Re: [Asterisk-Users] Re: Regarding Incoming Calls on PRI
set immediate=no sometimes you have to wait a bit more time for all the data to come in. On Wed, 9 Mar 2005 22:48:28 -0800 (PST), n a <[EMAIL PROTECTED]> wrote: > Hello > > Well i think that overlapdial=yes would be required if i am trying to dial > from the asterisk side, whereas in my case i am trying to do the opposite. > I think that asterisk would enter the overlap receiving if i send it a setup > request with either no called number or incomplete called number. When i do > this the asterisk unlike sending me a setup acknowledge message (as Euro > ISDN says) it sends me a connect message and then hangs up saying that its > an invalid extension :( > > Any Ideas ??? hope i am not posting on the wrong forum cause it says *-user > not coders :) > > > Regards > Nauman Bin Ali > On March 9, 2005 07:26 am, n a wrote: > How can i configure Asterisk to > enter the overlap receiving state if the > complete number is not obtained > in setup message. I take it the overlapdial=yes option isn't doing what you > want? Perhaps a more detailed explanation of what you're after would help, > including the output of "pri debug span x" with the relevant bits exposed. > -A. > > n a <[EMAIL PROTECTED]> wrote: > > > Hello, > > I am trying to make a call from our PABX to Asterisk on PRI interface. > > How can i configure Asterisk to enter the overlap receiving state if the > complete number is not obtained in setup message. > > Looking forward to any help in this regard > > Regards > > Nauman Bin Ali > > > > __ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > > > Do you Yahoo!? > Yahoo! Mail - Find what you need with new enhanced search. Learn more. > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- Joe Antkowiak antkojm1 (at) gmail.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM phones, bluetooth and general happiness
t only. If you are not the named recipient, please return the > message and enclosures immediately and delete them from your system. > Caution > Before advice received only by email (whether by attachment or > otherwise) may be relied on, the authenticity of the communication must > be verified by means independent of email. > Regulation > The firm is regulated by the Law Society. > Partners > A list of partners is available for inspection at each office of the > firm and on the firm's website at www.winckworths.co.uk > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Joe Antkowiak antkojm1 (at) gmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM phones, bluetooth and general happiness
There are quite a number of positive (for end users) implications of doing this too... just think about all those cell providers that offer unlimited mobile to mobile calls, and then all those unlimited LD packages from landline and voip providers. This has huge potential for people who use their cell phones alot... On Thu, 23 Sep 2004 12:10:11 -0500, Jay Milk <[EMAIL PROTECTED]> wrote: > When I installed my first home-PBX three years ago, I was looking at > "cellsockets" -- devices which will accept certain cellular phones and > provide an RJ11 jack, generating the ring-voltage and recognizing DTMF, > which in turn makes your cell-phone look like a CO line. Pretty cool > stuff, in theory, but it just didn't seem to be worth the cost, > especially since it locks you to a particular cell-phone. > > Since then, I've moved to Asterisk. I looked at at cell-sockets again > recently, but they haven't really gotten any cheaper... And on top of > that, I'd now require a precious FXO interface for *. > > I looked at some developer documentation for my particular phone (S/E > T610) while connecting it to my PC via Bluetooth. For those who are > unaware, all GSM phones have a built-in set of AT modem commands. Not > surprisingly, I was able to place calls as well as receive > ring-indicators, caller-id information and call-progress information via > the virtual serial port that the phone provides over bluetooth. But > what's more, I was also able to utilize my PC as a handsfree > speakerphone -- and all this over bluetooth. > > As I see it, all the pieces are available -- we got full phone control, > some form of digital audio going back and forth, call-progress > reporting. I know there's at least one bluetooth stack for linux, so > *technically* we're "there", no? > > I foresee a chan_blue which allow Asterisk to utilize a bluetooth/GSM > cellular phone as a CO line, connecting by nothing more than a $5 > bluetooth dongle and 5ft of air. > > Who's up for the challenge? If there's enough interest in the > community, I'll be the first to add a bounty on this -- it would be > worth at least $100 to me to have this functionality. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Joe Antkowiak antkojm1 (at) gmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXOs
Just recently installed a multitech mvp810 instead of a t100p and cac adit channel bank. Works perfectly, got rid of all echo issues that nothing else had been able to (all the zap echo cancelers, mediatrix gateway, vegastream gateway, etc etc...) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] successful echo cancellation!!! (multitech)
We recently had a customer install that went horribly wrong. Serious echo (pots lines into a cac cb) that, although * did a good job getting rid of alot of it, could not get rid of it all. We tried everything, every canceller, gain setting, etc... combination possible to no avail. Both the vegastream and mediatrix boxes also could not get rid of all of the echo. So, on an off chance, we bought an 8 port fxs/fxo/e&m gateway made by multitech. The echo cancellation on this device is amazing. There is no trace of the echo and the conversation is still full duplex. And, the box works perfectly with asterisk. Unfortunately, the retail price on these boxes is $3k. Just thought I'd share my experience... -- ____ Joe Antkowiak antkojm1 (at) gmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco phone and parked calls
h... Is there any way to make it say the parking lot space a call
is being parked into, on the channel that is calling into the
extension that is running the ValetParkCall app? My customer wants to
know what space it is without having to listen to all the parked
calls, and uses attended transfer...
On Thu, 1 Jul 2004 12:40:35 -0500, brian <[EMAIL PROTECTED]> wrote:
>
> http://65.38.28.146/app_valetparking.c
>
> bkw
>
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Joe Antkowiak
> > Sent: Thursday, July 01, 2004 12:42 AM
> > To: [EMAIL PROTECTED]
> > Subject: Re: [Asterisk-Users] cisco phone and parked calls
> >
> > hmmm... Where can I get this?
> >
> > On Thu, 1 Jul 2004 00:29:29 -0500, Brian K. West <[EMAIL PROTECTED]> wrote:
> > >
> > > Two words...
> > >
> > > Valet Parking...
> > >
> > > bkw
> > >
> > >
> > >
> > > - Original Message -
> > > From: "Joe Antkowiak" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > Sent: Thursday, July 01, 2004 12:25 AM
> > > Subject: Re: [Asterisk-Users] cisco phone and parked calls
> > >
> > > > Does anyone have any input on this? I tried using what Craig said
> > > > above, but it didn't work...
> > > >
> > > >
> > > >
> > > > On Wed, 30 Jun 2004 13:02:46 -0400, Joe Antkowiak <[EMAIL PROTECTED]>
> > > wrote:
> > > > >
> > > > > So, in order to use the parking extension configured in
> > parking.conf,
> > > > > I have to configure that extension under a [parkedcalls] context in
> > my
> > > > > extensions.conf? I thought the call parking app was supposed to
> > take
> > > > > care of that for me?
> > > > >
> > > > >
> > > > > On Tue, 29 Jun 2004 23:49:54 +0100, Craig Waddington
> > > > > <[EMAIL PROTECTED]> wrote:
> > > > > >
> > > > > >
> > > > > > In my sip extensions context I have
> > > > > >
> > > > > > include => parkedcalls
> > > > > >
> > > > > > In extensions.conf I have
> > > > > >
> > > > > > [parkedcalls]
> > > > > > Exten => 2000,1,Answer
> > > > > >
> > > > > > In parking.conf I have the same.
> > > > > >
> > > > > > -Original Message-
> > > > > > From: [EMAIL PROTECTED]
> > > > > > [mailto:[EMAIL PROTECTED] On Behalf Of Joe
> > > > > > Antkowiak
> > > > > > Sent: 29 June 2004 22:56
> > > > > > To: [EMAIL PROTECTED]
> > > > > > Subject: [Asterisk-Users] cisco phone and parked calls
> > > > > >
> > > > > > sent this before, but it bounced back and didn't show up on the
> > list.
> > > > > > If it did get sent, I apologize.
> > > > > >
> > > > > > -- Forwarded message --
> > > > > > From: Joe Antkowiak <[EMAIL PROTECTED]>
> > > > > > Date: Tue, 29 Jun 2004 14:55:25 -0400
> > > > > > Subject: cisco phone and parked calls
> > > > > > To: [EMAIL PROTECTED]
> > > > > >
> > > > > > So, I can't figure out how to get the parkandannounce application
> > to
> > > > > > work the way I want it to... I have cisco 7960 IP phones using
> > SIP,
> > > > > > and this is what I have in my extensions.conf:
> > > > > >
> > > > > > exten =>
> > > > > >
> > > 700,1,ParkAndAnnounce(pbx-transfer:PARKED|90|SIP/${EXTEN:1}|internal,${E
> > > > > > XTEN:1},1)
> > > > > > exten => 700,2,Hangup
> > > > > >
> > > > > > and in my parking.conf:
> > > > > >
> > > > > > [general]
> > > > > > parkext => 700 ; What ext. to dial to
> > park
> > > > > > parkpos => 701-720 ; What extensions to park
> > > calls
> > > > > > on
> > > > > > context => parkedcalls
Re: [Asterisk-Users] cisco phone and parked calls
hmmm... Where can I get this?
On Thu, 1 Jul 2004 00:29:29 -0500, Brian K. West <[EMAIL PROTECTED]> wrote:
>
> Two words...
>
> Valet Parking...
>
> bkw
>
>
>
> ----- Original Message -
> From: "Joe Antkowiak" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, July 01, 2004 12:25 AM
> Subject: Re: [Asterisk-Users] cisco phone and parked calls
>
> > Does anyone have any input on this? I tried using what Craig said
> > above, but it didn't work...
> >
> >
> >
> > On Wed, 30 Jun 2004 13:02:46 -0400, Joe Antkowiak <[EMAIL PROTECTED]>
> wrote:
> > >
> > > So, in order to use the parking extension configured in parking.conf,
> > > I have to configure that extension under a [parkedcalls] context in my
> > > extensions.conf? I thought the call parking app was supposed to take
> > > care of that for me?
> > >
> > >
> > > On Tue, 29 Jun 2004 23:49:54 +0100, Craig Waddington
> > > <[EMAIL PROTECTED]> wrote:
> > > >
> > > >
> > > > In my sip extensions context I have
> > > >
> > > > include => parkedcalls
> > > >
> > > > In extensions.conf I have
> > > >
> > > > [parkedcalls]
> > > > Exten => 2000,1,Answer
> > > >
> > > > In parking.conf I have the same.
> > > >
> > > > -Original Message-
> > > > From: [EMAIL PROTECTED]
> > > > [mailto:[EMAIL PROTECTED] On Behalf Of Joe
> > > > Antkowiak
> > > > Sent: 29 June 2004 22:56
> > > > To: [EMAIL PROTECTED]
> > > > Subject: [Asterisk-Users] cisco phone and parked calls
> > > >
> > > > sent this before, but it bounced back and didn't show up on the list.
> > > > If it did get sent, I apologize.
> > > >
> > > > -- Forwarded message --
> > > > From: Joe Antkowiak <[EMAIL PROTECTED]>
> > > > Date: Tue, 29 Jun 2004 14:55:25 -0400
> > > > Subject: cisco phone and parked calls
> > > > To: [EMAIL PROTECTED]
> > > >
> > > > So, I can't figure out how to get the parkandannounce application to
> > > > work the way I want it to... I have cisco 7960 IP phones using SIP,
> > > > and this is what I have in my extensions.conf:
> > > >
> > > > exten =>
> > > >
> 700,1,ParkAndAnnounce(pbx-transfer:PARKED|90|SIP/${EXTEN:1}|internal,${E
> > > > XTEN:1},1)
> > > > exten => 700,2,Hangup
> > > >
> > > > and in my parking.conf:
> > > >
> > > > [general]
> > > > parkext => 700 ; What ext. to dial to park
> > > > parkpos => 701-720 ; What extensions to park
> calls
> > > > on
> > > > context => parkedcalls ; Which context parked calls
> are
> > > > in
> > > > parkingtime => 180
> > > >
> > > > In order for the person parking the call to hear what parked extension
> > > > the call is on, they have to do the transfer by pressing # and dialing
> > > > 700. When the user uses the transfer function on the cisco phone, it
> > > > still correctly parks the call, but never tells the person what
> > > > extension its parked on.
> > > >
> > > > Also, for some reason, I had to create that 700 extension, it always
> > > > complains that it can't find 700 when I don't do that, even though
> > > > parkedcalls is included in the internal context...
> > > >
> > > > Any suggestions?
> > > > ___
> > > > Asterisk-Users mailing list
> > > > [EMAIL PROTECTED]
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > To UNSUBSCRIBE or update options visit:
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > > > ___
> > > > Asterisk-Users mailing list
> > > > [EMAIL PROTECTED]
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > To UNSUBSCRIBE or update options visit:
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > >
> >
> >
> > --
> >
> > Joe Antkowiak
> > antkojm1 (at) gmail.com
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Joe Antkowiak
antkojm1 (at) gmail.com
___
Asterisk-Users mailing list
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Re: [Asterisk-Users] cisco phone and parked calls
Does anyone have any input on this? I tried using what Craig said
above, but it didn't work...
On Wed, 30 Jun 2004 13:02:46 -0400, Joe Antkowiak <[EMAIL PROTECTED]> wrote:
>
> So, in order to use the parking extension configured in parking.conf,
> I have to configure that extension under a [parkedcalls] context in my
> extensions.conf? I thought the call parking app was supposed to take
> care of that for me?
>
>
> On Tue, 29 Jun 2004 23:49:54 +0100, Craig Waddington
> <[EMAIL PROTECTED]> wrote:
> >
> >
> > In my sip extensions context I have
> >
> > include => parkedcalls
> >
> > In extensions.conf I have
> >
> > [parkedcalls]
> > Exten => 2000,1,Answer
> >
> > In parking.conf I have the same.
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Joe
> > Antkowiak
> > Sent: 29 June 2004 22:56
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] cisco phone and parked calls
> >
> > sent this before, but it bounced back and didn't show up on the list.
> > If it did get sent, I apologize.
> >
> > -- Forwarded message --
> > From: Joe Antkowiak <[EMAIL PROTECTED]>
> > Date: Tue, 29 Jun 2004 14:55:25 -0400
> > Subject: cisco phone and parked calls
> > To: [EMAIL PROTECTED]
> >
> > So, I can't figure out how to get the parkandannounce application to
> > work the way I want it to... I have cisco 7960 IP phones using SIP,
> > and this is what I have in my extensions.conf:
> >
> > exten =>
> > 700,1,ParkAndAnnounce(pbx-transfer:PARKED|90|SIP/${EXTEN:1}|internal,${E
> > XTEN:1},1)
> > exten => 700,2,Hangup
> >
> > and in my parking.conf:
> >
> > [general]
> > parkext => 700 ; What ext. to dial to park
> > parkpos => 701-720 ; What extensions to park calls
> > on
> > context => parkedcalls ; Which context parked calls are
> > in
> > parkingtime => 180
> >
> > In order for the person parking the call to hear what parked extension
> > the call is on, they have to do the transfer by pressing # and dialing
> > 700. When the user uses the transfer function on the cisco phone, it
> > still correctly parks the call, but never tells the person what
> > extension its parked on.
> >
> > Also, for some reason, I had to create that 700 extension, it always
> > complains that it can't find 700 when I don't do that, even though
> > parkedcalls is included in the internal context...
> >
> > Any suggestions?
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
--
Joe Antkowiak
antkojm1 (at) gmail.com
___
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Re: [Asterisk-Users] Asterisk Causing Cisco 7200 Router to Crash?
Please tell us exactly what ios version you're running. Do a "sh ver" on the cli and paste it into an e-mail. 12.xx tells us nothing =) On Wed, 30 Jun 2004 16:42:14 -0500, Harold Workman <[EMAIL PROTECTED]> wrote: > > As far as loosing the configuration...the only reason I could see that > happening is if you either are doing one of the two... not saving the > configuration...or you have the configuration register set to something like > 0x2142. look on show version for the configuration register. it should be > 0x2102. And again, i would look for tracebacks...it could either be a > memory issue or a bug in the IOS. But you will know if you get console > access to the router as u bring up the asterisk... > > > > --- > Harold Workman > CCNA, CCNP > Cytel Communications > [EMAIL PROTECTED] > Ph. 281-449-4000 x3098 > > [EMAIL PROTECTED] wrote: > > IOS version 12.xx > > > > As far as a traceback, that's going to be difficult now since we've > > removed it from our switch and brought it back here to the office for > > testing. When we test it tomorrow or later in the week, I'll see if > > it crashes again in a test setting and try to get a traceback to the > > list. > > > > It is a Cisco 7206 and all the computers connect via IP. > > > > On Yaum al-Arbi'a 12 Jumaada al-Awal 1425 05:16 pm, Harold Workman > > wrote: > >> [EMAIL PROTECTED] wrote: > >>> Hi, > >>>We are having an issue here. It seems that whenever we initialize > >>> Asterisk on our network, the router that the Asterisk server is > >>> connected to (Cisco 7200) crashes and loses it configuration. This > >>> has happended five times and each time we have tested it, it is > >>> always when Asterisk starts up. Has anyone else seen this problem? > >>> It is very odd because this is a very good router and we had the > >>> Asterisk server on an exact same router but different network before > >>> and it did not cause a crash. We have gone through two different > >>> Cisco 7200 series routers and both exhibited the same problems. Any > >>> clues? Thanks - > >>> > >>> > >>> -- > >> > >> Brian, > >> > >> > >> Did you log into the Cisco console and watch the cause of the crash? > >> You should be getting a traceback. Or after the router crashes and > >> reloads look at show version. Could you give a little better > >> explanation of your network as well. Are you conecting to the 7200 > >> via serial or ip? sip? > >> > >> > >> --- > >> Harold Workman > >> CCNA, CCNP > >> Cytel Communications > >> [EMAIL PROTECTED] > >> Ph. 281-449-4000 x3098 > >> > >> ___ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> To UNSUBSCRIBE or update options visit: > >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Joe Antkowiak antkojm1 (at) gmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco phone and parked calls
So, in order to use the parking extension configured in parking.conf,
I have to configure that extension under a [parkedcalls] context in my
extensions.conf? I thought the call parking app was supposed to take
care of that for me?
On Tue, 29 Jun 2004 23:49:54 +0100, Craig Waddington
<[EMAIL PROTECTED]> wrote:
>
>
> In my sip extensions context I have
>
> include => parkedcalls
>
> In extensions.conf I have
>
> [parkedcalls]
> Exten => 2000,1,Answer
>
> In parking.conf I have the same.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Joe
> Antkowiak
> Sent: 29 June 2004 22:56
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] cisco phone and parked calls
>
> sent this before, but it bounced back and didn't show up on the list.
> If it did get sent, I apologize.
>
> -- Forwarded message --
> From: Joe Antkowiak <[EMAIL PROTECTED]>
> Date: Tue, 29 Jun 2004 14:55:25 -0400
> Subject: cisco phone and parked calls
> To: [EMAIL PROTECTED]
>
> So, I can't figure out how to get the parkandannounce application to
> work the way I want it to... I have cisco 7960 IP phones using SIP,
> and this is what I have in my extensions.conf:
>
> exten =>
> 700,1,ParkAndAnnounce(pbx-transfer:PARKED|90|SIP/${EXTEN:1}|internal,${E
> XTEN:1},1)
> exten => 700,2,Hangup
>
> and in my parking.conf:
>
> [general]
> parkext => 700 ; What ext. to dial to park
> parkpos => 701-720 ; What extensions to park calls
> on
> context => parkedcalls ; Which context parked calls are
> in
> parkingtime => 180
>
> In order for the person parking the call to hear what parked extension
> the call is on, they have to do the transfer by pressing # and dialing
> 700. When the user uses the transfer function on the cisco phone, it
> still correctly parks the call, but never tells the person what
> extension its parked on.
>
> Also, for some reason, I had to create that 700 extension, it always
> complains that it can't find 700 when I don't do that, even though
> parkedcalls is included in the internal context...
>
> Any suggestions?
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[Asterisk-Users] cisco phone and parked calls
sent this before, but it bounced back and didn't show up on the list.
If it did get sent, I apologize.
-- Forwarded message --
From: Joe Antkowiak <[EMAIL PROTECTED]>
Date: Tue, 29 Jun 2004 14:55:25 -0400
Subject: cisco phone and parked calls
To: [EMAIL PROTECTED]
So, I can't figure out how to get the parkandannounce application to
work the way I want it to... I have cisco 7960 IP phones using SIP,
and this is what I have in my extensions.conf:
exten =>
700,1,ParkAndAnnounce(pbx-transfer:PARKED|90|SIP/${EXTEN:1}|internal,${EXTEN:1},1)
exten => 700,2,Hangup
and in my parking.conf:
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are in
parkingtime => 180
In order for the person parking the call to hear what parked extension
the call is on, they have to do the transfer by pressing # and dialing
700. When the user uses the transfer function on the cisco phone, it
still correctly parks the call, but never tells the person what
extension its parked on.
Also, for some reason, I had to create that 700 extension, it always
complains that it can't find 700 when I don't do that, even though
parkedcalls is included in the internal context...
Any suggestions?
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[Asterisk-Users] cisco phone and parked calls
So, I can't figure out how to get the parkandannounce application to
work the way I want it to... I have cisco 7960 IP phones using SIP,
and this is what I have in my extensions.conf:
exten =>
700,1,ParkAndAnnounce(pbx-transfer:PARKED|90|SIP/${EXTEN:1}|internal,${EXTEN:1},1)
exten => 700,2,Hangup
and in my parking.conf:
[general]
parkext => 700 ; What ext. to dial to park
parkpos => 701-720 ; What extensions to park calls on
context => parkedcalls ; Which context parked calls are in
parkingtime => 180
In order for the person parking the call to hear what parked extension
the call is on, they have to do the transfer by pressing # and dialing
700. When the user uses the transfer function on the cisco phone, it
still correctly parks the call, but never tells the person what
extension its parked on.
Also, for some reason, I had to create that 700 extension, it always
complains that it can't find 700 when I don't do that, even though
parkedcalls is included in the internal context...
Any suggestions?
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RE: [Asterisk-Users] Adit 600 - Getting Dial Tone
Don't forget, you also have to connect the t1 channels to the slots on the adit: connect a:1:1-8 1:1-8 connect a:1:9-16 2:1-8 connect a:1:17-24 3:1-8 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua McClintock Sent: Tuesday, June 29, 2004 12:25 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Adit 600 - Getting Dial Tone Ok so to be clear about what I need to have setup: Asterisk side: Configure T1 card like this (it's the 2nd T1 card I have in the system). span=2,1,0,esf,b8zs,yellow fxsls=25-48 Use FXO signalling in my zapata.conf [channel-bank] signalling=fxo_ls context=cb-in group=2 channel=> 25-48 Adit 600 side: -Setting slot a. set a:1 up set a:1 fdl none set a:1 lbo 1 set a:1 framing esf set a:1 id "T1 to Asterisk" set a:1 linecode b8zs set a:1 loopdetect on set a:1:1-24 side drop set a:1:1-24 type voice set a:1:1-24 signal ls -Setting slot 1. set 1:1-8 signal ls set 1:1-8 txgain -3 set 1:1-8 rxgain -6 set 1:1-8 impedance 19 -Setting slot 2. set 2:1-8 signal ls set 2:1-8 txgain -3 set 2:1-8 rxgain -6 set 2:1-8 impedance 19 -Setting slot 3. set 3:1-8 signal ls set 3:1-8 txgain -3 set 3:1-8 rxgain -6 set 3:1-8 impedance 19 On Mon, 2004-06-28 at 22:12, Andrew Kohlsmith wrote: > On Monday 28 June 2004 23:04, Joshua McClintock wrote: > > Hello, I have an Adit 600 (3 FXS cards) hooked up to a digium T1 card in my > > asterisk box. I 'connected' the slots to the a:1 T1 interfaces via the > > command line. The slots (3 fxs) are configured with 'ls' signaling. I > > configured the T1 card with the same line settings as the T1 interfaces on > > the adit and I get green lights on both the T1 card and the T1 interface on > > the adit (so they seem to be happy at that link level). I configured the > > T1 interface in the asterisk box to do 'fxsls' signaling on all 24 > > channels. > > First things first: FXS ports use FXO signalling. > > > 1. Should I be getting dial tone on my punch down block yet? I can tell > > the handset it getting power when I touch it to the correct places on the > > punch down block, but no dial tone. Does asterisk need to be aware of > > these channels before dial tone will be heard? > > You won't get dialtone until Asterisk properly configured and running; it's > what generates the dialtone. > > > 2. Did I configure the digium card correctly? Does it need to be setup as > > fxo on the asterisk side and fxs on the channel bank side? > > No, it is set up as FXS, but remember that FXS ports use FXO signalling. > > out of curiousity, why did you choose an Adit600+T1 card for three FXS ports? > The TDM400P works great for FXS. 3 slots means 24 ports. 3 cards x 8 ports per card. BTW, when you get asterisk happy and the signalling correct, for looopstart or kewlstart, all the properly configured port indicators will turn green and turn yellow/orange on pick up. If in groundstart mode they will stay off and turn green upon connection. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7940 Phones as paging system?
Got so many people asking for it, here’s what I used for the intercom announce: http://www.jsci.net/asterisk/intercom-tone.gsm It’s not great, but it does the job. Actually trying to find something better… -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Antkowiak Sent: Friday, May 07, 2004 4:30 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7940 Phones as paging system? This is what we have for this customer. They have five phones right now. Their normal extensions are 610x, but for intercom its 510x: exten => 5101,1,Dial(SIP/5101,10,tA(intercom-tone)) exten => 5101,2,Congestion If you want the wav file, let me know. If you make your own, be sure to put a 1-2 second pause in the beginning, because when the cisco answers it takes a second or to before it will send any audio to the speaker. -Original Message- From: mitchel [mailto:[EMAIL PROTECTED] Sent: Friday, May 07, 2004 4:16 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7940 Phones as paging system? Hey Joe, Could I get a sample config for playing some intro tones on the intercom? I have the same thing but nobody is using it now because they are afraid of having someone call in and "listen in" so we need some way to announce the incoming intercom call. Thanks, Mitchel Joe Antkowiak <> wrote: I am currently using 7960's with *, and line 6 is set to auto answer. Works great, customer is happy. As far as an intro-tone, you can set the dial command to play a sound (using the announce option) before the call is connected. I grabbed a simple tone wav file, and made it play that. Now, when the intercom ext is called, it plays the tone on the destination phone, and wa-la, intercom So it works. Let me know if you need sample configs. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Friday, May 07, 2004 12:57 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 Phones as paging system? Hi! > able to support intercom/paging. Having searched the archives, it > appears that this question was asked about 6 months ago, and the answer > was that the Cisco phones support this using SCCP and having one line > set to auto-answer, but at the time this was not supported in the SIP > image. Is this still the case? Dunno about Cisco, but wanted to let you know that the recent Grandstream firmware (.55 and later) now also has an auto-answer option. Still I guess I should mention that the microphone of the GS phones in speakerphone mode is far from a brilliant implementation (-> echo for the remote speaker talker, and too thin sound from the person in the room). Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs
RE: [Asterisk-Users] Cisco 7940 Phones as paging system?
This is what we have for this customer. They have five phones right now. Their normal extensions are 610x, but for intercom its 510x: exten => 5101,1,Dial(SIP/5101,10,tA(intercom-tone)) exten => 5101,2,Congestion If you want the wav file, let me know. If you make your own, be sure to put a 1-2 second pause in the beginning, because when the cisco answers it takes a second or to before it will send any audio to the speaker. -Original Message- From: mitchel [mailto:[EMAIL PROTECTED] Sent: Friday, May 07, 2004 4:16 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7940 Phones as paging system? Hey Joe, Could I get a sample config for playing some intro tones on the intercom? I have the same thing but nobody is using it now because they are afraid of having someone call in and "listen in" so we need some way to announce the incoming intercom call. Thanks, Mitchel Joe Antkowiak <> wrote: I am currently using 7960's with *, and line 6 is set to auto answer. Works great, customer is happy. As far as an intro-tone, you can set the dial command to play a sound (using the announce option) before the call is connected. I grabbed a simple tone wav file, and made it play that. Now, when the intercom ext is called, it plays the tone on the destination phone, and wa-la, intercom So it works. Let me know if you need sample configs. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Friday, May 07, 2004 12:57 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 Phones as paging system? Hi! > able to support intercom/paging. Having searched the archives, it > appears that this question was asked about 6 months ago, and the answer > was that the Cisco phones support this using SCCP and having one line > set to auto-answer, but at the time this was not supported in the SIP > image. Is this still the case? Dunno about Cisco, but wanted to let you know that the recent Grandstream firmware (.55 and later) now also has an auto-answer option. Still I guess I should mention that the microphone of the GS phones in speakerphone mode is far from a brilliant implementation (-> echo for the remote speaker talker, and too thin sound from the person in the room). Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs
RE: [Asterisk-Users] Cisco 7940 Phones as paging system?
I am currently using 7960's with *, and line 6 is set to auto answer. Works great, customer is happy. As far as an intro-tone, you can set the dial command to play a sound (using the announce option) before the call is connected. I grabbed a simple tone wav file, and made it play that. Now, when the intercom ext is called, it plays the tone on the destination phone, and wa-la, intercom So it works. Let me know if you need sample configs. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Friday, May 07, 2004 12:57 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940 Phones as paging system? Hi! > able to support intercom/paging. Having searched the archives, it > appears that this question was asked about 6 months ago, and the answer > was that the Cisco phones support this using SCCP and having one line > set to auto-answer, but at the time this was not supported in the SIP > image. Is this still the case? Dunno about Cisco, but wanted to let you know that the recent Grandstream firmware (.55 and later) now also has an auto-answer option. Still I guess I should mention that the microphone of the GS phones in speakerphone mode is far from a brilliant implementation (-> echo for the remote speaker talker, and too thin sound from the person in the room). Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Legal Interception - tapping
So, you're providing public telephone service with *? -Original Message- From: Dan Tusa <[EMAIL PROTECTED]> Date: Thu, 11 Sep 2003 20:05:56 +0100 To: [EMAIL PROTECTED] <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Legal Interception - tapping Hi, Companies that offer telephone service to the public are obliged to offer tapping to all kind of authorities. Does anyone know how to tap in Asterisk? I.e. record (or copy) a conversation based upon their telephone number? Thanks Dan _ Stay in touch with absent friends - get MSN Messenger http://www.msn.co.uk/messenger ___ 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] Is my card bad?
I have the box powered down atm, and I'm not on site, but the only thing sitting on 5 is "t1xxp" -Original Message- From: Steven Critchfield <[EMAIL PROTECTED]> Date: Thu, 11 Sep 2003 13:50:13 -0500 To: [EMAIL PROTECTED] <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Is my card bad? On Thu, 2003-09-11 at 12:21, Joe Antkowiak wrote: > Hi, > > I have a 1 port T1 card in an asus p4p800-vm board with a 2.0g Celeron, and > 512m of 266mhz ram (256 on each channel). This board has video, Ethernet, > and serial ata all on-board, I got it because of that, there wouldn't be > anything else on the pci bus that would mess with the zaptel card(s). While these devices may not be cards plugged into the PCI bus, it is most likely the same PCI bus. Do a lspci and post the results and we will be able to point out how you tell they are on the same PCI bus. > So, here's my problem. I'm running redhat 9 with kernel 2.4.20-20.9. I can > load the zaptel module fine. I can also load the wct1xxp module fine. But, > after I load the wct1xxp module, I notice that the entire box just freezes > for about 3 seconds every 30 seconds. Zttool sees the card fine, doesn't > see any problems (other than the circuit down), and there are no irq misses. > Within 3 seconds of starting *, the entire box just hangs, and never > recovers. send a copy of your /proc/interupts file too. > The only other thing on the same pci bus as the t1 card is the onboard > Ethernet controller, and nothing is sharing its irq (5). I also tried > messing with the pci latency timer. The default was 64, I tried 32, 64, 96, > and 128. I did notice though, as the numbers got smaller, the time the > system was frozen and the interval of the freezing became shorter (ie 64 was > twice as long as 32). > > Just to try it, I removed the span config for this card from zaptel.conf, > and wa-la, no more freezing, but then my card wasn't configured at all. * > would then start, but its pointless then. > > This is a new box, and I've never been able to get this to work. Any > ideas? > > -Joe > > ___ > 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is my card bad?
Hi, I have a 1 port T1 card in an asus p4p800-vm board with a 2.0g Celeron, and 512m of 266mhz ram (256 on each channel). This board has video, Ethernet, and serial ata all on-board, I got it because of that, there wouldn't be anything else on the pci bus that would mess with the zaptel card(s). So, here's my problem. I'm running redhat 9 with kernel 2.4.20-20.9. I can load the zaptel module fine. I can also load the wct1xxp module fine. But, after I load the wct1xxp module, I notice that the entire box just freezes for about 3 seconds every 30 seconds. Zttool sees the card fine, doesn't see any problems (other than the circuit down), and there are no irq misses. Within 3 seconds of starting *, the entire box just hangs, and never recovers. The only other thing on the same pci bus as the t1 card is the onboard Ethernet controller, and nothing is sharing its irq (5). I also tried messing with the pci latency timer. The default was 64, I tried 32, 64, 96, and 128. I did notice though, as the numbers got smaller, the time the system was frozen and the interval of the freezing became shorter (ie 64 was twice as long as 32). Just to try it, I removed the span config for this card from zaptel.conf, and wa-la, no more freezing, but then my card wasn't configured at all. * would then start, but its pointless then. This is a new box, and I've never been able to get this to work. Any ideas? -Joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * and Zap on AMD64/Opteron
Hi, I have a server coming, which consists of an ASUS SK8N motherboard, an AMD Opteron 1.4g cpu, and 512m of dual channel memory. If you're not familiar with this, the amd opteron is a 64bit cpu that does hardware 32bit emulation. It will be running suse enterprise server for AMD64. I will not be running * on this box, but in the future it may be a good performer. So, before it goes into production, I have some zaptel cards I can put in the box, and I can provide access to it via dsl, if anyone is interested getting the zaptel modules working on this box (from what I'm told, the modules won't work with the kernel for this box). Please reply off-list if you are interested. Apologies if this has already been discussed, I searched the archives and couldn't find anything. -Joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Someone used ADIT 600 Channel Bank.
They work great, I have 3 up and running all with mixed fxo-8 and fxs-8 cards. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Tinchev Sent: Tuesday, August 05, 2003 4:56 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Someone used ADIT 600 Channel Bank. I must buy channel banks for ~120 lines. After some googling and ebay searching i see that ADIT 600 has exelant proce//... for me. Just wandering how it works with asterisk. ___ 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] Asterisk and AT&T 964 phones...
What kind of features do you want to work? If you want all the phone-proprietary features (intercom, transfer) to work, you have to use the same analog lines in the same order on every one. Asterisk can do most of the features though, but intercom would require a little more equipment (fm broadcaster, little radios at every desk) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Hale Sent: Wednesday, August 13, 2003 11:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and AT&T 964 phones... Anyone know if the AT&T 964/954 series phones have any issues with Asterisk? We have 5 phones and would like to reuse them if possible. Any restrictions or clunky workarounds needed? Thanks in advance, Chris ___ 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] FXO mode
I've had this happen with the x100p and analog phones as well... When I moved to a t1 and a channel bank, the problem never happened again... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Meyers Sent: Wednesday, August 13, 2003 12:04 PM To: Asterisk List Subject: Re: [Asterisk-Users] FXO mode On Wed, 2003-08-13 at 09:46, Dave Cotton wrote: > I've had a few problems with my system holding the line after a call has > been made, just now I rebooted and noticed the following in > /var/log/messages When you say "holding the line", do you mean that asterisk still believes a channel is in use even after you hang up? If so, I've seen the same thing happen several times with the X100P. If I do "show channels" it will show one of my SIP phones connected to one of the outside lines, but if I check that SIP phone, it is not in use, and there is no way to re-activate the channel from the SIP phone. Running "soft hangup " will hangup the channel (you don't need to reboot). I'm not entirely sure what causes it. So far, I've only seen it happen from 2 of our 9 SIP phones, but they're the ones most often on the phone. It always involves an outside line, so I believe the X100P is the problem, but I can't be sure. What other information can I gather to pinpoint the problem? ___ 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] bugs.digium.com
Hehe, that's slightly ironic... =P -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sharp Sent: Monday, August 04, 2003 3:16 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] bugs.digium.com Is anyone else having trouble accessing it with something besides IE on a Windows box? Opera on Mac/FreeBSD/Linux just hangs at the login page, IE on Mac and Netscape on Solaris & Linux explode when loading login_page.php. ___ 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] SCO/Linux concerns
What's your concern with it? If any of SCO code made it into GNU stuff, it will be removed and rewritten in a short time anyway... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ajit M Kallingal Sent: Wednesday, July 30, 2003 7:08 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SCO/Linux concerns Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users digest, Vol 1 #935 - 14 msgs > Send Asterisk-Users mailing list submissions to > [EMAIL PROTECTED] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > >1. RE: voicemail file access problems (Todd Lieberman) >2. sip -> h323 -> ptsn (Brian West) >3. RE: voicemail file access problems (Todd Lieberman) >4. Re: voicemail file access problems (Tilghman Lesher) >5. Re: sip -> h323 -> ptsn (Patrick) >6. RE: voicemail file access problems (Patrick) >7. Re: sip -> h323 -> ptsn (Brian West) >8. Re: sip -> h323 -> ptsn (Patrick) >9. X100P and incoming Context + CDR? (Darren Smith) > 10. Re: CVS Problem? (Kyle Hagan) > 11. Re: sip -> h323 -> ptsn (Eric Wieling) > 12. %unsuscribe (Carlos Crembil) > 13. Re: SetCIDName (Siggi Langauf) > 14. RE: X-Lite and Call transfer using Asterisk (Stuart Hirst) > > --__--__-- > > Message: 1 > From: "Todd Lieberman" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 15:49:56 -0400 > Reply-To: [EMAIL PROTECTED] > > I did the chown and now I get > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gid script > is writable by world., referer: > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Paulo > Mannheimer > Sent: Wednesday, July 30, 2003 3:23 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] voicemail file access problems > > > Thanks! > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman > Lesher > Sent: July 30, 2003 4:06 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] voicemail file access problems > > On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > > Hi folks, > > > > I'm having problems accessing my voicemail files through the web > > interface. > > > > I remember that this was discussed on the list, and it seems to be > > a permission problem, but I couldn't find any answer by searching > > the archives. > > > > Any hint? > > chown root vmail.cgi > chmod u+s vmail.cgi > > -Tilghman > > ___ > 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 > > > --__--__-- > > Message: 2 > Date: Wed, 30 Jul 2003 15:08:53 -0500 (CDT) > From: Brian West <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: [EMAIL PROTECTED] > > I have this setup: > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > Sip phones are setup for out of band dtmf > > but the h323 gateway is inband. Is their a way to pass the digits from > the sip phones to the ptsn via the h323 gateway? > > bkw > > --__--__-- > > Message: 3 > From: "Todd Lieberman" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 16:12:59 -0400 > Reply-To: [EMAIL PROTECTED] > > I fixed my own problem. I had just did chmod 755 vmail.cgi and it worked. > > you still need to make sure nobody has read/write permission on > /var/spool/asterisk/vm/$MBOX > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Todd > Lieberman > Sent: Wednesday, July 30, 2003 3:50 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] voicemail file access problems > > > I did the chown and now I get > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gid script > is writable by world., referer: > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Paulo > Ma
[Asterisk-Users] IRQ Misses?
Hi, One of my pbx's seems to be having some new issues. crackling interference on the zap channels running through the channel bank, and I noticed that these happen when I hit an "irq miss" in zttool: Current Alarms: No alarms. Sync Source:Digium Wildcard T100P T1/PRI C IRQ Misses: 82695 Bipolar Viol: 0 Tx/Rx Levels: 0/8 3 Total/Conf/Act: 24/ 24/ 0 But I have no cards in conflict... This started immediately after I did a reboot... Any ideas? [EMAIL PROTECTED] proc]# cat ioports -001f : dma1 0020-003f : pic1 0040-005f : timer 0060-006f : keyboard 0070-007f : rtc 0080-008f : dma page reg 00a0-00bf : pic2 00c0-00df : dma2 00f0-00ff : fpu 01f0-01f7 : ide0 03c0-03df : vga+ 03f6-03f6 : ide0 0cf8-0cff : PCI conf1 7400-740f : PCI device 1095:3112 (CMD Technology Inc) 7800-7803 : PCI device 1095:3112 (CMD Technology Inc) 8000-8007 : PCI device 1095:3112 (CMD Technology Inc) 8400-8403 : PCI device 1095:3112 (CMD Technology Inc) 8800-8807 : PCI device 1095:3112 (CMD Technology Inc) 9000-90ff : Linksys Network Everywhere Fast Ethernet 10/100 model NC100 9000-90ff : tulip 9400-94ff : Tiger Jet Network Inc. Model 300 128k b800-b80f : VIA Technologies, Inc. VT82C586B PIPC Bus Master IDE b800-b807 : ide0 b808-b80f : ide1 d000-dfff : PCI Bus #01 d800-d8ff : ATI Technologies Inc Radeon VE QY e400-e4ff : VIA Technologies, Inc. VT82C686 [Apollo Super ACPI] e800-e80f : VIA Technologies, Inc. VT82C686 [Apollo Super ACPI] [EMAIL PROTECTED] proc]# cat iomem -0009f7ff : System RAM 0009f800-0009 : reserved 000a-000b : Video RAM area 000c-000c7fff : Video ROM 000cc000-000d07ff : Extension ROM 000f-000f : System ROM 0010-17ffbfff : System RAM 0010-0024d583 : Kernel code 0024d584-00347a43 : Kernel data 17ffc000-17ffefff : ACPI Tables 17fff000-17ff : ACPI Non-volatile Storage d580-d58001ff : PCI device 1095:3112 (CMD Technology Inc) d580-d58001ff : SiI3112 Serial ATA d600-d60003ff : Linksys Network Everywhere Fast Ethernet 10/100 model NC100 d600-d60003ff : tulip d680-d6800fff : Tiger Jet Network Inc. Model 300 128k d700-d7df : PCI Bus #01 d700-d700 : ATI Technologies Inc Radeon VE QY d7f0-e3ff : PCI Bus #01 d800-dfff : ATI Technologies Inc Radeon VE QY e400-e7ff : VIA Technologies, Inc. VT82C693A/694x [Apollo PRO133x] - : reserved [EMAIL PROTECTED] proc]# cat interrupts CPU0 0:5406112 XT-PIC timer 1: 2153 XT-PIC keyboard 2: 0 XT-PIC cascade 4: 167078 XT-PIC ide2 5: 5180 XT-PIC eth0 8: 1 XT-PIC rtc 10: 53888794 XT-PIC t1xxp 12:680 XT-PIC PS/2 Mouse 14: 10 XT-PIC ide0 NMI: 0 ERR: 0 [EMAIL PROTECTED] proc]# cat pci PCI devices found: Bus 0, device 0, function 0: Host bridge: VIA Technologies, Inc. VT82C693A/694x [Apollo PRO133x] (rev 2). Prefetchable 32 bit memory at 0xe400 [0xe7ff]. Bus 0, device 1, function 0: PCI bridge: VIA Technologies, Inc. VT82C598/694x [Apollo MVP3/Pro133x AGP] (rev 0). Master Capable. No bursts. Min Gnt=8. Bus 0, device 4, function 0: ISA bridge: VIA Technologies, Inc. VT82C686 [Apollo Super South] (rev 34). Bus 0, device 4, function 1: IDE interface: VIA Technologies, Inc. VT82C586B PIPC Bus Master IDE (rev 16). Master Capable. Latency=32. I/O at 0xb800 [0xb80f]. Bus 0, device 4, function 4: Host bridge: VIA Technologies, Inc. VT82C686 [Apollo Super ACPI] (rev 48). IRQ 9. Bus 0, device 10, function 0: Network controller: Tiger Jet Network Inc. Model 300 128k (rev 0). IRQ 10. Master Capable. Latency=32. Min Gnt=1.Max Lat=128. I/O at 0x9400 [0x94ff]. Non-prefetchable 32 bit memory at 0xd680 [0xd6800fff]. Bus 0, device 12, function 0: Ethernet controller: Linksys Network Everywhere Fast Ethernet 10/100 model NC100 (rev 17). IRQ 5. Master Capable. Latency=32. Min Gnt=64.Max Lat=128. I/O at 0x9000 [0x90ff]. Non-prefetchable 32 bit memory at 0xd600 [0xd60003ff]. Bus 0, device 13, function 0: Unknown mass storage controller: PCI device 1095:3112 (CMD Technology Inc) (rev 2). IRQ 4. Master Capable. Latency=32. I/O at 0x8800 [0x8807]. I/O at 0x8400 [0x8403]. I/O at 0x8000 [0x8007]. I/O at 0x7800 [0x7803]. I/O at 0x7400 [0x740f]. Non-prefetchable 32 bit memory at 0xd580 [0xd58001ff]. Bus 1, device 0, function 0: VGA compatible controller: ATI Technologies Inc Radeon VE QY (rev 0). Master Capable. Latency=64. Min Gnt=8. Prefetchable 32 bit memory at 0xd800 [0xdfff]. I/O at 0xd800 [0xd8ff]. Non-prefetchable 32 bit memory at 0xd700 [0xd700]. [EMAI
RE: [Asterisk-Users] Supplementing Current phone system
Yep, works great -Original Message- From: Ben Turner <[EMAIL PROTECTED]> Date: Tue, 22 Jul 2003 16:37:21 -0700 To: '[EMAIL PROTECTED]' <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Supplementing Current phone system Has anyone used * in conjunction with an existing phone system to add certain features or to expand a department? Currently we have an existing nortel phone system. We would like to have the * server stand behind it and just handle VoIP services for our remote sales and technical people. We would like them to act as normal extensions (being able to dial them by extension from the existing phone system and have them dial extensions on the existing phone sytem). >From the documentation, it appears to be possible, but I have yet to find more information on how to do it. Has anyone done this or something similar? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ideal Prompt Recording Setup?
I've always just recorded messages into a voicemail box and copied the .gsm files to the sounds dir under the appropriate name... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, July 22, 2003 3:24 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Ideal Prompt Recording Setup? What have people found to be the ideal setup for recording asterisk prompts? I'm looking for both the ideal application to record them in, the ideal format, as well as hardware (do I need a fancy studio mic or will a headset mic work?). Thanks, Justin ___ 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] Asterisk and FWD
Did you move your box behind some type of nat device?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan
Sent: Tuesday, July 22, 2003 10:06 AM
To: Asterisk Users
Subject: [Asterisk-Users] Asterisk and FWD
Hi,
The line
register => fwdnr:[EMAIL PROTECTED]/101
does not work anymore in sip.con file.
I get the following error in the Asterisk console:
-- Got SIP response 479 "We dont accept private IP contacts. please Set
your external IP" back from 192.246.69.223
The sane error when I try to call a FWD extension defined like that:
Something changed in the mean time?
exten => _X,1,SetCallerID(${FWDUSERID})
exten => _X,2,SetCIDName(${FWDUSERNAME})
exten => _X,3,Dial(SIP/[EMAIL PROTECTED])
exten => _X,4,Hangup
exten => _X,104,Hangup
Thanks,
Dan
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RE: [Asterisk-Users] H3500CW recommendation
Also check out www.wantphones.com. Good phones, good prices. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, July 18, 2003 3:57 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] H3500CW recommendation On Fri, 2003-07-18 at 14:41, Howard White wrote: > So Steven, > > I looked at the page you listed for the AT&T 957 (by the way, the DG > stands for Dove Grey). I also traversed to find their > page hoping for more info about the 3 line, 15 character display. Nope, > none, nada, rien, zilch... > > So are the three lines addressable (as in ADSI) or are they just for > decoration. I know, I know; even on the AT&T site, they sell for $USD30 > plus shipping. No they aren't ADSI accessible. The first line is time and date, plus and indicator for whether the phone is off hook or not. From looking at an angle, I see there is what appears to be a message waiting icon too, but I don't remember seeing it set. Line 2 is number called/callerid. Line 3 is also part of the caller id, or while not on a call it displays how many calls are in memory, and then number of them that are new. BTW, mine are attached to a zhone too but only for the FXS lines. > > Besides, we have a zhone and hence, no caller ID. > > Howard White -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] serious dtmf recognition problem.
Other auto dialers that I tested connected to the same channel on the same channel
bank work fine, only because they wait longer before dialing after verifying they have
a dialtone.
But, in any case, this is a CAC adit 600 with an FXS-8 card.
-Original Message-
From: Steven Critchfield <[EMAIL PROTECTED]>
Date: 18 Jul 2003 02:55:44 -0500
To: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] serious dtmf recognition problem.
I haven't had any problem with DTMF detection, so lets also look at what
channel bank are you using.
A crude hack is to find out where the cc machine is dialing and just
assume any call from that line of the channel bank is only going to dial
the cc machine number.
On Fri, 2003-07-18 at 02:46, Joe Antkowiak wrote:
> Unfortunately, I have no way of changing the dialed number on the credit
> card machine =/
>
> Is there any way I can get asterisk to recognize those first few digits?
> Something I could modify in the source?
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood
> Sent: Friday, July 18, 2003 12:32 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] serious dtmf recognition problem.
>
> Hi Joe,
>
> Most auto-dialers will accept commas in the dial string, and insert
> delays where they occur. Will that work for you? Its normally used to
> insert a delay after a 9 on a PBX, to get a stable outside line before
> further dialing.
>
> Regards,
> Steve
>
>
> Joe Antkowiak wrote:
>
> > Hi,
> >
> > I am using a channel bank and zaptel hardware. I have a credit card
> > machine on one of the channels that appears to be dialing "too soon"
> > for asterisk, every complete number recognized by asterisk is missing
> > the first 1-4 numbers. This is a serious problem for me, anyone have
> > any ideas on whats going on? The pstn picks up on the dtmf tones just
> > fine.
> >
> > I was able to get it to work 50% of the time by adding:
> >
> > exten => _8XXNXX,1,Dial(Zap/g2/1${EXTEN})
> >
> > but that's really ugly.
> >
>
>
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--
Steven Critchfield <[EMAIL PROTECTED]>
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RE: [Asterisk-Users] serious dtmf recognition problem.
Unfortunately, I have no way of changing the dialed number on the credit
card machine =/
Is there any way I can get asterisk to recognize those first few digits?
Something I could modify in the source?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood
Sent: Friday, July 18, 2003 12:32 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] serious dtmf recognition problem.
Hi Joe,
Most auto-dialers will accept commas in the dial string, and insert
delays where they occur. Will that work for you? Its normally used to
insert a delay after a 9 on a PBX, to get a stable outside line before
further dialing.
Regards,
Steve
Joe Antkowiak wrote:
> Hi,
>
> I am using a channel bank and zaptel hardware. I have a credit card
> machine on one of the channels that appears to be dialing "too soon"
> for asterisk, every complete number recognized by asterisk is missing
> the first 1-4 numbers. This is a serious problem for me, anyone have
> any ideas on whats going on? The pstn picks up on the dtmf tones just
> fine.
>
> I was able to get it to work 50% of the time by adding:
>
> exten => _8XXNXX,1,Dial(Zap/g2/1${EXTEN})
>
> but that's really ugly.
>
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[Asterisk-Users] serious dtmf recognition problem.
Hi,
I am using a channel bank and zaptel hardware. I have
a credit card machine on one of the channels that appears to be dialing “too
soon” for asterisk, every complete number recognized by asterisk is
missing the first 1-4 numbers. This is a serious problem for me, anyone
have any ideas on whats going on? The pstn picks up on the dtmf tones
just fine…
I was able to get it to work 50% of the time by adding:
exten => _8XXNXX,1,Dial(Zap/g2/1${EXTEN})
but that’s really ugly.
Zapata.conf:
;CC Machine
context=cc-out
signalling=fxo_ks
usecallerid=yes
hidecallerid=no
callwaiting=no
callwaitingcallerid=no
threewaycalling=no
transfer=no
cancallforward=no
callreturn=no
echocancel=no
echocancelwhenbridged=no
relaxdtmf=no
rxgain=6.0
txgain=0.0
group=10
callgroup=10
pickupgroup=10
immediate=no
amaflags=documentation
accountcode=cc-outbound
adsi=no
busydetect=no
callprogress=no
callerid="CC Machine"<>
channel => 22
RE: [Asterisk-Users] PCI Master Abort
You can force IRQs in your BIOS config, I would set each card to its own IRQ that doesn't get shared with anything else. Disable your serial and parallel ports if you aren't using them and use 3,4,5,7 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Beaumont Sent: Monday, July 07, 2003 4:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PCI Master Abort I am always getting multiple PCI Master Abort messages when I try to plug in a second TDM400P. I have asked this question before, but I nothing really solved my problem and I just put it on the back burner for a while. I am at a point where this is a crucial issue. I have read that the Zaptel devices share an IRQ, is this causing the problem? Is there a way that I can manually change the IRQs of the devices? By the way, my hardware situation is as follows: 2 X100P 2 TDM400P Any help is always appreciated. ___ 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] Newbie Doubts
Because a channel bank converts a DS1 into 24 DS0s and vice versa. Your channel bank may have an Ethernet port on it, but it is just for management. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Saar Gemignani Sent: Monday, July 07, 2003 12:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie Doubts Can´t I connect to the channel bank using ethernet? Why? - Original Message - From: "Martin Pycko" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, July 07, 2003 11:52 AM Subject: Re: [Asterisk-Users] Newbie Doubts > You plug a channel bank to a T1 in your PC connected either over T100P or > T400P. > > regards > Martin > > On Mon, 7 Jul 2003, Ricardo Saar Gemignani wrote: > > > Hello everybody > > > > My doubt is about configuration. Can I use a channel bank like zplex-10 or adtran, plug on it an T1, 24 POTS, an ethernet cable connected at the computer with Asterisk installed? Will it work? Will asterisk be able to control the system? Receive a call and work with all its functions(transfer, conference, voice mail)? > > > > Thanks in advance, > > Ricardo Saar Gemignani > > > > ___ > 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] BIG problem with multiple rings before pickup
How do you tell asterisk to detect for fax tones? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Wednesday, July 02, 2003 2:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] BIG problem with multiple rings before pickup On Wed, 2003-07-02 at 13:34, Jim Archer wrote: > Hi All... > > I have a maddening problem... > > I have Asterisk configured to pick up a line after 4 rings. I do this to > allow my fax machine to pick up a particular distinctive ring pattern, so I > don't have to pay for a dedicated fax line. > > If someone calls the line, lets it ring 3 times and then hangs up, Asterisk > answers the line, and holds it off hook forever, constantly playing the > prompts. > > My hardware is 2 X100P cards. > > Any ideas? Get a TDM10B, cancel your distinctive ring, and let asterisk answer immediately and detect fax tones and forward it to your fax machine. -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] hunt group
I was wondering if someone could point me in the right direction with this... I have 3 4-line phones, all connected to a * box via a T100P and channel bank. I essentially want to create a "hunt group" for each phone, and then as calls come in, Dial to include the 3 different hunt groups, so that if one user is on line 1 of their phone, a new incoming call will go to line 2 of their phone, but line 1 of the others. Can I do this with hints? Thanks. -Joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CAC Access Bank
There isn't much to set... Just use B8ZS and ESF, clock source line or internal, just be sure to use the opposite on the * box. Also, you know you'll need a T1 crossover cable right? If you need to know the pins, let me know, or just search the list, its been posted. I would go ahead and start setting it up, and if you have any problems, just ask the list again. Every experience I've had with FXS on the AB1s has been very simple. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of denon Sent: Wednesday, June 18, 2003 6:01 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] CAC Access Bank They are AB1s. I was hoping to put FXOs in em, but I'm hearing a lot of that, so I guess I won't. Do you have any information on how the DIPs are set on a unit working with *? Thanks, -d At 04:11 PM 6/18/2003 -0400, you wrote: >Its pretty straight forward... Are they the AB1s or AB2s? The AB1s don't >have config, just switches. > >I wouldn't trust the AB product FXOs, I heard they don't work very well. If >you want to do a mix of fxo and fxs, the CAC Adit 600's are very good, and I >have some FXO-8 cards for them that I need to get rid of - e-mail me off the >list if you're interested. > >-Joe > >-Original Message- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] On Behalf Of denon >Sent: Wednesday, June 18, 2003 4:05 PM >To: [EMAIL PROTECTED] >Subject: [Asterisk-Users] CAC Access Bank > >I just picked up a couple CAC Access Bank 1s loaded with FXS that should be >arriving shortly. Does anyone have one that they use with Asterisk? If >so, would you be willing to shoot me a note with your current configs? I'm >not very familiar with CAC/etc, and it would save me countless hours of >muddling through .. :) > >Any others tips you can give me on these banks would be very >appreciated. Also, any cheap source for FXOs on them? I'm told they're >hard to come by .. > >Thanks again, > >-d > >___ >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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CAC Access Bank
Its pretty straight forward... Are they the AB1s or AB2s? The AB1s don't have config, just switches. I wouldn't trust the AB product FXOs, I heard they don't work very well. If you want to do a mix of fxo and fxs, the CAC Adit 600's are very good, and I have some FXO-8 cards for them that I need to get rid of - e-mail me off the list if you're interested. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of denon Sent: Wednesday, June 18, 2003 4:05 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] CAC Access Bank I just picked up a couple CAC Access Bank 1s loaded with FXS that should be arriving shortly. Does anyone have one that they use with Asterisk? If so, would you be willing to shoot me a note with your current configs? I'm not very familiar with CAC/etc, and it would save me countless hours of muddling through .. :) Any others tips you can give me on these banks would be very appreciated. Also, any cheap source for FXOs on them? I'm told they're hard to come by .. Thanks again, -d ___ 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] VoIP Provider
Iconnecthere seems to have better rates... -Original Message- From: Martin Dommermuth <[EMAIL PROTECTED]> Date: Thu, 12 Jun 2003 19:48:43 +0200 (MEST) To: [EMAIL PROTECTED] <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] VoIP Provider Hi, * Erik Lagerway wrote/schrieb: > > There is a provider in the US -> www.AddaLine.com, who just launched a > SIP> service with some great rates for North America > > I have been using their service for months and I am extremely happy with the > service. looks like Germany is again laggin behind all others in the communication field. Or I asked at the wrong place. There might not be to many people from Germany in this list. Anyway, thanks for the answer. CU MartinD: -- +++ GMX - Mail, Messaging & more http://www.gmx.net +++ Bitte lächeln! Fotogalerie online mit GMX ohne eigene Homepage! ___ 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] Voicemail notification
it should be added to zapata.conf, and you can specify multiple mailboxes separated by , On Wed, 2003-06-11 at 20:10, Andy Powell wrote: > I'd like to use either the message waiting light or stutter tone but on searching > the archives I found conflicting answers. > > Everyone seems to agree that you should add > > mainbox= > > but some people are saying that it should be added to zapata.conf and > others are saying zaptel.conf > > Can someone who has it working clarify this? If it is zaptel.conf can somone > supply a sample.. my zaptel.conf file only consists of > > fxsks=1 > fxoks=2 > fxoks=3 > loadzone=uk > defaultzone=uk > > and that's it... > > Thanks in advance > > Andy > > > > *** REPLY SEPARATOR *** > > On 11/06/2003 at 16:53 Steven Critchfield wrote: > > >On Wed, 2003-06-11 at 15:16, Derek Beaumont wrote: > >> Besides email notification, is there another way to get asterisk notify > >> the user that they have a message? > >> > >> Example: Some analog phones have a blinking light that lets the user > >> know that they have a voicemail message. > >> Is asterisk capable of doing this? > > > >Yes, and I know it works on Sip and Zap channels. Check archive for MWI, > >for Message waiting indicator. > >-- > >Steven Critchfield <[EMAIL PROTECTED]> > > > >___ > >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] Valiant Comms VCL 30 Channel bank + Digium E100P
Also, are all the fxo channels "connected" to the t1 channels on the channel bank? I believe you have to map them just like on the CAC Adits... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Pycko Sent: Thursday, June 05, 2003 10:40 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Valiant Comms VCL 30 Channel bank + Digium E100P Do you really have the channels in asterisk ? "zap show channels" Is the alarm on the E1 circuit ? Martin On Thu, 5 Jun 2003, Jay Banda wrote: > Hello All. > > Does anyone have experience with the Valiant Comms vcl30 channel > and the Digium E100P in asterisk ? We have the vcl30 channel bank, > loaded with FXO interfaces. We have set up * for fxs in zaptel.conf > and in zapata.conf, but are not able to get any incoming calls. > > The vcl fxs interfaces show that they are ringing ( incoming call from > PSTN ) , but the * does not answer ( or am I missing something , I think > the E100P card is supposed to indicate that there is an incoming call ?) > > If anyone is able to point me in a remotely correct direction, I will be > eternally grateful :) > > Kind regards to all > > Jay Banda > Snr Network Eng > CopperNET Solutions > > ___ > 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] CAC ADIT600 / T400 config
Ah, ok, yes I was misunderstanding your config. >From the sample zaptel.conf: # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of "1". For a secondary, use "2", and so on. # To not use this as a sync source, just use "0" So using 1,1 means use span 1 as the primary sync source. 0 would mean don't use it for sync... Right? My link wouldn't come up when set to 0. On Sun, 2003-06-01 at 00:34, TC wrote: > >> But now i dont get any dial tone when i pick up a hand set, the adit 600 > >> recognizes the off hook & goes amber but * does not see it off hook > >> And it has registered the channels > >> == Parsing '/etc/asterisk/zapata.conf': Found > >> -- Registered channel 1, FXO Kewlstart signalling > >> > >I think you may have fxs and fxo reversed. \ > Nope, I think maybe there is a misunderstanding of my config > I have 3x8 port FXS Cards slots 1,2,3 and 3x8 port fxo cards slots 4-6, > what i show above is the signaling to channel 1 on the first fxs card > so i am signaling using fxo kewlstart > > >for the channels bound to > >the fxo-8 boards, asterisk has to be set to fxs signalling, and vice > >versa. asterisk needs to be told what signalling to PUT on the line, as > >opposed to the signalling the adits expect to GET from the line. > Got that right :) > >I have 1 fxo-8 board (for the inbound pots lines) and 5 fxs-8 boards (for > the > >phones), and asterisk has channels 1-8 fxsks and 9-24 fxoks. > Sure if your using span 1 from the frist t400 card in * > and you are only using the first 3 service slots on interface T1-1, > with 1 fxo, & 2 fxs card. > If you have 6 slots used then you have a possible 48 channels > on the ADIT 600, the first 24 are on interface T1-1 and rest on T1-2. > The break out is slots 1-3 are channels 1-24, and > slots 4-6 are channels 25-48. > > >>After mucking around with it a bit, I was able to get it to sync with > the > > >>channel bank as the clock source for the line: > > >>jsci-cbank1> show clock > > >> Primary Master Transmit Clock Source:Internal > > >> Secondary Transmit Clock Source: Slot A DS1 2 > > So this means you have the adit supplying timing to asterisk t1 spans > > so your zaptel looks like this > > span=1,0,0,esf,b8zs > > span=2,0,0,esf,b8zs > > span=3,0,0,esf,b8zs > > span=4,0,0,esf,b8zs > > fxoks=1-24 > > No, I have: > fxsks=1-8 > fxoks=9-24 > span=1,1,0,esf,b8zs > OK my understanding here is that > "Primary Master Transmit Clock Source:Internal" > means that the primary timing source for the channel bank is its own > internal clock therefore > asterisk should *NOT* be the Primary Timing source for this span because the > ADIT 600 > is already the Primary Timing Source from its own internal clock, so timing > on this span > as specified in zaptel.conf should not be Primary or Secondary > thus span=1,0,0,esf,b8zs *NOT* span=1,1,0,esf,b8zs > > Anyone else please pipe up here that knows this stuff cold :) > > sure, here ya go: > jsci-cbank1> print config > Thxs that quite old that version of 1.5 mine is 3.01 > hmm that config is mostly similar, > think i am gonna conclude I have a completely DOA t400 > > > > > ___ > 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] musiconhold.conf problems
no. Use: default => mp3:/var/lib/asterisk/mohmp3 and make track01.mp3 the only file in that directory. On Sat, 2016-05-21 at 04:42, Randal Law wrote: > Hi All, > > For musiconhold.conf file, how can I play the track01.mp3 as default music > when it is on hold? > > Below conf is it correct? > > [classes] > default => mp3:/var/lib/asterisk/mohmp3/track01.mp3 > > Please help! > > Thanks, > Randal > > ___ > 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] CAC ADIT600 / T400 config
On Sat, 2003-05-31 at 21:34, TC wrote: > Yea looks like i have a issue with t400 spans 2 & 4 :( > I now can get 2 pretty green lights on span 1 and span 3 & > on the adit 600 on T1-1 and T1-2 ..and the 2 fxs/3fxo card are all green > So i plow'd ahead on possible shakey ground ... > Eep... Bad card maybe? > But now i dont get any dial tone when i pick up a hand set, the adit 600 > recognizes the off hook & goes amber but * does not see it off hook > And it has registered the channels > == Parsing '/etc/asterisk/zapata.conf': Found > -- Registered channel 1, FXO Kewlstart signalling > > just curious what signaling did you need on the ports for * to be a happy > camper > with adit fxs cards > I think you may have fxs and fxo reversed. for the channels bound to the fxo-8 boards, asterisk has to be set to fxs signalling, and vice versa. asterisk needs to be told what signalling to PUT on the line, as opposed to the signalling the adits expect to GET from the line. I have 1 fxo-8 board (for the inbound pots lines) and 5 fxs-8 boards (for the phones), and asterisk has channels 1-8 fxsks and 9-24 fxoks. > >>After mucking around with it a bit, I was able to get it to sync with the > >>channel bank as the clock source for the line: > >>jsci-cbank1> show clock > >> Primary Master Transmit Clock Source:Internal > >> Secondary Transmit Clock Source: Slot A DS1 2 > So this means you have the adit suppling timing to asterisk t1 spans > so your zaptel looks like this > span=1,0,0,esf,b8zs > span=2,0,0,esf,b8zs > span=3,0,0,esf,b8zs > span=4,0,0,esf,b8zs > fxoks=1-24 No, I have: fxsks=1-8 fxoks=9-24 span=1,1,0,esf,b8zs > > > Could post the output of > print config > > from the adit 600 cli sure, here ya go: jsci-cbank1> print config - -Adit 600 configuration file -Created on 01/09/1999 at 00:48:13 for antkojm1 -This file is valid for the following configuration only: - -CardType - -SLOT AT1x2 SW Version: 1.5.0 -SLOT 1FXOx8 -SLOT 2FXSx8 -SLOT 3FXSx8 -SLOT 4FXSx8 -SLOT 5FXSx8 -SLOT 6FXSx8 -NOTES: - -Turning off verification messages. set verification off -Setting local off. set local off -Disconnecting all connections. disconnect a disconnect 1 disconnect 2 disconnect 3 disconnect 4 disconnect 5 disconnect 6 -Setting IP addresses. set ethernet ip address 10.109.97.62 255.255.255.0 set ip gateway 10.109.97.1 -Setting the SNMP MIB-II System Group objects. -Setting slot a. set a:1 up set a:1 fdl none set a:1 lbo 1 set a:1 framing esf set a:1 id "asterisk-internal trunk" set a:1 linecode b8zs set a:1 loopdetect on set a:1 threshold min15 uas default set a:1 threshold min15 ses default set a:1 threshold min15 es default set a:1 threshold min15 sefs default set a:1 threshold min15 les default set a:1 threshold min15 css default set a:1 threshold min15 bes default set a:1 threshold min15 dm default set a:1 threshold min15 lcv default set a:1 threshold min15 pcv default set a:1 threshold day uas default set a:1 threshold day ses default set a:1 threshold day es default set a:1 threshold day sefs default set a:1 threshold day les default set a:1 threshold day css default set a:1 threshold day bes default set a:1 threshold day dm default set a:1 threshold day lcv default set a:1 threshold day pcv default set a:1:1-24 signal ls set a:1:1-24 type voice set a:2 down set a:2 fdl none set a:2 lbo 1 set a:2 framing esf set a:2 id "CAC DS1# A:2" set a:2 linecode b8zs set a:2 loopdetect on set a:2 threshold min15 uas default set a:2 threshold min15 ses default set a:2 threshold min15 es default set a:2 threshold min15 sefs default set a:2 threshold min15 les default set a:2 threshold min15 css default set a:2 threshold min15 bes default set a:2 threshold min15 dm default set a:2 threshold min15 lcv default set a:2 threshold min15 pcv default set a:2 threshold day uas default set a:2 threshold day ses default set a:2 threshold day es default set a:2 threshold day sefs default set a:2 threshold day les default set a:2 threshold day css default set a:2 threshold day bes default set a:2 threshold day dm default set a:2 threshold day lcv default set a:2 threshold day pcv default set a:2:1-24 signal ls set a:2:1-24 type voice -Setting slot 1. set 1:1-8 signal lscpd set 1:1-8 txgain 0 set 1:1-8 rxgain 3 -Setting slot 2. set 2:1-8 signal ls set 2:1-8 txgain 0 set 2:1-8 rxgain 0 set 2:1-8 linelength short -Setting slot 3. set 3:1-8 signal ls set 3:1-8 txgain 0 set 3:1-8 rxgain 0 set 3:1-8 linelength short -Setting slot 4. set 4:1-8 signal ls set 4:1-8 txgain -3 set 4:1-8 rxgain -6 set 4:1-8 linelength short -Setting slot 5. set 5:1-8 signal ls set 5:1-8 txgain -3 set 5:1-8 rxgain -6 set 5:1-8 linelength short -Setting slot 6. set 6:1-8 signal ls set 6:1-8 txgain -3 set 6:1-8 rxgain -6 set
RE: [Asterisk-Users] CAC ADIT600 / T400 config
After mucking around with it a bit, I was able to get it to sync with the channel bank as the clock source for the line: jsci-cbank1> show clock Primary Master Transmit Clock Source:Internal Secondary Transmit Clock Source: Slot A DS1 2 Also, are you using T1 crossover cables? The ADIT 600 channel banks and the digium cards are both DTE... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of TC Sent: Saturday, May 31, 2003 1:22 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CAC ADIT600 / T400 config >>I have an adit 600, but I'm not using the second span. >>I did have some small difficulties getting a single link to come up, had to set * to provide timing to the line, and the CB to recover it >>from the line. If you get it working, could you let us know what it was? what was the exact set clock stmts you used were they like this set clock1 a:1 set clock2 internal Also I now see I dont get dial tone even on the T1-1 span for the fxs devices Could I see your print config output from the ADIT 600 thx and yea when i get this up I send it to Wade's channel ban info site for others ___ 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] adsi and voicemail application not working
Can anyone tell me what I need to do to tell the voicemail and voicemail2 applications to download to a different slot? The only problem I have remaining is this one... I dial the voicemailmain extension from a PT480, and the display tells me "services is full, download refused", but asterisk PBX only occupies the first slot (I changed this)... Any help would be greatly appreciated. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jayson Vantuyl Sent: Friday, May 30, 2003 9:12 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] aastra pt480 and adsi On Thu, May 29, 2003 at 08:58:43PM -0400, Joe Antkowiak wrote: > Ok, so I figured out my problem with my pt480s. But, now I have a few more. > > 1. When I dial into the voicemailmain or voicemailmain2 application, the > phone and * start talking adsi, but then the phone tells me "programming > download canceled, services is full.", but my services list isn't full, only > "Asterisk PBX" occupies slot 2, slots 1, 3 and 4 are available. Any ideas? > I have tried erasing all the services programmed in, and reloading them with > ADSIProg, or even before that trying voicemail, but the same thing always > happens. It all depends on the FDN that you are programming it with. You can't use the same one for the Asterisk PBX script and the voicemail script. If you have the option, load the Asterisk PBX into the slot marked SL (for self-load) so it will load automatically when the phone is idle. > 2. I can't seem to get call waiting id to work. I hear the "adsi" tones on > the line when another call is coming in, but the phone doesn't seem to > recognize it. Any ideas? Actually, that's not ADSI, it's standard call waiting. It could be that your script doesn't have CALLWAITING event. > 3. Is there a list or some documentation somewhere on what all the available > adsi programming options there are, that I can use in .adsi files? I've been using adsiprog.c. It's all pretty fuzzy on what the stuff means, though. Jayson ___ 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] CAC ADIT600 / T400 config
I have an adit 600, but I’m not using the second span… I did have some small difficulties getting a single link to come up, had to set * to provide timing to the line, and the CB to recover it from the line… If you get it working, could you let us know what it was? Will be doing a lot more adits and * soon… -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of TC Sent: Saturday, May 31, 2003 5:04 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] CAC ADIT600 / T400 config I know a few ppl have those CAC Adit 600's with t400 I can't seem to get my second span up on the T400 connected to the second spand on the adit (A:2) A:1 seems ok Can someone post they zaptel.conf span defintions And maybe a "print config" from the adit 600 cli I think my issue is timing srcs the coding, framing. bld out are all matched thx
RE: [Asterisk-Users] SIP echo?
Just an FYI, my problem was fixed by changing my call format to ulaw and setting echocancel and echocancelwhenbridged=128 instead of yes. Works like a charm now =) Does ulaw use 64k? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Alexander Sent: Friday, May 30, 2003 7:57 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP echo? I have had similar problems but one thing that seemed to help in my case was to back off the rxgain and txgain for the X100P. I haven't yet had the chance to experiment fully. I think I also have echocancel=128 for the X100P channel and the echo canceller does seem to train up over the first few seconds of a call now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Antkowiak Sent: Friday, May 30, 2003 2:10 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP echo? I noticed a few other messages posted about this problem, but I couldn't find an answer... I'm having a problem with SIP echo when calls are received into asterisk via an x100p and bridged with a sip extension (back to the pstn with iconnecthere). the person calling in to asterisk has no echo problems, but the recipient of the pstn call, everything they say, they hear back about 1 second later. echocancel=yes and echocancelwhenbridged=yes are in the applicable channels in zapata.conf. I can also use the PC client from iconnecthere and I do not have the problem. Any ideas? Also, what would be the best codec to use to send fax transmissions via SIP? ___ 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] receptionist application for asterisk?
You could do this with a combination of some adsi phones + custom scripts + even changing the caller id name when a call is received to the company the call should be answered as... If you need any help, just reply... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Sellens Sent: Friday, May 30, 2003 7:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] receptionist application for asterisk? I've tried searching the archives but come up empty. I'm looking for a "receptionist" application, that could be used in a business center. Lots of different incoming phone numbers (200+), all need to be answered in a different way, by a human. This is typically enabled with a terminal of some form that pops up the company and contact information for the called number, and a keyboard interface that allows the recptionist/operator to park and retrieve calls, toss to voicemail, hangup, forward, etc. I think I have some ideas on how this could be done, but I hate reinventing the wheel. Does anyone know of code (free or commercial) that implements this sort of thing for asterisk? Thanks very much for any pointers, hints, suggestions, etc. John [EMAIL PROTECTED] ___ 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] SIP echo?
I noticed a few other messages posted about this problem, but I couldn't find an answer... I'm having a problem with SIP echo when calls are received into asterisk via an x100p and bridged with a sip extension (back to the pstn with iconnecthere). the person calling in to asterisk has no echo problems, but the recipient of the pstn call, everything they say, they hear back about 1 second later. echocancel=yes and echocancelwhenbridged=yes are in the applicable channels in zapata.conf. I can also use the PC client from iconnecthere and I do not have the problem. Any ideas? Also, what would be the best codec to use to send fax transmissions via SIP? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] aastra pt480 and adsi
how do I specify a different one for the voicemail script? This is what the top of my asterisk.adsi looks like: DESCRIPTION "Asterisk PBX" ; Name of vendor VERSION 0x00; Version of stuff FDN 0x85efd9da SECURITY 0x78921d49 And asterisk is currently occupying the SL slot, I don't have the option to specify which slot I want it to be loaded into... -Original Message- From: Jayson Vantuyl <[EMAIL PROTECTED]> Date: Fri, 30 May 2003 08:12:08 -0500 To: [EMAIL PROTECTED] <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] aastra pt480 and adsi On Thu, May 29, 2003 at 08:58:43PM -0400, Joe Antkowiak wrote: > Ok, so I figured out my problem with my pt480s. But, now I have a few more. > > 1. When I dial into the voicemailmain or voicemailmain2 application, the > phone and * start talking adsi, but then the phone tells me "programming > download canceled, services is full.", but my services list isn't full, only > "Asterisk PBX" occupies slot 2, slots 1, 3 and 4 are available. Any ideas? > I have tried erasing all the services programmed in, and reloading them with > ADSIProg, or even before that trying voicemail, but the same thing always > happens. It all depends on the FDN that you are programming it with. You can't use the same one for the Asterisk PBX script and the voicemail script. If you have the option, load the Asterisk PBX into the slot marked SL (for self-load) so it will load automatically when the phone is idle. > 2. I can't seem to get call waiting id to work. I hear the "adsi" tones on > the line when another call is coming in, but the phone doesn't seem to > recognize it. Any ideas? Actually, that's not ADSI, it's standard call waiting. It could be that your script doesn't have CALLWAITING event. > 3. Is there a list or some documentation somewhere on what all the available > adsi programming options there are, that I can use in .adsi files? I've been using adsiprog.c. It's all pretty fuzzy on what the stuff means, though. Jayson ___ 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] aastra pt480 and adsi
Ok, so I figured out my problem with my pt480s. But, now I have a few more. 1. When I dial into the voicemailmain or voicemailmain2 application, the phone and * start talking adsi, but then the phone tells me "programming download canceled, services is full.", but my services list isn't full, only "Asterisk PBX" occupies slot 2, slots 1, 3 and 4 are available. Any ideas? I have tried erasing all the services programmed in, and reloading them with ADSIProg, or even before that trying voicemail, but the same thing always happens. 2. I can't seem to get call waiting id to work. I hear the "adsi" tones on the line when another call is coming in, but the phone doesn't seem to recognize it. Any ideas? 3. Is there a list or some documentation somewhere on what all the available adsi programming options there are, that I can use in .adsi files? Thanks. -Joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1-PRI deployment questions...
B8ZS/ESF I believe is the usual for a PRI DID calls in asterisk are routed just like dtmf dialed extensions, but there are not DTMF tones passed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Harragin Sent: Thursday, May 29, 2003 11:05 AM To: Asterisk Subject: [Asterisk-Users] T1-PRI deployment questions... I am ordering T1-PRI service from local service provider and have a few questions. Is there framing and coding considerations (or is it all one standard), if so what is best? How are calls routed based on DIDs - are these just dtmf tones passed after the call is picked up and treated as normal exten=> definitions? John This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus. ___ 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] 2 4-port T1 cards
>Also, NFS mounting of the voicemail for such a large install is probably >not the best idea. Unless you really need it available to another >machine, you _may_ want to rethink this idea. NFS can be a major speed >hit on a machine, especially if the client is overworked. Also if you >are planning on running most all the channels to voicemail, then do you >think you are going to be able to have your NFS server keep high speed >writing going so as not to slow you asterisk machine down with it's 96 >channels running full tilt? Well, I have to have some centralized way of storing voicemail that all the * boxes can access. Its either network storage, SAN, or all the * boxes with t1 cards do IAX to a central * box that will do all the voicemail. I didn't think I'd have much of a problem getting the nfs box to keep up as long as its got enough horsepower (dual cpu, gig NIC, serial ata raid 5+1), and if I do, SAN is the next step... What do you think? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 4-port T1 cards
Cool, dual CPU it is =) I'll watch out for the frame buffers. Other than that, are there any other known issues with doing this? I will also be using dual-channel memory... Thanks! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer Sent: Wednesday, May 28, 2003 12:38 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 4-port T1 cards > 3. Dual MB won't help much in pure telephony. > In pure telephony, you are basically dealing with serial line > IO. A T1 is little more than I long distance serial line. 8 T1s > is just 11.7megs per second each way, or 23.4 megs in and out. > Not too much for a good machine to do. Granted, if you are doing VoIP > then you add another set of ins and outs with compression in the middle > of it too. This is where the second CPU comes in handy. Actually, with Zaptel, and the T400P especially, dual CPU makes a *big* difference. The T400P and E400P are slave-only designs, so the CPU spends a lot of time just cramming I/O down the PCI bus. Having a second CPU free to do work will definitely help. > 4. AGP Video. > Make sure not to use the frame buffer, it has been reported that the > frame buffer generates large amounts of interupts and will > degrade the performance. Don't underestimate this effect or think that a fast CPU will get around it. frame buffer is a definite no-no because it disables interrupts during screen redraws which take an enormous amount of time. If your call quality drops while you're playing quake on your PBX, don't come crying to us ;-) Mark ___ 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] 2 4-port T1 cards
1. Voicemail, and the voicemail itself will be stored on another box, NFS mounted, or I might use mysql. There will be a little bit of call routing via iax to a separate * box with a channel bank on it. 2. I don't disagree with you, they do throw in a lot, but redhat does have its advantages, IMHO. I've always been able to get things to work quickly with redhat, and there is that whole 24 hour support contract we have with them... 3. Mmm, ok. 4. Does the ati radeon 9000 have a frame buffer? That's the card I was going to use for all the * boxes. Thank you very much. -Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Tuesday, May 27, 2003 6:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 2 4-port T1 cards On Tue, 2003-05-27 at 16:05, Joe Antkowiak wrote: > Are there any known issues with putting 2 4-port T1 cards in a single box > and having all ports and all channels in use at the same time? Planning on > 4 of these boxes, dual AMD cpu MB from MSI, 512m, redhat 9, agp video, on > board NICs, serial ata raid. Newbie 101 (Not deragatory) 1. What are you doing with these ports? If you are routing calls from one side of the cards to the other, then you should have no problems with a 1gig P3 or so. But if you are doing more than routing, it will depend on what that something is, and what kind of overhead it is going to impose. 2. RH blows chunks. (Personal opinion) RH is known to make kitchen sink installs when you don't need them, and would be better off without most of the install base. 3. Dual MB won't help much in pure telephony. In pure telephony, you are basically dealing with serial line IO. A T1 is little more than I long distance serial line. 8 T1s is just 11.7megs per second each way, or 23.4 megs in and out. Not too much for a good machine to do. Granted, if you are doing VoIP then you add another set of ins and outs with compression inthe middle of it too. This is where the second CPU comes in handy. 4. AGP Video. Make sure not to use the frame buffer, it has been reported that the frame buffer generates large amounts of interupts and will degrade the performance. Here is for discussion as it is parts I don't know real well. Will the serial ATA buy you any flexibilty or lowered CPU load while accessing the disk? Don't take this question as shooting down the SATA, just don't know if there is real benefit in it yet. Also what chipset is the onboard nics? -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] 2 4-port T1 cards
Are there any known issues with putting 2 4-port T1 cards in a single box and having all ports and all channels in use at the same time? Planning on 4 of these boxes, dual AMD cpu MB from MSI, 512m, redhat 9, agp video, on board NICs, serial ata raid. Any input would be appreciated. -Joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] The Phantom Call.. T1 card too
No popping/bad audio on this one, clear as can be, asterisk just decides to pick up the channel after about a minute and use the "s" extension in the context... immediate=no is set on this channel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Tuesday, May 27, 2003 1:36 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] The Phantom Call.. T1 card too I had a similar problem when there was timing problems with my T100P. You could also hear lots of popping and generally bad audio during the dial tone. After we fixed the timing problem, the audio was clear as could be, and the problems went away. Of course this doesn't fix a X100P as it is strictly analog. On Tue, 2003-05-27 at 12:03, Joe Antkowiak wrote: > I've had the same thing happen, only on the single port T1 card and a > channel bank, and one of the FXO channels also having a phone attached > elsewhere... > > I just wound up putting that channel in a different context and running > > Exten => s,1,Hangup > > (I'm just using the line for outbound dialing) > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Tamas Levente > Sent: Tuesday, May 27, 2003 11:45 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] The Phantom Call.. > > Same thing happened with me too. X100P. Same US tones > Sometimes it gets into the voicemail too:)) And the voicemail record 3 > minutes tone, after 1.5minutes it's service not available or something > similar. > Is there a fix for that? > > - Original Message - > From: "Mark Street" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, May 27, 2003 5:39 PM > Subject: Re: [Asterisk-Users] The Phantom Call.. > > > > Funny, I just noticed this happening on my box with 2 X101P's installed > and a > > phone connected to the same line as one of the X101P's. I pick up the > phone > > after 1 ring, or call someone. After a minute or two * picks up the line > and > > starts the greeting. I pull the plug on the asterisk box to continue > the > > conversation. I just noticed it happening a couple of weeks ago. US > > dialtone here... > > > > On Tuesday 27 May 2003 08:13, Mark Spencer wrote: > > > > Could it be that the X100P is detecting the UK dial tone as a ring?? > > > > or Has anyone else had a similar problem when using the X100P/S100U > > > > combination?? > > > > > > It's possible there is *something* on the line that is confusing > Asterisk > > > into thinking a ring takes place. You might try adjusting the value of > > > PEGCOUNT in wcfxo.c to a higher value (say, 10). > > > > -- > > Mark Street, D.C. > > Red Hat Certified Engineer > > Cert# 807302251406074 > > -- > > Key fingerprint = 3949 39E4 6317 7C3C 023E 2B1F 6FB3 06E7 D109 56C0 > > GPG key http://www.streetchiro.com/pubkey.asc > > > > ___ > > 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] The Phantom Call.. T1 card too
I've had the same thing happen, only on the single port T1 card and a channel bank, and one of the FXO channels also having a phone attached elsewhere... I just wound up putting that channel in a different context and running Exten => s,1,Hangup (I'm just using the line for outbound dialing) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tamas Levente Sent: Tuesday, May 27, 2003 11:45 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] The Phantom Call.. Same thing happened with me too. X100P. Same US tones Sometimes it gets into the voicemail too:)) And the voicemail record 3 minutes tone, after 1.5minutes it's service not available or something similar. Is there a fix for that? - Original Message - From: "Mark Street" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, May 27, 2003 5:39 PM Subject: Re: [Asterisk-Users] The Phantom Call.. > Funny, I just noticed this happening on my box with 2 X101P's installed and a > phone connected to the same line as one of the X101P's. I pick up the phone > after 1 ring, or call someone. After a minute or two * picks up the line and > starts the greeting. I pull the plug on the asterisk box to continue the > conversation. I just noticed it happening a couple of weeks ago. US > dialtone here... > > On Tuesday 27 May 2003 08:13, Mark Spencer wrote: > > > Could it be that the X100P is detecting the UK dial tone as a ring?? > > > or Has anyone else had a similar problem when using the X100P/S100U > > > combination?? > > > > It's possible there is *something* on the line that is confusing Asterisk > > into thinking a ring takes place. You might try adjusting the value of > > PEGCOUNT in wcfxo.c to a higher value (say, 10). > > -- > Mark Street, D.C. > Red Hat Certified Engineer > Cert# 807302251406074 > -- > Key fingerprint = 3949 39E4 6317 7C3C 023E 2B1F 6FB3 06E7 D109 56C0 > GPG key http://www.streetchiro.com/pubkey.asc > > ___ > 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
