the X100P instead of the BRI on the i4l card. Regards, Adam > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Iain > Stevenson > Sent: Wednesday, 18 June 2003 9:17 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] i4l - summary of patches? > > > > You probably want to remove the i4l handling of DTMF and silence > suppression. You can do this by commenting out the signal processing > routines in the kernel i4l code. This stops wasted work from being done. > I posted the patch below a while back - not sure it it still works. > > Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
