Re: [Asterisk-Users] X-Lite & Asterisk: Speex & iLBC not working?

2004-01-28 Thread Wim Venneman 

If this may be of any use:

I'm not an expert but I did the test with the FWD soft phone from X-ten and
iLBC & SPX don't work.
Asterisk wasn't between the connections. Just x-lite and fwd (who is an
Asterisk Server?)
The soft phone makes the connection but I can't hear any sound.

Wim


- Original Message -
From: "Fran Boon" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, January 26, 2004 3:01 PM
Subject: [Asterisk-Users] X-Lite & Asterisk: Speex & iLBC not working?


> This seems to have been reported before, but I've seen no resolution:
> http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html
> http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html
> http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html
>
> When forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the
> Asterisk server)
> When forcing iLBC, there is some very garbled noise, but nothing
> intelligible.
>
> Sniffing the packets, I can see that X-Lite & Asterisk have chosen
> differing 'Payload type' numbers:
> X-Lite:
> a=rtpmap:97 speex/8000
> a=rtpmap:98 iLBC/8000
> Asterisk:
> a=rtpmap:97 iLBC/8000
> a=rtpmap:110 SPEEX/8000
>
> According to the Speex RFC, this is acceptable:
> http://speex.org/drafts/draft-herlein-speex-rtp-profile-00.txt
> "Dynamic payload type codes MUST be negotiated 'out-of-band' for the
> assignment of a dynamic payload type from the range of 96-127."
>
> I'm wondering whether the system is at all case sensitive?
>  From the RFC:
> "When conveying information by SDP [4], the encoding name SHALL be
"speex"."
>
> NB Ethereal shows payload-type as being 97 when X-Lite reports iLBC &
> 110 when X-Lite reports Speex, so the Asterisk numbers seem to 'win'.
>
> Any light shed on this would be great.
> Whilst GSM is ok, it would be great to leverage the power of Speex :)
>
> Thanks a lot,
> Fran.
> ___
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Unable to create channem of type 'Zap'

2004-02-23 Thread Wim Venneman 




Can anyone 
help me, (after a two day search, also on the mailing list)
I have the 
following situation:
Asterisk 
works fine, until I added a FXO card. (Digium)
When I 
tried to call to the pstn I have the following error
Executing 
Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack
NOTIVE[16401]: 
FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE CHANNEL OF TYPE 
'ZAP'
 == 
Everyone is busy at this time
When I 
start Asterisk I have no error
Only the 
following isn't right: 
ZAP 
SHOW CHANNELS = No channels 
modprobe 
wcfxo = ok (no errors)
 I have 
following config.
ZAPATA
[channels]language=encontext=incomingsignalling=fxs_ksusecallerid=yeshidecallerid=nocallwaiting=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0group=1pickupgroup=1immediate=yesmusiconhold=default channel => 1
ZAPTEL 

loadzone = 
usdefaultzone = usfxsks = 
1
EXTENSION
[incoming]exten => 
s,1,Dial(SIP/Phone1&SIP/Phone3,20,tr)
[outgoing]exten => _0X.,1,Dial,Zap/1/${EXTEN:1}
IN 
[SIP]
include 
=> outgoing
I'm don't 
know what I can change to the config.
Anyone an 
idea
Thanks,
Wim

Re: [Asterisk-Users] Unable to create channem of type 'Zap'

2004-02-23 Thread Wim Venneman 
Made changes:

1)
musiconhold= default
channel => 1

2)
reboot
modprobe wcfxo = ok
ztcfg -v

result = 1 channel configured

Try to dial, still the same problem. (error)

Wim


- Original Message - 
From: "Brent Franks" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, February 23, 2004 9:19 PM
Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap'


> Make sure you run a ztcfg after you do a modprobe.
> 
> ztcfg will configure (or bring up) the zap channels on zaptel interface
> cards.  Do this before starting * and after the modprobe.
> 
> (You may also do a ztcfg -v to see whats configured)
> 
> - Brent
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Wim Venneman
> Sent: Monday, February 23, 2004 3:10 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Unable to create channem of type 'Zap'
> 
> Can anyone help me, (after a two day search, also on the mailing list)
> I have the following situation:
> Asterisk works fine, until I added a FXO card. (Digium)
> When I tried to call to the pstn I have the following error
> Executing Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack
> NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE
> CHANNEL OF TYPE 'ZAP'
> == Everyone is busy at this time
> When I start Asterisk I have no error
> Only the following isn't right: 
> ZAP SHOW CHANNELS = No channels 
> modprobe wcfxo = ok (no errors)
> I have following config.
> ZAPATA
> [channels]
> language=en
> context=incoming
> signalling=fxs_ks
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> group=1
> pickupgroup=1
> immediate=yes
> musiconhold=default channel => 1
> ZAPTEL 
> loadzone = us
> defaultzone = us
> fxsks = 1
> EXTENSION
> [incoming]
> exten => s,1,Dial(SIP/Phone1&SIP/Phone3,20,tr)
> [outgoing]
> exten => _0X.,1,Dial,Zap/1/${EXTEN:1}
> IN [SIP]
> include => outgoing
> I'm don't know what I can change to the config.
> Anyone an idea
> Thanks,
> Wim
> 
> ___
> 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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Re: [Asterisk-Users] Unable to create channem of type 'Zap'

2004-02-23 Thread Wim Venneman 
Thanks for the help !

Made changes, still the same message.
I have two NIC's with IRQ 11
The FXO card has IRQ10 (and no other card has IRQ10)

Wim


- Original Message - 
From: "Brent Franks" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, February 23, 2004 10:21 PM
Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap'


> Wim, I made some changes to your Zapata.conf and zaptel.conf config
> files below.
> 
> Hope this helps.
> 
> Also, do a less /proc/interrupts and see if the card is on it's own IRQ.
> 
> - Brent
> 
> -Original Message-----
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Wim Venneman
> Sent: Monday, February 23, 2004 3:10 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Unable to create channem of type 'Zap'
> 
> Can anyone help me, (after a two day search, also on the mailing list)
> I have the following situation:
> Asterisk works fine, until I added a FXO card. (Digium)
> When I tried to call to the pstn I have the following error
> Executing Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack
> NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE
> CHANNEL OF TYPE 'ZAP'
> == Everyone is busy at this time
> When I start Asterisk I have no error
> Only the following isn't right: 
> ZAP SHOW CHANNELS = No channels 
> modprobe wcfxo = ok (no errors)
> I have following config.
> ZAPATA
> [channels]
> language=en
> group=1
> pickupgroup=1
> context=incoming
> signalling=fxs_ks
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> immediate=yes
> musiconhold=default
> channel = 1
> 
> ZAPTEL 
> loadzone = us
> defaultzone = us
> fxsks = 1
> 
> EXTENSION
> [incoming]
> exten => s,1,Dial(SIP/Phone1&SIP/Phone3,20,tr)
> [outgoing]
> exten => _0X.,1,Dial,Zap/1/${EXTEN:1}
> 
> IN [SIP]
> include => outgoing
> I'm don't know what I can change to the config.
> Anyone an idea
> Thanks,
> Wim
> 
> ___
> 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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Re: [Asterisk-Users] Unable to create channem of type 'Zap'

2004-02-24 Thread Wim Venneman 
Thanks Derek,

Changed the channel = 1 to channel => 1, makes no difference.

Wim

- Original Message - 
From: "Derek Samford" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Cc: <[EMAIL PROTECTED]>
Sent: Tuesday, February 24, 2004 6:38 PM
Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap'


> Wim,
> Made one more change below in Zapata.conf
>  It should be channel => 1
> 
> -Original Message-
> From: Wim Venneman [mailto:[EMAIL PROTECTED] 
> Sent: Monday, February 23, 2004 4:46 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap'
> 
> Thanks for the help !
> 
> Made changes, still the same message.
> I have two NIC's with IRQ 11
> The FXO card has IRQ10 (and no other card has IRQ10)
> 
> Wim
> 
> 
> - Original Message - 
> From: "Brent Franks" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Monday, February 23, 2004 10:21 PM
> Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap'
> 
> 
> > Wim, I made some changes to your Zapata.conf and zaptel.conf config
> > files below.
> > 
> > Hope this helps.
> > 
> > Also, do a less /proc/interrupts and see if the card is on it's own
> IRQ.
> > 
> > - Brent
> > 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Wim
> Venneman
> > Sent: Monday, February 23, 2004 3:10 PM
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] Unable to create channem of type 'Zap'
> > 
> > Can anyone help me, (after a two day search, also on the mailing list)
> > I have the following situation:
> > Asterisk works fine, until I added a FXO card. (Digium)
> > When I tried to call to the pstn I have the following error
> > Executing Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack
> > NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE
> > CHANNEL OF TYPE 'ZAP'
> > == Everyone is busy at this time
> > When I start Asterisk I have no error
> > Only the following isn't right: 
> > ZAP SHOW CHANNELS = No channels 
> > modprobe wcfxo = ok (no errors)
> > I have following config.
> > ZAPATA
> > [channels]
> > language=en
> > group=1
> > pickupgroup=1
> > context=incoming
> > signalling=fxs_ks
> > usecallerid=yes
> > hidecallerid=no
> > callwaiting=yes
> > callwaitingcallerid=yes
> > threewaycalling=yes
> > transfer=yes
> > cancallforward=yes
> > callreturn=yes
> > echocancel=yes
> > echocancelwhenbridged=yes
> > rxgain=0.0
> > txgain=0.0
> > immediate=yes
> > musiconhold=default
> > channel => 1
> > 
> > ZAPTEL 
> > loadzone = us
> > defaultzone = us
> > fxsks = 1
> > 
> > EXTENSION
> > [incoming]
> > exten => s,1,Dial(SIP/Phone1&SIP/Phone3,20,tr)
> > [outgoing]
> > exten => _0X.,1,Dial,Zap/1/${EXTEN:1}
> > 
> > IN [SIP]
> > include => outgoing
> > I'm don't know what I can change to the config.
> > Anyone an idea
> > Thanks,
> > Wim
> > 
> > ___
> > 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
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 

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Re: [Asterisk-Users] Unable to create channem of type 'Zap'

2004-02-24 Thread Wim Venneman 
Brent,

Zaptel configuration
==
Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

1 channel configured.

Registered tone zone 0 (United States / North America)

Wim


- Original Message -
From: "Brent Franks" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, February 24, 2004 8:13 PM
Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap'


> Wim,
>
> What happens when you do a ztcfg -vv
>
> - Brent
>
> On Tue, 24 Feb 2004, Wim Venneman wrote:
>
> > Thanks Derek,
> >
> > Changed the channel = 1 to channel => 1, makes no difference.
> >
> > Wim
> >
> > - Original Message -
> > From: "Derek Samford" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Cc: <[EMAIL PROTECTED]>
> > Sent: Tuesday, February 24, 2004 6:38 PM
> > Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap'
> >
> >
> > > Wim,
> > > Made one more change below in Zapata.conf
> > >  It should be channel => 1
> > >
> > > -Original Message-
> > > From: Wim Venneman [mailto:[EMAIL PROTECTED]
> > > Sent: Monday, February 23, 2004 4:46 PM
> > > To: [EMAIL PROTECTED]
> > > Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap'
> > >
> > > Thanks for the help !
> > >
> > > Made changes, still the same message.
> > > I have two NIC's with IRQ 11
> > > The FXO card has IRQ10 (and no other card has IRQ10)
> > >
> > > Wim
> > >
> > >
> > > - Original Message -
> > > From: "Brent Franks" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > Sent: Monday, February 23, 2004 10:21 PM
> > > Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap'
> > >
> > >
> > > > Wim, I made some changes to your Zapata.conf and zaptel.conf config
> > > > files below.
> > > >
> > > > Hope this helps.
> > > >
> > > > Also, do a less /proc/interrupts and see if the card is on it's own
> > > IRQ.
> > > >
> > > > - Brent
> > > >
> > > > -Original Message-
> > > > From: [EMAIL PROTECTED]
> > > > [mailto:[EMAIL PROTECTED] On Behalf Of Wim
> > > Venneman
> > > > Sent: Monday, February 23, 2004 3:10 PM
> > > > To: [EMAIL PROTECTED]
> > > > Subject: [Asterisk-Users] Unable to create channem of type 'Zap'
> > > >
> > > > Can anyone help me, (after a two day search, also on the mailing
list)
> > > > I have the following situation:
> > > > Asterisk works fine, until I added a FXO card. (Digium)
> > > > When I tried to call to the pstn I have the following error
> > > > Executing Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack
> > > > NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO
CREATE
> > > > CHANNEL OF TYPE 'ZAP'
> > > > == Everyone is busy at this time
> > > > When I start Asterisk I have no error
> > > > Only the following isn't right:
> > > > ZAP SHOW CHANNELS = No channels
> > > > modprobe wcfxo = ok (no errors)
> > > > I have following config.
> > > > ZAPATA
> > > > [channels]
> > > > language=en
> > > > group=1
> > > > pickupgroup=1
> > > > context=incoming
> > > > signalling=fxs_ks
> > > > usecallerid=yes
> > > > hidecallerid=no
> > > > callwaiting=yes
> > > > callwaitingcallerid=yes
> > > > threewaycalling=yes
> > > > transfer=yes
> > > > cancallforward=yes
> > > > callreturn=yes
> > > > echocancel=yes
> > > > echocancelwhenbridged=yes
> > > > rxgain=0.0
> > > > txgain=0.0
> > > > immediate=yes
> > > > musiconhold=default
> > > > channel => 1
> > > >
> > > > ZAPTEL
> > > > loadzone = us
> > > > defaultzone = us
> > > > fxsks = 1
> > > >
> > > > EXTENSION
> > > > [incoming]
> > > > exten => s,1,Dial(SIP/Phone1&SIP/Phone3,20,tr)
> > > > [outgoing]
> > > > exten => _0X.,1,Dial,Zap/1/${EXTEN:1}
> > > >
> > > > IN [SIP]
> &

Re: [Asterisk-Users] Unable to create channel of type 'Zap'

2004-02-24 Thread Wim Venneman 
Dave,

I have fxsks = 1
(fxsls is for loopstart prot. and it doesn't work for me)

Wim

- Original Message -
From: "David J Carter" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, February 24, 2004 8:33 PM
Subject: RE: [Asterisk-Users] Unable to create channel of type 'Zap'


> I had this after my last CVS update.
>
> A line in Zaptel.conf was set to fxsls=1 instaead of fxsks=1
>
> Dave
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Wim Venneman
> Sent: 24 February 2004 19:17
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap'
>
>
> Thanks Derek,
>
> Changed the channel = 1 to channel => 1, makes no difference.
>
> Wim
>
> - Original Message -
> From: "Derek Samford" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Cc: <[EMAIL PROTECTED]>
> Sent: Tuesday, February 24, 2004 6:38 PM
> Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap'
>
>
> > Wim,
> > Made one more change below in Zapata.conf
> >  It should be channel => 1
> >
> > -Original Message-
> > From: Wim Venneman [mailto:[EMAIL PROTECTED]
> > Sent: Monday, February 23, 2004 4:46 PM
> > To: [EMAIL PROTECTED]
> > Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap'
> >
> > Thanks for the help !
> >
> > Made changes, still the same message.
> > I have two NIC's with IRQ 11
> > The FXO card has IRQ10 (and no other card has IRQ10)
> >
> > Wim
> >
> >
> > - Original Message -
> > From: "Brent Franks" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Monday, February 23, 2004 10:21 PM
> > Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap'
> >
> >
> > > Wim, I made some changes to your Zapata.conf and zaptel.conf config
> > > files below.
> > >
> > > Hope this helps.
> > >
> > > Also, do a less /proc/interrupts and see if the card is on it's own
> > IRQ.
> > >
> > > - Brent
> > >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] On Behalf Of Wim
> > Venneman
> > > Sent: Monday, February 23, 2004 3:10 PM
> > > To: [EMAIL PROTECTED]
> > > Subject: [Asterisk-Users] Unable to create channem of type 'Zap'
> > >
> > > Can anyone help me, (after a two day search, also on the mailing list)
> > > I have the following situation:
> > > Asterisk works fine, until I added a FXO card. (Digium)
> > > When I tried to call to the pstn I have the following error
> > > Executing Dial("SIP/Phone2-fc49", "Zap/1/2355") in new stack
> > > NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE
> > > CHANNEL OF TYPE 'ZAP'
> > > == Everyone is busy at this time
> > > When I start Asterisk I have no error
> > > Only the following isn't right:
> > > ZAP SHOW CHANNELS = No channels
> > > modprobe wcfxo = ok (no errors)
> > > I have following config.
> > > ZAPATA
> > > [channels]
> > > language=en
> > > group=1
> > > pickupgroup=1
> > > context=incoming
> > > signalling=fxs_ks
> > > usecallerid=yes
> > > hidecallerid=no
> > > callwaiting=yes
> > > callwaitingcallerid=yes
> > > threewaycalling=yes
> > > transfer=yes
> > > cancallforward=yes
> > > callreturn=yes
> > > echocancel=yes
> > > echocancelwhenbridged=yes
> > > rxgain=0.0
> > > txgain=0.0
> > > immediate=yes
> > > musiconhold=default
> > > channel => 1
> > >
> > > ZAPTEL
> > > loadzone = us
> > > defaultzone = us
> > > fxsks = 1
> > >
> > > EXTENSION
> > > [incoming]
> > > exten => s,1,Dial(SIP/Phone1&SIP/Phone3,20,tr)
> > > [outgoing]
> > > exten => _0X.,1,Dial,Zap/1/${EXTEN:1}
> > >
> > > IN [SIP]
> > > include => outgoing
> > > I'm don't know what I can change to the config.
> > > Anyone an idea
> > > Thanks,
> > > Wim
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.c

Re: [Asterisk-Users] Unable to create channel of type 'Zap'

2004-02-24 Thread Wim Venneman 
Thanks Brent,

You will never believe, it works.
I copied a config from a friend (ZAPATA.CONF) and now it works fine.
I can call to * from pstn and can call from * to pstn.
I will take a look tomorrow what was wrong in my config and let you know.

Wim


- Original Message - 
From: "Brent Franks" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, February 24, 2004 10:21 PM
Subject: RE: [Asterisk-Users] Unable to create channel of type 'Zap'


> The best thing I can recommend is giving Digium a call.  That's part of
> the reason to buy the cards from Digium, as the support is included free
> with the hardware you purchase from them.  Well worth the money.  They
> have an excellent support staff that should be able to see where the
> problem is cropping up.
> 
> - B
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Wim Venneman
> > Sent: Tuesday, February 24, 2004 2:47 PM
> > To: [EMAIL PROTECTED]
> > Subject: Re: [Asterisk-Users] Unable to create channel of type 'Zap'
> > 
> > Dave,
> > 
> > I have fxsks = 1
> > (fxsls is for loopstart prot. and it doesn't work for me)
> > 
> > Wim
> > 
> > - Original Message -
> > From: "David J Carter" <[EMAIL PROTECTED]>
> > To: <[EMAIL PROTECTED]>
> > Sent: Tuesday, February 24, 2004 8:33 PM
> > Subject: RE: [Asterisk-Users] Unable to create channel of type 'Zap'
> > 
> > 
> > > I had this after my last CVS update.
> > >
> > > A line in Zaptel.conf was set to fxsls=1 instaead of fxsks=1
> > >
> > > Dave
> > >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] Behalf Of Wim
> Venneman
> > > Sent: 24 February 2004 19:17
> > > To: [EMAIL PROTECTED]
> > > Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap'
> > >
> > >
> > > Thanks Derek,
> > >
> > > Changed the channel = 1 to channel => 1, makes no difference.
> > >
> > > Wim
> > >
> > > - Original Message -
> > > From: "Derek Samford" <[EMAIL PROTECTED]>
> > > To: <[EMAIL PROTECTED]>
> > > Cc: <[EMAIL PROTECTED]>
> > > Sent: Tuesday, February 24, 2004 6:38 PM
> > > Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap'
> > >
> > >
> > > > Wim,
> > > > Made one more change below in Zapata.conf
> > > >  It should be channel => 1
> > > >
> > > > -Original Message-
> > > > From: Wim Venneman [mailto:[EMAIL PROTECTED]
> > > > Sent: Monday, February 23, 2004 4:46 PM
> > > > To: [EMAIL PROTECTED]
> > > > Subject: Re: [Asterisk-Users] Unable to create channem of type
> 'Zap'
> > > >
> > > > Thanks for the help !
> > > >
> > > > Made changes, still the same message.
> > > > I have two NIC's with IRQ 11
> > > > The FXO card has IRQ10 (and no other card has IRQ10)
> > > >
> > > > Wim
> > > >
> > > >
> > > > - Original Message -
> > > > From: "Brent Franks" <[EMAIL PROTECTED]>
> > > > To: <[EMAIL PROTECTED]>
> > > > Sent: Monday, February 23, 2004 10:21 PM
> > > > Subject: RE: [Asterisk-Users] Unable to create channem of type
> 'Zap'
> > > >
> > > >
> > > > > Wim, I made some changes to your Zapata.conf and zaptel.conf
> config
> > > > > files below.
> > > > >
> > > > > Hope this helps.
> > > > >
> > > > > Also, do a less /proc/interrupts and see if the card is on it's
> own
> > > > IRQ.
> > > > >
> > > > > - Brent
> > > > >
> > > > > -Original Message-
> > > > > From: [EMAIL PROTECTED]
> > > > > [mailto:[EMAIL PROTECTED] On Behalf Of Wim
> > > > Venneman
> > > > > Sent: Monday, February 23, 2004 3:10 PM
> > > > > To: [EMAIL PROTECTED]
> > > > > Subject: [Asterisk-Users] Unable to create channem of type 'Zap'
> > > > >
> > > > > Can anyone help me, (after a two day search, also on the mailing
> > list)
> > > > > I have the f

Re: [Asterisk-Users] Dial up adapter

2004-03-01 Thread Wim Venneman 
Maybe this??: http://www.grandstream.com/y-product.htm


- Original Message -
From: "Jason Miller" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, March 01, 2004 10:53 PM
Subject: [Asterisk-Users] Dial up adapter


I was wondering if anyone has used an adapter to dial up to a local internet
service then used the VOIP phone instead of needing a computer. If so what
product do you suggest? An idea of what I am looking for,

configure the device which has a analog port to dial said ISP and
authenticate
Has an ethernet port to hook up to the phone


Or am I just dreaming up a new product to market?



Jason




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[Asterisk-Users] Host unspecified ??

2003-10-29 Thread Wim Venneman 




Dear,
 
When I start asterisk  -vvgrc and I 
ask 'sip show peers', I don't get the ip adress in the 'Host" 
field.
 
Name = phone1 and phone2
Host=unspecified
mask 255.255.255.255 
port = 0
status = unmonitored
 
I can ping the two phone's and get a reply (also 
from the laptop)
phone ip adres 192.168.10.12 and 192.168.10.13 
(server 192.168.10.11and laptop 192.168.10.14)
hardware config: server - phone1 - phone2 - 
laptop
 
configurations used
 
 SIP.CONF
 
[phone1]
type=friend
host=dynamic
defaultip=192.168.10.12
dtmfmode=info
mailbox=1000
context=sip
callerid="phone1"<100>
 
[phone2]
type=friend
host=dynamic
defaultip=192.168.10.13
dtmfmode=info 
mailbox=1000
context=sip
callerid="phone2"<200>
 
EXTENSIONS.CONF
 
exten => 
100,1,dial(sip/phone1,20,tr)
exten => 
200,1,dial(sip/phone2,20,tr)
 
Wim
 
 

Re: [Asterisk-Users] Host unspecified ??

2003-10-30 Thread Wim Venneman 
Dear,

I changed the host to a fixed ip address (host1=192.168.10.12 and
host2=192.168.10.13) now the ip address shows up in the 'host' field = ok.
Try to call, no succes, nothing happens!

What's wrong?

Wim

- Original Message -
From: "Florian Overkamp" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, October 30, 2003 9:25 AM
Subject: Re: [Asterisk-Users] Host unspecified ??


> Hi Wim,
>
> It doesnt show the host (at least) until the phones have registered with
> asterisk, because you've set the host to dynamic in your config. Either
> verify if the phones register with asterisk, or set the host to their
> static IP-adresses.
>
> Best regards,
> Florian
>
> At 19:51 29-10-2003 +0100, you wrote:
> >When I start asterisk  -vvgrc and I ask 'sip show peers', I don't get
> >the ip adress in the 'Host" field.
> >
> >Name = phone1 and phone2
> >Host=unspecified
> >mask 255.255.255.255
> >port = 0
> >status = unmonitored
> >
> >I can ping the two phone's and get a reply (also from the laptop)
> >phone ip adres 192.168.10.12 and 192.168.10.13 (server 192.168.10.11and
> >laptop 192.168.10.14)
> >hardware config: server - phone1 - phone2 - laptop
> >
> >configurations used
> >
> >  SIP.CONF
> >
> >[phone1]
> >type=friend
> >host=dynamic
> >defaultip=192.168.10.12
> >dtmfmode=info
> >mailbox=1000
> >context=sip
> >callerid="phone1"<100>
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [Asterisk-Users] Host unspecified ??

2003-11-02 Thread Wim Venneman 
Hi,

Here is wath happens:

Asterisk*CLI>sip debug
SIP Debugging Enabled
Asterisk*CLI>




Nothing happens when I use 'sip debug'.
It seems that sip doesn't work.

Wim


- Original Message -
From: "Florian Overkamp" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, October 30, 2003 9:02 PM
Subject: Re: [Asterisk-Users] Host unspecified ??


> Hi Wim,
>
> Citeren Wim Venneman <[EMAIL PROTECTED]>:
>
> > I changed the host to a fixed ip address (host1=192.168.10.12 and
> > host2=192.168.10.13) now the ip address shows up in the 'host' field =
ok.
> > Try to call, no succes, nothing happens!
> >
> > What's wrong?
>
> That's a bit difficult to determine without more info. Could you enter the
> command 'sip debug', try calling with the phones and then copy what the
> console says ? Feel free to send off-list :-)
>
> --
> Met vriendelijke groet,
> Florian Overkamp
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

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[Asterisk-Users] Grandstream problem

2003-11-06 Thread Wim Venneman 



Hi,
 
I installed Asterisk an all works fine exept 
for Grandstream.
When I call with a softphone (ex X-ten) to a 
Grandstream (BudgetTone-100), I can make a conversation. = ok
When I call to a softphone with a Grandstream 
I can pich up the call with the softphone but the Grandstream keeps ringing like 
on the other site you didn't pick up the phone.(even if you do so)
It's the same when I call between two Grandstream 
phone's. Call from phone1 to phone 2, I pick up phone2 and afther 3 seconds I 
get congestion tone from both phone's.
 
Info from command *CLI>
-- Executing Dial("SIP/phone2-a030a", "sip/phone1") 
in new stack
-- Called phone1
-- SIP/phone1-663a is ringing
-- SIP/phone1-663a answered 
SIP/phone2-a030a
-- Attempting native bridge of SIP/phone2-a030a and 
SIP/phone1-663a
== Spawn extension (sip, 1,1)  exited non-zero 
on 'SIP/phone2-a030a'
 
and I get congestion
 
Can anyone give me a direction to solve my 
problem?
Thanks in advance,
 
Wim
 

Re: [Asterisk-Users] Grandstream problem

2003-11-07 Thread Wim Venneman 



Thanks William,
 
Works fine now.
 
Wim

  - Original Message - 
  From: 
  William Carlson 
  
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, November 06, 2003 9:43 
  PM
  Subject: Re: [Asterisk-Users] Grandstream 
  problem
  
  try 
  disallow=all
  allow=ulaw
   
  under the general section of 
sip.conf
   
  that half fixes it for me calls between phones 
  work but talking to asterisk has some problems.
  
- Original Message - 
From: 
Wim 
Venneman 
To: [EMAIL PROTECTED] 

Sent: Thursday, November 06, 2003 2:29 
PM
Subject: [Asterisk-Users] Grandstream 
problem

Hi,
 
I installed Asterisk an all works 
fine exept for Grandstream.
When I call with a softphone (ex X-ten) to a 
Grandstream (BudgetTone-100), I can make a conversation. = ok
When I call to a softphone with a 
Grandstream I can pich up the call with the softphone but the Grandstream 
keeps ringing like on the other site you didn't pick up the phone.(even if 
you do so)
It's the same when I call between two 
Grandstream phone's. Call from phone1 to phone 2, I pick up phone2 and 
afther 3 seconds I get congestion tone from both phone's.
 
Info from command *CLI>
-- Executing Dial("SIP/phone2-a030a", 
"sip/phone1") in new stack
-- Called phone1
-- SIP/phone1-663a is ringing
-- SIP/phone1-663a answered 
SIP/phone2-a030a
-- Attempting native bridge of SIP/phone2-a030a 
and SIP/phone1-663a
== Spawn extension (sip, 1,1)  exited 
non-zero on 'SIP/phone2-a030a'
 
and I get congestion
 
Can anyone give me a direction to solve my 
problem?
Thanks in advance,
 
Wim
 

[Asterisk-Users] SIP (peer to peer?)

2003-12-08 Thread Wim Venneman 



 Hi 
all,Has anyone have an idea why, if you capture the files on a Asterisk 
network (ex with Ethereal) you always see the communication between the two sip 
phones( hard or soft) passing through the asterisk server (on UDP 
layer)
Isn't SIP 
a protocol that (after that it has established the call) , he connects the two 
users with each other?
 
Maybe a 
stupid question, but I'm not a SIP expert.
 
Thank you 
for your help.
 
Wim

[Asterisk-Users] Dial plan

2003-12-17 Thread Wim Venneman 



Hi,
 
For our study we are searching for a (working) dial 
plan that is used in a Cisco call manager or other PBX, to compare it with 
Asterisk.
If anyone has a (simple) working example, can I 
have a copy?
 
Thanks

Re: [Asterisk-Users] Re: Analogue telephone cards for the UK

2004-04-10 Thread Wim Venneman 



Paul,
 
http://www.voip-info.org/wiki-Asterisk
 
Wim

  - Original Message - 
  From: 
  Paul 
  Tyreman 
  To: [EMAIL PROTECTED] 
  
  Sent: Saturday, April 10, 2004 6:47 
  PM
  Subject: Re: [Asterisk-Users] Re: 
  Analogue telephone cards for the UK
  
  Sorry to sound stupid, but where can I get copied 
  of the Asterisk manual ?
   
  What is the VoIP wiki and where can I get that 
  too ?
  Thanks, Paul.
   
   
   
  -Original Message-From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]] 
  On Behalf Of Iain StevensonPosted At: 10 April 2004 17:20Posted To: 
  Asterisk-UsersConversation: [Asterisk-Users] Re: Analogue telephone cards 
  for the UKSubject: Re: [Asterisk-Users] Re: Analogue telephone cards for 
  the UK
   
  ...   Most is explained in the asterisk 
  manual or on the VoIP wiki.
   
    Iain

Re: [Asterisk-Users] sip software

2004-04-14 Thread Wim Venneman 
www.xten.com

http://www.sjlabs.com/


- Original Message - 
From: "James Moran" <[EMAIL PROTECTED]>
To: "Asterisk" <[EMAIL PROTECTED]>
Sent: Wednesday, April 14, 2004 8:52 PM
Subject: [Asterisk-Users] sip software


> Anyone have any suggestions on free sip phone software for windows??
> Only have one IP phone and want to have one other computer hooked up to
> my Asterisk box for testing.
> 
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