date:20050930




I've been seeing ppl talking about * fax, anyone 
got it successful, which means in good quality?

is it done by T.38? Is spandsp rxfax() a T.38 
implementation or where to find some * T.38 fax modules or code that can be 
written into *?

Cheers!

Matt

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[Asterisk-Users] Siemens TC35 GSM gateway

2005-09-30 Thread Andrew Smith


Hi all,

I have a TC35 and am keen to see if anyone has both voice and sms working from 
Asterisk through this device?  Google tells me that a few people have theorised 
about it, I can't find anyone claiming to be doing it.  What would be the best 
way to put it into practice?  Build a new channel for it?


Thanks
Andrew
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[Asterisk-Users] G.729 patent in France

2005-09-30 Thread Amaury BOSSE









Hi all,

I am building an Asterisk PBX with voicemail and
music on hold functions.

An ISDN BRI line will also be available and G.729
IP-phones will be used.



Are there patents rights applicable to France?



Which licence could I use and how many ones are
required (only one per phone or also for voicemail and MOH)?



Regards 

Amaury






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Re: [Asterisk-Users] G.729 patent in France

2005-09-30 Thread Steve Kennedy

On Fri, Sep 30, 2005 at 10:19:59AM +0200, Amaury BOSSE wrote:

Are there patents rights applicable to France?

Yes, most of the world.

Which licence could I use and how many ones are required (only one per
phone or also for voicemail and MOH)?

One per translating service (concurrent use). Digium sell them.

Steve

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Re: [Asterisk-Users] Fritz, mISDN, Help

I big problem (for me) is that it seems that the AVM Fritz CAPI drivers 
cannot support more than one PCI card per server.  There is a hack out 
there for multiple cards and a 2.4.XX kernel but this is not supported 
for the 2.6.XX kernels.


mISDN on the other hand does support multiple AVM Fritz cards but I 
couldn't seem to get mISDN - CAPI - chan_capi working right (I tried 
for hours and I can't remember what exactly the problem was now - 
everything did load OK but calls were not being received correctly I 
think).  I didn't try using chan_misdn.


Now I'm running a solution which does not make me happy at all - I've 
got a second asterisk box beside the main one for the other AVM Fritz 
ISDN card and sending the received calls over via. IAX2 to the primary 
server (I shudder when I think of the waste of electricity).


Is anyone out there running two AVM Fritz ISDN cards?  Are you using a 
2.6.XX kernel?  How are you doing it?


Thanks,

Derek



Konrads Smelkovs wrote:


Unfourtunatley, mISDN  is far from production quality. So going
miSDN-CAPI-chan_capi might not work. chan_misdn is even more flakey
at the moment.

Your best option is to use just CAPI and  chan_capi,it had support for fritz

On 10/09/05, Jon Dean [EMAIL PROTECTED] wrote:
 


A plea to all!

Has anyone had any success with two or more avm fritz pci cards with either
misdn, chan_misdn, or chan_capi, and any version of linux 2.6.x?

I have managed to get misdn to load under 2.6.13 and detect two cards using
misdn-capi and chan-capi (using capiinfo and capi info under asterisk) - but
the second card/controller doesn't answer or dial calls.

But if I try misdn without capi I get the following error

mISDN: INTERNAL ERROR in drivers/isdn/hardware/mISDN/stack.c:596


Any help would be greatly appreciated.

Jon
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[Asterisk-Users] IAXPhone

2005-09-30 Thread Erwan DESVERGNES









Hi,



Im looking for IAXphone 2.0 (from
sokol-associates) source code but the site is unavailable did some one can help
me please. ??



thanks



_

Erwan
 Desvergnes - ANDIUM -

82/86 rue Château Gaillard

69100 Villeurbanne



Tel. 04 3743 44 45
/ Fax 04 37 43 44 44

E-mail: [EMAIL PROTECTED]








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[Asterisk-Users] Empty ACK

2005-09-30 Thread Ronald Voermans

Hello,

I have asterisk connected to SER/RTPProxy which is again connected to a
IP-PSTN gateway. When calling with a UA, registered at * to a SIP phone
connected to the IP-PSTN gateway, I get 'empty ACKs':

U 192.168.0.173:5060 - 10.254.254.1:5060
ACK  SIP/2.0.
Via: SIP/2.0/UDP 192.168.0.173:5060;branch=z9hG4bK5cb7d048.
Route: sip:[EMAIL PROTECTED]:5060,sip:212.241.48.70:5060.
From: 0161801019 sip:[EMAIL PROTECTED];tag=as628d39c1.
To: sip:[EMAIL PROTECTED];tag=00-04094-52dc5953-7c1293c27.
Contact: sip:[EMAIL PROTECTED].
Call-ID: [EMAIL PROTECTED]
CSeq: 103 ACK.
User-Agent: Asterisk PBX.
Content-Length: 0.

As you can see, there is no URI after the ACK statement, and SER doesn't
know what to do with it. Is this a bug in *, or is this normal?

Regards,

Ronald
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[Asterisk-Users] Zaptel TDM questions

2005-09-30 Thread Chee Foong




Hello,

I have a 
TDM04B. I make a call into the first port of the card. Once asterisk receive the 
call, it will make another call out using the second port. 

From what i have 
observerd as soon as the called party on the second port starts ringing asterisk 
show the following :

-- Zap/2-1 answered 
Zap/1-1

Any idea why 
asterisk thinks the call has been answered while actually the phone is still 
ringing?

Anybody know how to 
avoid asterisk to answer the call while ringing? 
Also, I have no 
Answer or any Playbackcommand in the dial plan before making a call out of 
second port. I have also try setting callprogress to yes/no but the results are 
the same.

Thanks


CCF
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[Asterisk-Users] Don't call

I receive a call, but don't call...
Asterisk show this message.
Are codecs the problem?

Sep 30 11:25:54 WARNING[4475]: chan_sip.c:1899
create_addr: No such host: sip.uni.it,r
Sep 30 11:25:54 NOTICE[4475]: app_dial.c:1109
dial_exec_full: Unable to create channel of type 'SIP'
(cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)






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[Fwd: [Asterisk-Users] TDM40B - Unable to play dialtone on channel X ?]


Hi everyone,

Sorry for forwarding and top-posting this email again but its as if my 
TDM40b has keeled over yesterday.  After a few hours last night and 
swapping the card to another asterisk server (with exactly the same 
result) I needed to have the FXS ports working ASAP this morning so I 
have repaced the functionality of the TDM40b with some Grandstream 
handytones which I already had in stock.


I purchased the TDM40b directly from Digium - I'll check and see if the 
cards have a one year warranty.  I'm email the list again in case the 
problem is obvious (many times I've spent a long time looking at an 
obvious problem).


Thanks,

Derek

- original message  -
Today my TDM40B (a TDM400 with 4 FXS modules) has gone funny - there is 
no dialtone from any port.


When I look at the CLI display in * and pick up a line it says Sep 29 
20:36:25 WARNING[1093299120]: chan_zap.c:5313 handle_init_event: Unable 
to play dialtone on channel 3 and it does this on, and gives this 
message for, every channel.


Its a bit weird because I have not changed the configuration of asterisk 
at all and cables were not even unplugged.


Does anyone know why this is happening?

thanks,

Derek

--
Derek Conniffe
Rivertower Ltd
DID Number: 01 440 1806 (International: 00 353 1 440 1806)
Ireland: (Freephone) 1800 719 400
Ireland: (Local) 01 440 1800
United Kingdom: 0870 068 2368
International: 00 353 1 440 1800
Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823)
Fax: 01 201 0085 (International: 00 353 1 201 0085)
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--
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Rivertower Ltd
DID Number: 01 440 1806 (International: 00 353 1 440 1806)
Ireland: (Freephone) 1800 719 400
Ireland: (Local) 01 440 1800
United Kingdom: 0870 068 2368
International: 00 353 1 440 1800
Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823)
Fax: 01 201 0085 (International: 00 353 1 201 0085)
Email: [EMAIL PROTECTED]
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RE: [Asterisk-Users] Fritz, mISDN, Help

2005-09-30 Thread Jörg Wolf

Derek,

I've got up to 3 Fritz cards (of different types) up and running on my debian 
sarge (2.6.8 kernel).

That's how it should work:
- build and install mISDN and chan_misdn from the current install-misdn.tar.gz 
which can be downloaded from http://www.beronet.com/download/
- use the chan_misdn and change /etc/asterisk/misdn.conf according to your 
special config
- be sure that no capi modules are loaded
- do /etc/init.d/misdn-init config to automatically create 
/etc/misdn-init.conf
- start the cards: /etc/init.d/misdn-init start
- start your * and check if chan_misdn.so started properly

Hope it helps

Cheers
Jörg 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Derek Conniffe
 Sent: Friday, September 30, 2005 10:39 AM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
 Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Fritz, mISDN, Help
 
 I big problem (for me) is that it seems that the AVM Fritz 
 CAPI drivers cannot support more than one PCI card per 
 server.  There is a hack out there for multiple cards and a 
 2.4.XX kernel but this is not supported for the 2.6.XX kernels.
 
 mISDN on the other hand does support multiple AVM Fritz cards 
 but I couldn't seem to get mISDN - CAPI - chan_capi working 
 right (I tried for hours and I can't remember what exactly 
 the problem was now - everything did load OK but calls were 
 not being received correctly I think).  I didn't try using chan_misdn.
 
 Now I'm running a solution which does not make me happy at 
 all - I've got a second asterisk box beside the main one for 
 the other AVM Fritz ISDN card and sending the received calls 
 over via. IAX2 to the primary server (I shudder when I think 
 of the waste of electricity).
 
 Is anyone out there running two AVM Fritz ISDN cards?  Are 
 you using a 2.6.XX kernel?  How are you doing it?
 
 Thanks,
 
 Derek
 
 
 
 Konrads Smelkovs wrote:
 
 Unfourtunatley, mISDN  is far from production quality. So going
 miSDN-CAPI-chan_capi might not work. chan_misdn is even more flakey
 at the moment.
 
 Your best option is to use just CAPI and  chan_capi,it had 
 support for 
 fritz
 
 On 10/09/05, Jon Dean [EMAIL PROTECTED] wrote:
   
 
 A plea to all!
 
 Has anyone had any success with two or more avm fritz pci 
 cards with 
 either misdn, chan_misdn, or chan_capi, and any version of 
 linux 2.6.x?
 
 I have managed to get misdn to load under 2.6.13 and detect 
 two cards 
 using misdn-capi and chan-capi (using capiinfo and capi info under 
 asterisk) - but the second card/controller doesn't answer 
 or dial calls.
 
 But if I try misdn without capi I get the following error
 
 mISDN: INTERNAL ERROR in drivers/isdn/hardware/mISDN/stack.c:596
 
 
 Any help would be greatly appreciated.
 
 Jon
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 http://lists.digium.com/mailman/listinfo/asterisk-users
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 --
 Derek Conniffe
 Rivertower Ltd
 DID Number: 01 440 1806 (International: 00 353 1 440 1806)
 Ireland: (Freephone) 1800 719 400
 Ireland: (Local) 01 440 1800
 United Kingdom: 0870 068 2368
 International: 00 353 1 440 1800
 Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 
 856 3823)
 Fax: 01 201 0085 (International: 00 353 1 201 0085)
 Email: [EMAIL PROTECTED]
 Web: http://www.rivertowerhosting.com
 
 
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Re: [Asterisk-Users] Fritz, mISDN, Help


Hi Jorg,

Thanks very much for the information - I'll give it a try over the weekend.

Derek

Jörg Wolf wrote:


Derek,

I've got up to 3 Fritz cards (of different types) up and running on my debian 
sarge (2.6.8 kernel).

That's how it should work:
- build and install mISDN and chan_misdn from the current install-misdn.tar.gz 
which can be downloaded from http://www.beronet.com/download/
- use the chan_misdn and change /etc/asterisk/misdn.conf according to your 
special config
- be sure that no capi modules are loaded
- do /etc/init.d/misdn-init config to automatically create 
/etc/misdn-init.conf
- start the cards: /etc/init.d/misdn-init start
- start your * and check if chan_misdn.so started properly

Hope it helps

Cheers
Jörg	 

 


-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Derek Conniffe

Sent: Friday, September 30, 2005 10:39 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
Non-Commercial Discussion

Subject: Re: [Asterisk-Users] Fritz, mISDN, Help

I big problem (for me) is that it seems that the AVM Fritz 
CAPI drivers cannot support more than one PCI card per 
server.  There is a hack out there for multiple cards and a 
2.4.XX kernel but this is not supported for the 2.6.XX kernels.


mISDN on the other hand does support multiple AVM Fritz cards 
but I couldn't seem to get mISDN - CAPI - chan_capi working 
right (I tried for hours and I can't remember what exactly 
the problem was now - everything did load OK but calls were 
not being received correctly I think).  I didn't try using chan_misdn.


Now I'm running a solution which does not make me happy at 
all - I've got a second asterisk box beside the main one for 
the other AVM Fritz ISDN card and sending the received calls 
over via. IAX2 to the primary server (I shudder when I think 
of the waste of electricity).


Is anyone out there running two AVM Fritz ISDN cards?  Are 
you using a 2.6.XX kernel?  How are you doing it?


Thanks,

Derek



Konrads Smelkovs wrote:

   


Unfourtunatley, mISDN  is far from production quality. So going
miSDN-CAPI-chan_capi might not work. chan_misdn is even more flakey
at the moment.

Your best option is to use just CAPI and  chan_capi,it had 
 

support for 
   


fritz

On 10/09/05, Jon Dean [EMAIL PROTECTED] wrote:


 


A plea to all!

Has anyone had any success with two or more avm fritz pci 
   

cards with 
   

either misdn, chan_misdn, or chan_capi, and any version of 
   


linux 2.6.x?
   

I have managed to get misdn to load under 2.6.13 and detect 
   

two cards 
   

using misdn-capi and chan-capi (using capiinfo and capi info under 
asterisk) - but the second card/controller doesn't answer 
   


or dial calls.
   


But if I try misdn without capi I get the following error

mISDN: INTERNAL ERROR in drivers/isdn/hardware/mISDN/stack.c:596


Any help would be greatly appreciated.

Jon
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--
Derek Conniffe
Rivertower Ltd
DID Number: 01 440 1806 (International: 00 353 1 440 1806)
Ireland: (Freephone) 1800 719 400
Ireland: (Local) 01 440 1800
United Kingdom: 0870 068 2368
International: 00 353 1 440 1800
Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 
856 3823)

Fax: 01 201 0085 (International: 00 353 1 201 0085)
Email: [EMAIL PROTECTED]
Web: http://www.rivertowerhosting.com


   


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--
Derek Conniffe
Rivertower Ltd
DID Number: 01 440 1806 (International: 00 353 1 440 1806)
Ireland: (Freephone) 1800 719 400
Ireland: (Local) 01 440 1800
United Kingdom: 0870 068 2368
International: 00 353 1 440 1800
Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823)
Fax: 01 201 0085 (International: 00 353 1 201 0085)
Email: [EMAIL PROTECTED]
Web: http://www.rivertowerhosting.com

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n:Conniffe;Derek
org:Rivertower Ltd;IT
adr:Dublin 2;;46 Upper Mount Street;Dublin;Dublin;Dublin 2;Ireland
email;internet:[EMAIL PROTECTED]
tel;work:+353 1 201 0146
tel;fax:+353 1 201 0085
tel;cell:+353 86 856 3823
note;quoted-printable:Ireland: (Freephone) 1800 719 400=0D=0A=
	Ireland: (Local) 01 244 9719=0D=0A=
	United Kingdom: 0870 068 2368=0D=0A=
	International: 00 353 1 244 9719=0D=0A=
	
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[Asterisk-Users] Compile broken on FreeBSD ?

2005-09-30 Thread Kim Culhan


I'm seeing this trying to compile on FreeBSD with source via cvs from cvs.digium.com
at ~1000 UTC 6-30:

func_enum.c: In function `function_enum':func_enum.c:126: error: too many arguments to function `ast_get_enum'gmake[1]: *** [func_enum.o] Error 1

Also, the cvsup server on cvs.digium.com has been refusing connections for
some time.

Is cvsup no longer available on this server ?

regards
-kim

--
w8hdkim er gmail.com


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Re: [Asterisk-Users] Fritz, mISDN, Help


Hi again Jörg,

The install_misdn Makefile doesn't seem to like my SMP yes in my 
kernel .config (it does a grep on the .config file, finds the line, and 
tells me) - I'm going to try the chan_misdn driver anyway and the server 
is an old HP netserver e800 which is a dual processor PIII board.  
Actually I'm only running one processor in it because, for reasons 
unknown to me, SuSE linux (9.1) grinds to almost a halt with two 
processors on board but I'd say I need to SMP option on in the kernel 
config in order to use this motherboard (although I'm not against 
changing the motherboard if necessary).


I'll let you know how I get on,

Derek

Jörg Wolf wrote:


Derek,

I've got up to 3 Fritz cards (of different types) up and running on my debian 
sarge (2.6.8 kernel).

That's how it should work:
- build and install mISDN and chan_misdn from the current install-misdn.tar.gz 
which can be downloaded from http://www.beronet.com/download/
- use the chan_misdn and change /etc/asterisk/misdn.conf according to your 
special config
- be sure that no capi modules are loaded
- do /etc/init.d/misdn-init config to automatically create 
/etc/misdn-init.conf
- start the cards: /etc/init.d/misdn-init start
- start your * and check if chan_misdn.so started properly

Hope it helps

Cheers
Jörg	 

 


-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Derek Conniffe

Sent: Friday, September 30, 2005 10:39 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
Non-Commercial Discussion

Subject: Re: [Asterisk-Users] Fritz, mISDN, Help

I big problem (for me) is that it seems that the AVM Fritz 
CAPI drivers cannot support more than one PCI card per 
server.  There is a hack out there for multiple cards and a 
2.4.XX kernel but this is not supported for the 2.6.XX kernels.


mISDN on the other hand does support multiple AVM Fritz cards 
but I couldn't seem to get mISDN - CAPI - chan_capi working 
right (I tried for hours and I can't remember what exactly 
the problem was now - everything did load OK but calls were 
not being received correctly I think).  I didn't try using chan_misdn.


Now I'm running a solution which does not make me happy at 
all - I've got a second asterisk box beside the main one for 
the other AVM Fritz ISDN card and sending the received calls 
over via. IAX2 to the primary server (I shudder when I think 
of the waste of electricity).


Is anyone out there running two AVM Fritz ISDN cards?  Are 
you using a 2.6.XX kernel?  How are you doing it?


Thanks,

Derek



Konrads Smelkovs wrote:

   


Unfourtunatley, mISDN  is far from production quality. So going
miSDN-CAPI-chan_capi might not work. chan_misdn is even more flakey
at the moment.

Your best option is to use just CAPI and  chan_capi,it had 
 

support for 
   


fritz

On 10/09/05, Jon Dean [EMAIL PROTECTED] wrote:


 


A plea to all!

Has anyone had any success with two or more avm fritz pci 
   

cards with 
   

either misdn, chan_misdn, or chan_capi, and any version of 
   


linux 2.6.x?
   

I have managed to get misdn to load under 2.6.13 and detect 
   

two cards 
   

using misdn-capi and chan-capi (using capiinfo and capi info under 
asterisk) - but the second card/controller doesn't answer 
   


or dial calls.
   


But if I try misdn without capi I get the following error

mISDN: INTERNAL ERROR in drivers/isdn/hardware/mISDN/stack.c:596


Any help would be greatly appreciated.

Jon
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Ireland: (Freephone) 1800 719 400
Ireland: (Local) 01 440 1800
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International: 00 353 1 440 1800
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856 3823)

Fax: 01 201 0085 (International: 00 353 1 201 0085)
Email: [EMAIL PROTECTED]
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--
Derek Conniffe
Rivertower Ltd
DID Number: 01 440 1806 (International: 00 353 1 440 1806)
Ireland: (Freephone) 1800 719 400
Ireland: (Local) 01 440 1800
United Kingdom: 0870 068 2368
International: 00 353 1 440 1800
Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86 856 3823)
Fax: 01 201 0085 (International: 00 353 1 201 0085)
Email: [EMAIL PROTECTED]
Web: http://www.rivertowerhosting.com

begin:vcard
fn:Derek

RE: [Asterisk-Users] Fritz, mISDN, Help

2005-09-30 Thread Jörg Wolf

Hi Derek,

well, don't know anything about the SMP warning, in my case there's a simliar 
warning regarding PREEMPTIBLE setting which I simply ignored...

You might also have a look onto this: http://bugs.digium.com/view.php?id=4077

Crossing fingers...

cheers
Jörg


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Derek Conniffe
 Sent: Friday, September 30, 2005 12:48 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Fritz, mISDN, Help
 
 Hi again Jörg,
 
 The install_misdn Makefile doesn't seem to like my SMP yes 
 in my kernel .config (it does a grep on the .config file, 
 finds the line, and tells me) - I'm going to try the 
 chan_misdn driver anyway and the server is an old HP 
 netserver e800 which is a dual processor PIII board.  
 Actually I'm only running one processor in it because, for 
 reasons unknown to me, SuSE linux (9.1) grinds to almost a 
 halt with two processors on board but I'd say I need to SMP 
 option on in the kernel config in order to use this 
 motherboard (although I'm not against changing the 
 motherboard if necessary).
 
 I'll let you know how I get on,
 
 Derek
 
 Jörg Wolf wrote:
 
 Derek,
 
 I've got up to 3 Fritz cards (of different types) up and 
 running on my debian sarge (2.6.8 kernel).
 
 That's how it should work:
 - build and install mISDN and chan_misdn from the current 
 install-misdn.tar.gz which can be downloaded from 
 http://www.beronet.com/download/
 - use the chan_misdn and change /etc/asterisk/misdn.conf 
 according to 
 your special config
 - be sure that no capi modules are loaded
 - do /etc/init.d/misdn-init config to automatically create 
 /etc/misdn-init.conf
 - start the cards: /etc/init.d/misdn-init start
 - start your * and check if chan_misdn.so started properly
 
 Hope it helps
 
 Cheers
 Jörg  
 
   
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Derek 
 Conniffe
 Sent: Friday, September 30, 2005 10:39 AM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
 Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Fritz, mISDN, Help
 
 I big problem (for me) is that it seems that the AVM Fritz CAPI 
 drivers cannot support more than one PCI card per server.  
 There is a 
 hack out there for multiple cards and a 2.4.XX kernel but 
 this is not 
 supported for the 2.6.XX kernels.
 
 mISDN on the other hand does support multiple AVM Fritz cards but I 
 couldn't seem to get mISDN - CAPI - chan_capi working 
 right (I tried 
 for hours and I can't remember what exactly the problem was now - 
 everything did load OK but calls were not being received 
 correctly I 
 think).  I didn't try using chan_misdn.
 
 Now I'm running a solution which does not make me happy at 
 all - I've 
 got a second asterisk box beside the main one for the other 
 AVM Fritz 
 ISDN card and sending the received calls over via. IAX2 to 
 the primary 
 server (I shudder when I think of the waste of electricity).
 
 Is anyone out there running two AVM Fritz ISDN cards?  Are 
 you using a 
 2.6.XX kernel?  How are you doing it?
 
 Thanks,
 
 Derek
 
 
 
 Konrads Smelkovs wrote:
 
 
 
 Unfourtunatley, mISDN  is far from production quality. So going
 miSDN-CAPI-chan_capi might not work. chan_misdn is even 
 more flakey
 at the moment.
 
 Your best option is to use just CAPI and  chan_capi,it had
   
 
 support for
 
 
 fritz
 
 On 10/09/05, Jon Dean [EMAIL PROTECTED] wrote:
  
 
   
 
 A plea to all!
 
 Has anyone had any success with two or more avm fritz pci
 
 
 cards with
 
 
 either misdn, chan_misdn, or chan_capi, and any version of
 
 
 linux 2.6.x?
 
 
 I have managed to get misdn to load under 2.6.13 and detect
 
 
 two cards
 
 
 using misdn-capi and chan-capi (using capiinfo and capi info under
 asterisk) - but the second card/controller doesn't answer
 
 
 or dial calls.
 
 
 But if I try misdn without capi I get the following error
 
 mISDN: INTERNAL ERROR in drivers/isdn/hardware/mISDN/stack.c:596
 
 
 Any help would be greatly appreciated.
 
 Jon
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 --
 Derek Conniffe
 Rivertower Ltd
 DID Number: 01 440 1806 (International: 00 353 1 440 1806)
 Ireland: (Freephone) 1800 719 400
 Ireland: (Local) 01 440 1800
 United Kingdom: 0870 068 2368
 International: 00 353 1 440 1800
 Derek Conniffe Mobile: 086 856 3823 (International: 00 353 86
 856 3823)
 Fax: 01 201 0085 (International: 00 353 1 201 0085)
 Email: [EMAIL PROTECTED]
 Web: http://www.rivertowerhosting.com

[Asterisk-Users] Diva

2005-09-30 Thread Giordano Grandis









Hi all,

just a question: can i use this kind of
diva for asterisk?



00:14.0 Network controller: Eicon Networks
Corporation Diva ISDN Pro 3.0 PCI



Thanks all



Giordano 








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Re: [Asterisk-Users] chan_cap-cm-0.6 deflect support

2005-09-30 Thread Armin Schindler

On Thu, 29 Sep 2005, Louis-David Mitterrand wrote:
 Hi,
 
 I've recently reinstalled a Diva in my asterisk server (alongside a
 QuadBRI :-) to test the nice features Armin has been adding in
 chan_capi.
 
 The capi.conf format has changed, so my question is how do I define a
 deflect= statement for different incoming MSN's?
 
 I've tried to define a section for each (group of) MSN with a different
 deflect. Is that correct?
 
 [DIVA1]
 isdnmode=msn
 incomingmsn=146472130
 controller=1
 group=5
 accountcode=diva
 context=default
 deflect=0612110618
 devices=2
...

What exactly do you want to do?
You can use capicommand(deflect|) in extensions.conf to use call 
deflection.
 
 Also, is there a way to detect that a SIP phone has an active forward
 number and capi-deflect any incoming calls to that number?

If you can retrieve this information from extensions.conf, then you can use 
my example above.

Anyway, I noticed that the original implementation of deflect specified in
capi.conf does not work in all cases.
I plan to remove that and to allow capicommand(deflect|...) only.
It's not necessary to do that in capi.conf and using different MSNs is 
difficult too.
My idea is provide information about 'this is a call-waiting call, no 
b-channel' to extensions.conf via a variable. And the user then can decide
what to do with that call using all features of the dialplan.
I plan to do this for version 0.6.1.

Armin
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[Asterisk-Users] Why does the s extension not work in my extensions.conf file


Hello

In my extensions.conf file:

[frompstnisdn]
exten = s,1,Dial(SIP/200SIP/202,20)
exten = s,2,Voicemail(su200)
exten = s,3,Hangup

I use the s, start, extension to handle incoming calls.

In my zapata.conf:
context=frompstnisdn


This works ok on another asterisk box I setup.  But on incoming calls I get:

   -- Extension '787367' in context 'frompstnisdn' from '07768385144' does 
not exist.  Rejecting call on channel 0/1, span 1

   -- Saved useragent X-Lite release 1103m for peer 202
   -- Extension '787367' in context 'frompstnisdn' from '07768385144' does 
not exist.  Rejecting call on channel 0/1, span 1


Do I need to enable something to be able to use the s in extensions.conf?

Angus





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Re: [Asterisk-Users] Diva

2005-09-30 Thread Armin Schindler

On Fri, 30 Sep 2005, Giordano Grandis wrote:
 Hi all,
 
 just a question:   can i use this kind of diva for asterisk?
 
  
 
 00:14.0 Network controller: Eicon Networks Corporation Diva ISDN Pro 3.0
 PCI


No, as far as I know the 'Pro' versions of Diva card are not supported by 
any driver in Linux yet.

Armin

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Re: [Asterisk-Users] Zaptel TDM questions




I think the Asterisk must answer the call to be 
able to then dial out on the second port. This is what happens on any 
other PBX I have worked with in this sort of scenario. Is this a problem 
for you?

Angus



  - Original Message - 
  From: 
  Chee 
  Foong 
  To: asterisk-users@lists.digium.com 
  
  Sent: Friday, September 30, 2005 10:20 
  AM
  Subject: [Asterisk-Users] Zaptel TDM 
  questions
  
  Hello,
  
  I have a 
  TDM04B. I make a call into the first port of the card. Once asterisk receive 
  the call, it will make another call out using the second port. 
  
  From what i have 
  observerd as soon as the called party on the second port starts ringing 
  asterisk show the following :
  
  -- Zap/2-1 
  answered Zap/1-1
  
  Any idea why 
  asterisk thinks the call has been answered while actually the phone is still 
  ringing?
  
  Anybody know how 
  to avoid asterisk to answer the call while ringing? 
  Also, I have no 
  Answer or any Playbackcommand in the dial plan before making a call out 
  of second port. I have also try setting callprogress to yes/no but the results 
  are the same.
  
  Thanks
  
  
  CCF
  
  

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[Asterisk-Users] One-way audio with VPN

2005-09-30 Thread Alex Lake

Thanks for that. We did track it down to a problem with native bridging. 
In this case, Asterisk assumed that the VPN was publicly accessible - 
but it isn't!


The fix we've found is to setup all VPN-based sip devices with 
canreinvite=no, but I'm not sure if this is the best way to do that.

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[Asterisk-Users] chan_capi-0.3.5

2005-09-30 Thread Giordano Grandis









Hi all,

im tryinf to install chan_capi but i get this
error



[EMAIL PROTECTED]:/usr/src/chan_capi-0.3.5# make

gcc -pipe -Wall -Wmissing-prototypes
-Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6
-march=i586 -DNEVER_EVER_EARLY_B3_CONNECTS -DCAPI_ES -DCAPI_GAIN
-DDEFLECT_ON_CIRCUITBUSY -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes
-Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c

chan_capi.c:36:20: capi20.h: No such file or
directory

In file included from chan_capi.c:39



Anyone cha help me?



Thanks



Giordano








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[Asterisk-Users] Register times out on internet outage

2005-09-30 Thread Chris Mason (Lists)


I am using AstLinux with
Asterisk CVS-HEAD-01/10/05-02:11:15-AstLinux built by [EMAIL PROTECTED] on a i686 
running Linux


On this box I am registered to two different providers for long distance 
and international. If there is an internet outage of more than a few 
minutes, I'm not sure how long it takes to make this happen, the 
registration times out and when internet comes back the pbx never 
re-registers and consequently Asterisk has to be reset. As there is 
no-one there that can access the PBX it means the PBX has to be power 
cycled, not an optimal solution.

Has anyone seen this problem, is it a CVS-HEAD problem?

--
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NetConcepts
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Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


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[Asterisk-Users] VideoConference with UMTS

2005-09-30 Thread Sergio Serrano




Hi 
Srs.,

 Do you know if it's possible make a videocall from asterisk to UMTS 
mobile phone?. Both technologies use H.263 like videocodec.


Any 
idea?
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RE: [Asterisk-Users] Don't call

2005-09-30 Thread Jason Walker




It looks like your * server is not able to see the destination 
(presumably sip.uni.it).No route to 
destination


-Original Message-From: 
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] 
On Behalf Of Fabio MontemaggioreSent: Friday, September 30, 2005 2:34 
AMTo: asteriskSubject: [Asterisk-Users] Don't callI receive a 
call, but don't call...Asterisk show this message.Are codecs the 
problem?Sep 30 11:25:54 WARNING[4475]: chan_sip.c:1899create_addr: 
No such host: sip.uni.it,rSep 30 11:25:54 NOTICE[4475]: 
app_dial.c:1109dial_exec_full: Unable to create channel of type 
'SIP'(cause 3 - No route to destination) == Everyone is 
busy/congested at this time 
(1:0/0/1) 
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RE: [Asterisk-Users] Don't call

2005-09-30 Thread Andreas Sikkema

[EMAIL PROTECTED] wrote:

 I receive a call, but don't call...
 Asterisk show this message.
 Are codecs the problem?
 
 Sep 30 11:25:54 WARNING[4475]: chan_sip.c:1899
 create_addr: No such host: sip.uni.it,r

If you pasted this directly from Asterisk, then 
there's an error in your configuration somewhere.

Host names cannot contain , characters.

-- 
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Re: [Asterisk-Users] Register times out on internet outage

2005-09-30 Thread brett

On 9/30/2005, Chris Mason (Lists) [EMAIL PROTECTED] wrote:

 I am using AstLinux with
 Asterisk CVS-HEAD-01/10/05-02:11:15-AstLinux built by [EMAIL PROTECTED] on a 
 i686
 running Linux

 On this box I am registered to two different providers for long distance
 and international. If there is an internet outage of more than a few
 minutes, I'm not sure how long it takes to make this happen, the
 registration times out and when internet comes back the pbx never
 re-registers and consequently Asterisk has to be reset. As there is
 no-one there that can access the PBX it means the PBX has to be power
 cycled, not an optimal solution.
 Has anyone seen this problem, is it a CVS-HEAD problem?

Chris - just do a reload.  For my SIP stuff it comes back with a warning
and I don't think it says anything for IAX.

It times out after the 10th try (I think) and you should get notices in
message file.

Reload has always done it for me.

Brett
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RE: [Asterisk-Users] OOH323C

2005-09-30 Thread brett

On 9/30/2005, Dan Austin [EMAIL PROTECTED] wrote:

 Asking which H323 channel is the best turns out to be a deeply
 personal issue, at least noting the responses in the past.

You got that right! 8-)

 I've tried and used all three. Here are my thoughts-

 Chan_h323 (the original)-
 Did not work in our environment.  Known issues with Cisco's
 Call Manager.  Other than the requirements for OpenH323 and
 PWLib, it was easy to setup and configure.

 Chan_oh323
 Worked fine for us.  Has the same dependencies as chan_h323,
 also easy to setup and configure.

 Chan_h323 (ooh323c based)
 This one has been a winner for us.  No dependencies on OpenH323
 or PWLib, which while not terrible to build/setup, is extra effort
 and can be tricky to match known working versions.
 Setup and configuration has been very simple.  If you have configured
 the other channels, this one should seem familiar.

 A seperate note in favor of the new chan_h323 is the developer support.
 I found a couple little bugs that related to our use of Cisco Call 
 Manager, and expected little or no interest in getting them resolved.
 I had a test version made available to me in just over a day and
 complete resolution a few hours later.

Dan - as a thought - I am messing with a H323 'capable' IP Phone and I
am (maybe foolishly) trying to use ooh323 with no gateway, gatekeeper,
or anything else and I am not getting it to work too well.  It seems
'sometimes' it does work.

Is there any way - as far as you (or anyone else) knows that this will
work with any flavor of H323 on Asterisk?  I could just be messing up
the configs.

Brett
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RE: [Asterisk-Users] chan_capi-0.3.5

2005-09-30 Thread Jörg Wolf




Giordano,

you simply don't have capi installed...

On debian sarge you can install the following 
packages:
 - capiutils
 - libcapi20-dev

Hope it helps
cheers
Jörg

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Giordano 
  GrandisSent: Friday, September 30, 2005 1:37 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
  [Asterisk-Users] chan_capi-0.3.5
  
  
  Hi 
  all,
  im tryinf to install chan_capi 
  but i get this error
  
  [EMAIL PROTECTED]:/usr/src/chan_capi-0.3.5# 
  make
  gcc -pipe -Wall 
  -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include 
  -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 
  -DNEVER_EVER_EARLY_B3_CONNECTS -DCAPI_ES -DCAPI_GAIN -DDEFLECT_ON_CIRCUITBUSY 
  -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations 
  -DCRYPTO -c -o chan_capi.o 
chan_capi.c
  chan_capi.c:36:20: capi20.h: No 
  such file or directory
  In file included from 
  chan_capi.c:39
  
  Anyone cha help 
  me?
  
  Thanks
  
  Giordano
  
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RE: [Asterisk-Users] G.729 patent in France

2005-09-30 Thread Kevin Walsh

Amaury BOSSE [EMAIL PROTECTED] wrote:
 I am building an Asterisk PBX with voicemail and music on hold functions.
 An ISDN BRI line will also be available and G.729 IP-phones will be used.
 
 Are there patents rights applicable to France?
 
The European Parliament recently voted 648 to 14 to reject the Computer
Implemented Inventions Directive.  The directive was supported by large
monopolists such as Microsoft and would have thrown us into the same
software patent minefield as the USA.

The defeat of the bill means that individual EU member countries will
continue to make their own decisions on what is patentable, rather than
being hamstrung by the proposed EU-wide bill.  Software-only patents are
not valid in England and probably not in France, although that's for you
to check.  I understand that a limited number of software patents are
valid in Italy.

Software is protected by copyright, and that's enough.  Ideas are free.

http://www.nosoftwarepatents.com/

-- 
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Re: [Asterisk-Users] Asterisk Echo problems, Urgent, please help,

There is no tx/rxgain on a sip call (other then on the sip phones).   
 Also, no echocancel on sip--sip calls (unless you turn on when
bridged)... but I believe he has stated he is already doing this.

On 9/29/05, Matt [EMAIL PROTECTED] wrote:
 hi:

 We are using 1.0.9 * with sangoma 104 quad card, hooked to 4 E1s. We have no
 echo problems at all.

 The voice qualities sound and clear, try adjust tx/rxgain a bit. and make
 sure your zapata.conf's echocancel param is enabled.

 Best Regards

 Matt
 
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 - Original Message -
 From: Tom Hayden [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Thursday, September 29, 2005 6:02 AM
 Subject: Re: [Asterisk-Users] Asterisk Echo problems, Urgent, please help,


  What kind of POTS trunks/cards are you using?
 
  --
  Tom
 
  On 9/29/05, Ian Bonham [EMAIL PROTECTED] wrote:
   Hi all,
  
   I hope someone can help, as I have an urgent problem.
  
   I've got a production Asterisk server thats been deployed, but we are
 seeing
   a strange voice echo problem. There is about a 250ms echo for the users
 in
   the office, and they are hearing their own voice back at them.
  
   I'm running the CVS Head code, on RH9.0. This is on a P4 box with 2gb of
   memory. The client SIP phones are Polycom Soundpoint IP600's, WiFi ZyXel
   2000w handsets, and X-Lite (free) PC clients. All see the same problem.
   There is a bridge into the POTS (BT's SystemX) using a Voicetronix
   OpenSwitch12 card and the vpbhp driver.
  
   The echo occurs on both SIP-POTS calls, and SIP-SIP calls. I've tried
 a
   number of volume adjustments to correct the echo but it is always the
 same.
  
   If anyone has any ideas I'd really appriciate some help, as this is a
 major
   urgency,
  
   Many many thanks,
  
   Ian Bonham
  
   _
   FREE pop-up blocking with the new MSN Toolbar - get it now!
   http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/
  
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Re: [Asterisk-Users] Any way to not overwrite sound files on compile?

I don't do make samples.

When I do a
./configure
make
make install

I end up with the version of Asterisk I wanted installed, my sound
files get over written, and my config files stay in place =\  
very odd and slightly frustraighting!

On 9/30/05, gincantalupo [EMAIL PROTECTED] wrote:
 Hi,
 just do NOT type make samples: this commands writes original sounds in
 your dirs even if your old custom messages are present.
 (I'm talking about 1.0.7 * version, maybe in newer * versions this
 command is included inside some install script).


 Giorgio.


 Matt wrote:

 Every time I recompile Asterisk (or upgrade to a new CVS-HEAD,
 whatever) asterisk overwrites custom files I have made.  Granted,
 these files are named the same as the asterisk default files
 (vm-login.gsm, etc) because we had a person here record them to
 customize them a bit more for our application.
 
 Short of keeping them somewhere and copying them back every time
 (which isn't all that often) I do a re-compile.  Is there some flag or
 something to tell Asterisk not to install sound files, or at the very
 least not to overwrite ones already existing?
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Re: [Asterisk-Users] Register times out on internet outage

Yeah I've never had any major issue needing a restart.   Additionally,
if you require uptime on your LD and it's due to net outages at your
site, you may want to invest in BGP or at least a fail-over box of
some sort.

On 9/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 On 9/30/2005, Chris Mason (Lists) [EMAIL PROTECTED] wrote:

  I am using AstLinux with
  Asterisk CVS-HEAD-01/10/05-02:11:15-AstLinux built by [EMAIL PROTECTED] on 
  a i686
  running Linux
 
  On this box I am registered to two different providers for long distance
  and international. If there is an internet outage of more than a few
  minutes, I'm not sure how long it takes to make this happen, the
  registration times out and when internet comes back the pbx never
  re-registers and consequently Asterisk has to be reset. As there is
  no-one there that can access the PBX it means the PBX has to be power
  cycled, not an optimal solution.
  Has anyone seen this problem, is it a CVS-HEAD problem?

 Chris - just do a reload.  For my SIP stuff it comes back with a warning
 and I don't think it says anything for IAX.

 It times out after the 10th try (I think) and you should get notices in
 message file.

 Reload has always done it for me.

 Brett
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[Asterisk-Users] Will a VIA Epia ME6000 with a 600MHz Eden fanless CPU be suffiecient for 8 extension system?


Hello

I am using a VIA Epia ME6000 with a 600MHz Eden Fanless CPU.  Is this likely 
to be enough power for a 8 extension system with 6 external pstn lines?


How important is cpu?  Is there some measure, eg xMHz CPU per extension or 
something benchmark?


I have installed 512MB memory - again any benchmark for asterisk memory 
usage?


Angus


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Re: [Asterisk-Users] chan_cap-cm-0.6 deflect support

2005-09-30 Thread Louis-David Mitterrand

On Fri, Sep 30, 2005 at 01:11:19PM +0200, Armin Schindler wrote:
  Also, is there a way to detect that a SIP phone has an active forward
  number and capi-deflect any incoming calls to that number?
 
 If you can retrieve this information from extensions.conf, then you can use 
 my example above.
 
 Anyway, I noticed that the original implementation of deflect specified in
 capi.conf does not work in all cases.
 I plan to remove that and to allow capicommand(deflect|...) only.
 It's not necessary to do that in capi.conf and using different MSNs is 
 difficult too.
 My idea is provide information about 'this is a call-waiting call, no 
 b-channel' to extensions.conf via a variable. And the user then can decide
 what to do with that call using all features of the dialplan.
 I plan to do this for version 0.6.1.

Yes, that would be perfect! Looking forward to that implementation.

Thanks,

-- 
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1) Write down the problem
2) Think real hard
3) Write down the answer
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Re: [Asterisk-Users] Will a VIA Epia ME6000 with a 600MHz Eden fanless CPU be suffiecient for 8 extension system?

2005-09-30 Thread Dustin Wildes


Angus Comber wrote:


Hello

I am using a VIA Epia ME6000 with a 600MHz Eden Fanless CPU.  Is this 
likely to be enough power for a 8 extension system with 6 external 
pstn lines?


How important is cpu?  Is there some measure, eg xMHz CPU per 
extension or something benchmark?


I have installed 512MB memory - again any benchmark for asterisk 
memory usage?


Angus




Hello Angus!
We are using the MII6000 at several locations.  Some with 4port FXO, 
others with T1.  Users range from 3-15.
They have been running fine, one location with only 4 users is running 
with 128meg ram because our 1gig chip was bad - and even they haven't 
had any trouble.

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[Asterisk-Users] analog phone/door buzzer going through a Sipura SPA2000 ATA dials really slowly


Hello

We have setup a doorbell which has an inbuilt analog phone which is 
connected to our Asterisk via a SPA2000 ATA.  The problem we are getting is 
that when a caller presses the buzzer it is taking two or more minutes to 
finally call the reception phone.


In the SPA2000 I have set dtmfmode to be inband.

I notice that with the asterisk you dial a number and then it waits for a 
timeout before dialing number.  I think you use a # to say - just dial now. 
Well we can't program a # into the door system, but could program in another 
character.  Is it possible to use another character?


Any ideas would be much appreciated.

Angus




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Re: [Asterisk-Users] G.729 patent in France

2005-09-30 Thread Steve Underwood


Amaury BOSSE wrote:


Hi all,

I am building an Asterisk PBX with voicemail and music on hold functions.

An ISDN BRI line will also be available and G.729 IP-phones will be used.

 


Are there patents rights applicable to France?

 

Which licence could I use and how many ones are required (only one per 
phone or also for voicemail and MOH)?


A large percentage of the patents applicable to G.729 are held by France 
Telecom. Now guess whether they bothered to get those patents in France. :-)


There are some software patents in the US for algorithms to speed up the 
computation of G.729 on a processor. I doubt those could have got 
through the European patent systems. The basic signal processing patents 
certainly have.


Regards,
Steve

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Re: [Asterisk-Users] VideoConference with UMTS

2005-09-30 Thread Matt Riddell

Sergio Serrano wrote:
 Hi Srs.,
  
 Do you know if it's possible make a videocall from asterisk to UMTS
 mobile phone?. Both technologies use H.263 like videocodec.

Not yet, working on it.

-- 
Cheers,

Matt Riddell
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RE: [Asterisk-Users] Voice Prompts, what do you think? Good voice.

2005-09-30 Thread Faris Raouf

Gregory,

My advice is to go for it. Allison is nice but there are times when her
accent doesn't pass the International test ( e.g. everyone I've ever spoken
to in the UK roll about on the floor laughing when they first hear her in
the Voicemail prompt, telling you to leave a message ).

Others will probably disagree with me (in fact there was a discussion on
this very recently), but if you do go for it, I would personally like to see
the recordings in .wav format (8k, mono, PCM, 16 bit) - Wavelab allows this
to be done very easily. I save all my final prompts in this format because
they provide great sound quality compared to GSM, and also allow for high
quality sonic idents (something I'll be posting about soon. Watch this
space).

If people prefer them in different formats, they can then use the wavs as
the basis and re-encode them (e.g. gsm). But like I said, that's just my
opinion. I'm not saying this is what everybody wants, or what you should
definitely do.

Faris.

-Original Message-
From: Gregory Wiktor - ADCom Corp. 

 There is a good chance I will do it, but want some feedback. 


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Re: [Asterisk-Users] analog phone/door buzzer going through a Sipura SPA2000 ATA dials really slowly

2005-09-30 Thread Matt Riddell

Angus Comber wrote:
 Hello
 
 We have setup a doorbell which has an inbuilt analog phone which is
 connected to our Asterisk via a SPA2000 ATA.  The problem we are getting
 is that when a caller presses the buzzer it is taking two or more
 minutes to finally call the reception phone.

Asterisk will not cause it to wait two or more minutes.  3 seconds yes, 2
minutes, no...

Unless you have some funky gotoifs or loops or waits etc..

-- 
Cheers,

Matt Riddell
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RE: [Asterisk-Users] OOH323C

2005-09-30 Thread Dan Austin

 Dan - as a thought - I am messing with a H323 'capable' IP Phone and I
 am (maybe foolishly) trying to use ooh323 with no gateway, gatekeeper,
 or anything else and I am not getting it to work too well.  It seems
 'sometimes' it does work.
I'm using it to connect to Cisco Call Manager.  I set the connection
up as a friend.  99.9% of the calls will be inbound, but if I need
to test an odd feature or two, I can.  I am not using a gatekeer, or
in a traditional sense a gateway.

I'd recommend testing with the latest available release of ooh323c.
I am not sure how often they push updates into the asterisk-addons,
but I do know that the source from the obj-sys website includes 
chan_h323.  If the latest code doesn't help, then I would send an 
email to their developer list, as I said the developers are clearly
working hard to make it a viable channel, and their support is
outstanding.

 Is there any way - as far as you (or anyone else) knows that this will
 work with any flavor of H323 on Asterisk?  I could just be messing up
 the configs.
I don't have any H.323 endpoints, but suspect it should work for you.

Dan
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Re: [Asterisk-Users] Why does the s extension not work in my extensions.conf file

2005-09-30 Thread Matt Riddell

Angus Comber wrote:
 Hello
 
 In my extensions.conf file:
 
 [frompstnisdn]
 exten = s,1,Dial(SIP/200SIP/202,20)
 exten = s,2,Voicemail(su200)
 exten = s,3,Hangup

If you really want to use s, you will need to add an extension:

exten = 787367,1,Goto(s,1)

-- 
Cheers,

Matt Riddell
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Re: [Asterisk-Users] Best echo canceller?

2005-09-30 Thread Tzafrir Cohen

On Thu, Sep 29, 2005 at 05:23:07PM -0400, Andrew Kohlsmith wrote:
 On Thursday 29 September 2005 17:04, Claudio Canseco wrote:
   In your experience what is the best choice for echo canceller ?
  Which one should work better: STEVE, STEVE2, MARK, MARK2, MARK3, KB1 ?
 
 KB1 is a refactored MARK2 which seems to work VERY, very well.
 
 The others are different attempts at different algorithms.  KB1's the new 
 default, from MARK2.  Try the others, see if they work better for you.

Is there a difference in cpu consumption? (which may translate to
latency if you have enough channels, I guess)

-- 
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http://tzafrir.org.il |   | a Mutt's  
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R: [Asterisk-Users] chan_capi-0.3.5

2005-09-30 Thread Giordano Grandis









Thanks Jorg,

its worked, but
what modules i need to use it with asterisk? 



I insert load =
chan_capi.so in /etc/asterisk/modules.conf and chan_capi.so=yes under [globals]
section.



When asterisk start, I get
this error:



  == Parsing
'/etc/asterisk/modules.conf': Found

 [chan_capi.so] =
(Common ISDN API for Asterisk)

  == Parsing
'/etc/asterisk/capi.conf': Found

Sep 30 16:00:06
WARNING[8294]: loader.c:345 ast_load_resource: chan_capi.so: load_module
failed, returning -1

Sep 30 16:00:06
WARNING[8294]: chan_capi.c:2812 unload_module: Unable to unregister from CAPI!

  == Unregistered channel
type 'CAPI'

Sep 30 16:00:06
WARNING[8294]: loader.c:391 load_modules: Loading module chan_capi.so failed!



Thanks again!





Giordano Grandis

g.grand[EMAIL PROTECTED]



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Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Per conto di Jörg Wolf
Inviato: venerdì 30 settembre 2005
14.15
A: Asterisk
 Users Mailing List - Non-Commercial Discussion
Oggetto: RE: [Asterisk-Users]
chan_capi-0.3.5





Giordano,



you simply don't have capi installed...



On debian sarge you can install the
following packages:

 - capiutils

 - libcapi20-dev



Hope it helps

cheers

Jörg











From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Giordano Grandis
Sent: Friday, September 30, 2005
1:37 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
chan_capi-0.3.5

Hi all,

im tryinf to install chan_capi but i get this
error



[EMAIL PROTECTED]:/usr/src/chan_capi-0.3.5# make

gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations
-g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586
-DNEVER_EVER_EARLY_B3_CONNECTS -DCAPI_ES -DCAPI_GAIN -DDEFLECT_ON_CIRCUITBUSY
-DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations
-DCRYPTO -c -o chan_capi.o chan_capi.c

chan_capi.c:36:20: capi20.h: No such file or
directory

In file included from chan_capi.c:39



Anyone cha help me?



Thanks



Giordano










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[Asterisk-Users] Re:Any way to not overwrite sound files on compile?

2005-09-30 Thread Gustavo A. Gonzalez

When compile only type 'make'  and copy manually your module/s from asterisk
apps directory into your asterisk modules directory.

regards.
G.


Matt wrote:

Every time I recompile Asterisk (or upgrade to a new CVS-HEAD,
whatever) asterisk overwrites custom files I have made.  Granted,
these files are named the same as the asterisk default files
(vm-login.gsm, etc) because we had a person here record them to
customize them a bit more for our application.

Short of keeping them somewhere and copying them back every time
(which isn't all that often) I do a re-compile.  Is there some flag or
something to tell Asterisk not to install sound files, or at the very
least not to overwrite ones already existing?

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Re: [Asterisk-Users] analog phone/door buzzer going through a SipuraSPA2000 ATA dials really slowly


The unit dials 300 and in my extensions.conf I have:

exten = 300,1,Dial(SIP/200SIP/201,30)
exten = 300,2,Hangup

So perhaps it is some setting in the Sipura ATA?





- Original Message - 
From: Matt Riddell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Friday, September 30, 2005 1:51 PM
Subject: Re: [Asterisk-Users] analog phone/door buzzer going through a 
SipuraSPA2000 ATA dials really slowly




Angus Comber wrote:

Hello

We have setup a doorbell which has an inbuilt analog phone which is
connected to our Asterisk via a SPA2000 ATA.  The problem we are getting
is that when a caller presses the buzzer it is taking two or more
minutes to finally call the reception phone.


Asterisk will not cause it to wait two or more minutes.  3 seconds yes, 2
minutes, no...

Unless you have some funky gotoifs or loops or waits etc..

--
Cheers,

Matt Riddell
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[Asterisk-Users] TE410P not working

2005-09-30 Thread Simone Cittadini

I'm trying to install a TE410P this is what happens with compiled zaptel 
1.0.9, 1.2-beta and 1.0.9 from http://updates.xorcom.com/iso/


this is my zaptel.conf (checked with the provider the values):

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone=it
defaultzone=it

then I modprobe wct4xxp debug=1 t1e1override=15 and the kernel says :

Sep 30 16:12:40 localhost kernel: Zapata Telephony Interface Registered 
on major 196
Sep 30 16:12:40 localhost kernel: Found TE4XXP at base address df5ffc00, 
remapped to f8aa2c00
Sep 30 16:12:40 localhost kernel: TE4XXP version c01a0164, burst ON, 
slip debug: OFF

Sep 30 16:12:40 localhost kernel: FALC version: 0005, Board ID: 00
Sep 30 16:12:40 localhost kernel: Reg 0: 0x17f2f400
Sep 30 16:12:40 localhost kernel: Reg 1: 0x17f2f000
Sep 30 16:12:40 localhost kernel: Reg 2: 0x
Sep 30 16:12:40 localhost kernel: Reg 3: 0x
Sep 30 16:12:40 localhost kernel: Reg 4: 0x0001
Sep 30 16:12:40 localhost kernel: Reg 5: 0x
Sep 30 16:12:40 localhost kernel: Reg 6: 0xc01a0164
Sep 30 16:12:40 localhost kernel: Reg 7: 0x1000
Sep 30 16:12:40 localhost kernel: Reg 8: 0x
Sep 30 16:12:40 localhost kernel: Reg 9: 0x00ff
Sep 30 16:12:40 localhost kernel: Reg 10: 0x
Sep 30 16:12:40 localhost kernel: TE4XXP: Launching card: 0
Sep 30 16:12:40 localhost kernel: TE4XXP: Setting up global serial 
parameters
Sep 30 16:12:40 localhost kernel: Successfully initialized serial bus 
for unit 0
Sep 30 16:12:40 localhost kernel: Successfully initialized serial bus 
for unit 1
Sep 30 16:12:40 localhost kernel: Successfully initialized serial bus 
for unit 2
Sep 30 16:12:40 localhost kernel: Successfully initialized serial bus 
for unit 3
Sep 30 16:12:40 localhost kernel: Found a Wildcard: Wildcard TE410P (2nd 
Gen)



so I do  /sbin/ztcfg -vvv, which tells me :

Zaptel Configuration
==

SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: Individual Clear channel (Default) (Slaves: 03)
Channel 04: Individual Clear channel (Default) (Slaves: 04)
Channel 05: Individual Clear channel (Default) (Slaves: 05)
Channel 06: Individual Clear channel (Default) (Slaves: 06)
Channel 07: Individual Clear channel (Default) (Slaves: 07)
Channel 08: Individual Clear channel (Default) (Slaves: 08)
Channel 09: Individual Clear channel (Default) (Slaves: 09)
Channel 10: Individual Clear channel (Default) (Slaves: 10)
Channel 11: Individual Clear channel (Default) (Slaves: 11)
Channel 12: Individual Clear channel (Default) (Slaves: 12)
Channel 13: Individual Clear channel (Default) (Slaves: 13)
Channel 14: Individual Clear channel (Default) (Slaves: 14)
Channel 15: Individual Clear channel (Default) (Slaves: 15)
Channel 16: D-channel (Default) (Slaves: 16)
Channel 17: Individual Clear channel (Default) (Slaves: 17)
Channel 18: Individual Clear channel (Default) (Slaves: 18)
Channel 19: Individual Clear channel (Default) (Slaves: 19)
Channel 20: Individual Clear channel (Default) (Slaves: 20)
Channel 21: Individual Clear channel (Default) (Slaves: 21)
Channel 22: Individual Clear channel (Default) (Slaves: 22)
Channel 23: Individual Clear channel (Default) (Slaves: 23)
Channel 24: Individual Clear channel (Default) (Slaves: 24)
Channel 25: Individual Clear channel (Default) (Slaves: 25)
Channel 26: Individual Clear channel (Default) (Slaves: 26)
Channel 27: Individual Clear channel (Default) (Slaves: 27)
Channel 28: Individual Clear channel (Default) (Slaves: 28)
Channel 29: Individual Clear channel (Default) (Slaves: 29)
Channel 30: Individual Clear channel (Default) (Slaves: 30)
Channel 31: Individual Clear channel (Default) (Slaves: 31)

31 channels configured.

while the kernel logs :

Sep 30 16:21:07 localhost kernel: About to enter spanconfig!
Sep 30 16:21:07 localhost kernel: TE4XXP: Configuring span 1
Sep 30 16:21:07 localhost kernel: Done with spanconfig!
Sep 30 16:21:07 localhost kernel: TE4XXP: Configured channel 1 
(TE4/0/1/1) sigtype 128

Sep 30 16:21:07 localhost kernel: Unassigning channel 0/1!
Sep 30 16:21:07 localhost kernel: TE4XXP: Configured channel 2 
(TE4/0/1/2) sigtype 128

Sep 30 16:21:07 localhost kernel: Unassigning channel 0/2!
Sep 30 16:21:07 localhost kernel: TE4XXP: Configured channel 3 
(TE4/0/1/3) sigtype 128

Sep 30 16:21:07 localhost kernel: Unassigning channel 0/3!
Sep 30 16:21:07 localhost kernel: TE4XXP: Configured channel 4 
(TE4/0/1/4) sigtype 128

Sep 30 16:21:07 localhost kernel: Unassigning channel 0/4!
Sep 30 16:21:07 localhost kernel: TE4XXP: Configured channel 5 
(TE4/0/1/5) sigtype 128

Sep 30 16:21:07 localhost kernel: Unassigning channel 0/5!
Sep 30 16:21:07 localhost kernel: TE4XXP: Configured channel 6 
(TE4/0/1/6) sigtype 128

Sep 30 16:21:07 localhost kernel: Unassigning channel 0/6!
Sep 30 16:21:07 localhost kernel: TE4XXP: Configured channel

Re: [Asterisk-Users] Best echo canceller?

On Friday 30 September 2005 08:57, Tzafrir Cohen wrote:
 Is there a difference in cpu consumption? (which may translate to
 latency if you have enough channels, I guess)

No.  it's just refactored and fixes a few variable inits and stuff IIRC.  The 
patches on the bugtracker explained it quite well.

-A.
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[Asterisk-Users] Not Authenticate

Why Asterisk show this message?
What I can do?

Sep 30 15:45:18 NOTICE[3608]: chan_sip.c:9096
handle_response_invite: Failed to authenticate on
INVITE to '100
sip:[EMAIL PROTECTED];tag=as413bd6a8'
-- SIP/sip.uni.it-df15 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)

Thanks!!



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[Asterisk-Users] Not authenticate

Why Asterisk show this message?
What I can do?

Sep 30 15:45:18 NOTICE[3608]: chan_sip.c:9096
handle_response_invite: Failed to authenticate on
INVITE to '100
sip:[EMAIL PROTECTED];tag=as413bd6a8'
-- SIP/sip.uni.it-df15 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)

Thanks






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[Asterisk-Users] Echo Cancellation not working in Zapata.conf

2005-09-30 Thread Alberto Risco









I have echocancel=yes in zapata.conf but when I do a zap show
channel 1, I notice echo cancellation is turned off.



I followed the article that talks about the order in which
the statements need to be in zapata.conf to get echo canceling to work:





http://lists.digium.com/pipermail/asterisk-users/2005-June/110615.html



But it is still not working. Does anyone know how to
get echo cancellation to work?



We have Asterisk 1.0.7 and Zaptel 1.0.9 with 2 PRIs using a
TE410P board.



Here is the output from the CLI:



zap show channel 1

Channel: 1*CLI

File Descriptor: 25

Span: 1

Extension:

Dialing: no

Context: aheeva

Caller ID string:

Destroy: 0

InAlarm: 0

Signalling Type: PRI Signalling

Owner: None

Real: NoneLI

Callwait: None

Threeway: None

Confno: -1

Propagated Conference: -1

Real in conference: 0

DSP: no

Relax DTMF: yes

Dialing/CallwaitCAS: 0/0

Default law: ulaw

Fax Handled: no

Pulse phone: no 

Echo Cancellation: 128 taps, currently OFF

PRI Flags:

PRI Logical Span: Implicit

Actual Hookstate: Onhook







Here is my Zapata.conf:



; Zapata telephony interface

;

; Configuration file



[channels]

;

usecallerid=yes

hidecallerid=no

callwaiting=no

restrictcid=no

usecallingpres=yes

callwaitingcallerid=yes

threewaycalling=no

transfer=no

cancallforward=yes

callreturn=yes

callerid=asreceived

;

relaxdtmf=yes

;

rxgain=0.0

txgain=0.0 

immediate=no

;

; Configure jitter buffers in zapata (each one is 20ms,
default is 4)

;

jitterbuffers=4

;



context=aheeva

switchtype=national

signalling=pri_cpe

pridialplan=unknown ; needed to pass proper #
digits to PRI

echocancel=yes

echotraining=yes

echocancelwhenbridged=yes

group = 1

channel = 1-23

;channel = 25-48 









Thanks,



Alberto





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Re: Mathematicians wanted (was RE: [Asterisk-Users] Best echo canceller?)

2005-09-30 Thread Steve Underwood

Kris Boutilier wrote:

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew
Kohlsmith
Sent: Thursday, September 29, 2005 2:23 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Best echo canceller?

On Thursday 29 September 2005 17:04, Claudio Canseco wrote:

In your experience what is the best choice for echo canceller ?
Which one should work better: STEVE, STEVE2, MARK, MARK2, MARK3, KB1 ?

KB1 is a refactored MARK2 which seems to work VERY, very well.

{clip}

The refactoring applied to MARK2 to create KB1 was basically intended to make 
the code generally consistent with the Texas Instruments whitepaper referenced 
in the comments at the top of the file. In that document they completely 
outline the operating theory of one particular echo cancellation algorithm and 
completely document an implementation of it in a general purpose TI processor. 
The implementation is also benchmarked and deviations/performance issues 
explored. The MARK2/KB1 implementation is not a 100% complete version of the 
reference code - there are some autotuning elements and perhaps additional 
optimizations suggested by TI that can certainly still be implemented.

I would strongly encourage anyone with a good understanding of mathematics to 
take a look at the whitepaper, compare the KB1 the source and see what can be 
improved on. The echo canceller code itself isn't complicated but the math is 
somewhat of a dark art... Just make sure to add copious plain English 
explanations to any changes you submit so the rest of us can keep up.

Would you be interested in making this echo canceller reasonably G.168 
compliant?

Regards,
Steve

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RE: [Asterisk-Users] Echo Cancellation not working in Zapata.conf

2005-09-30 Thread Alberto Risco









Nevermind. It
turns out that if you are not on an active call on the channel, the zap
show channel x shows OFF by default. After placing a call and
checking the channel, it showed Echo Cancellation: 128 taps, currently
on as it should. So our setup is correct after all.





Alberto











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alberto Risco
Sent: Friday, September 30, 2005
9:58 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Echo
Cancellation not working in Zapata.conf





I have echocancel=yes in zapata.conf but when I do a zap
show channel 1, I notice echo cancellation is turned off.



I followed the article that talks about the order in which the
statements need to be in zapata.conf to get echo canceling to work:





http://lists.digium.com/pipermail/asterisk-users/2005-June/110615.html



But it is still not working. Does anyone know how to
get echo cancellation to work?



We have Asterisk 1.0.7 and Zaptel 1.0.9 with 2 PRIs using a
TE410P board.



Here is the output from the CLI:



zap show channel 1

Channel: 1*CLI

File Descriptor: 25

Span: 1

Extension:

Dialing: no

Context: aheeva

Caller ID string:

Destroy: 0

InAlarm: 0

Signalling Type: PRI Signalling

Owner: None

Real: NoneLI

Callwait: None

Threeway: None

Confno: -1

Propagated Conference: -1

Real in conference: 0

DSP: no

Relax DTMF: yes

Dialing/CallwaitCAS: 0/0

Default law: ulaw

Fax Handled: no

Pulse phone: no 

Echo Cancellation: 128 taps, currently OFF

PRI Flags:

PRI Logical Span: Implicit

Actual Hookstate: Onhook







Here is my Zapata.conf:



; Zapata telephony interface

;

; Configuration file



[channels]

;

usecallerid=yes

hidecallerid=no

callwaiting=no

restrictcid=no

usecallingpres=yes

callwaitingcallerid=yes

threewaycalling=no

transfer=no

cancallforward=yes

callreturn=yes

callerid=asreceived

;

relaxdtmf=yes

;

rxgain=0.0

txgain=0.0 

immediate=no

;

; Configure jitter buffers in zapata (each one is 20ms,
default is 4)

;

jitterbuffers=4

;



context=aheeva

switchtype=national

signalling=pri_cpe

pridialplan=unknown ; needed to pass proper #
digits to PRI

echocancel=yes

echotraining=yes

echocancelwhenbridged=yes

group = 1

channel = 1-23

;channel = 25-48 









Thanks,



Alberto





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RE: [Asterisk-Users] OT: SIPSAK usage

2005-09-30 Thread Colin Anderson

I'm using sipsak to send messages to Snoms in my subnet. At work, works
fine:

sipsak -M -O desktop -B foo -s sip:[EMAIL PROTECTED] -H 192.168.1.46

displays foo on the Snom display

On my home LAN (AAH 1.5, Snom 190 3.60s, switched 100, no VLAN, no routing)
the same command (modified for my LAN) always yields:

(type: 3, code: 3): from 192.168.171.8 

at the console of the sending machine. Same if I use FQDN.  Type 3 Code 3
means ICMP port unreachable 

Doing a PCAP from the phone indicates that the Snom gets the message, but
nothing shows up in the SIP log. Doing tcpdump on the originating machine
yields something like Reply from 192.168.171.8  192.168.171.10 UDP port
5060 unreachable

Same phones, same firmware rev, same version of sipsak, no IPTABLES on the
originating machine, DNS lookups work, call behavior is normal, other SIP
behavior like MWI works fine. I got nuthin here, anyone got a tip?
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Re: [Asterisk-Users] Echo Cancellation not working in Zapata.conf

On Friday 30 September 2005 09:57, Alberto Risco wrote:
 I have echocancel=yes in zapata.conf but when I do a zap show channel 1,
 I notice echo cancellation is turned off.

If the channel is not in use, echo cancellation will be off.

Your show zap channel output shows it's on-hook, so the DSP and echo canceller 
will be off.  

Are you actually having issues or are you falling into the trap of 
overanalyzing everything?

-A.
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[Asterisk-Users] 911 Q

2005-09-30 Thread Joel Newkirk

OK, got a question on 911.

Looking into setting up a couple asterisk servers at a country club,
with VOIP phones in each of 100 short-term residential rental units.
Approx 100 extensions, approx 24 outside lines.

Since everything is geographically at one location, reaching 911
correctly shouldn't present a problem.  However, the club wishes to
ensure that 911 authorities are able to identify the precise rental unit
placing the call.

How can we achieve this, short of 'reciting' the unit number aloud at
the beginning of the placed call?

Thanks for any advice/tips.

j


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[Asterisk-Users] chan_zap.so ?

2005-09-30 Thread cyril SIMON

Hi,

I've a little problem with my asterisk server.

I have managed an asterisk server for a few months
one.

Today, I wanted to restart it and when I did it, my
asterisk server didn't want to start again.

I looked at the various messages in /var/log
and the same thing appears.

 [chan_zap.so] = (Zapata Telephony)
  == Parsing '/etc/asterisk/zapata.conf': Found
Sep 30 16:51:53 WARNING[2343]: chan_zap.c:816 zt_open:
Unable to specify channel 1: No such device or address
Sep 30 16:51:53 ERROR[2343]: chan_zap.c:6398 mkintf:
Unable to open channel 1: No such device or address
here = 0, tmp-channel = 1, channel = 1
Sep 30 16:51:53 ERROR[2343]: chan_zap.c:9497
setup_zap: Unable to register channel '1'
Sep 30 16:51:53 WARNING[2343]: loader.c:388
__load_resource: chan_zap.so: load_module failed,
returning -1
  == Unregistered channel type 'Zap'
Sep 30 16:51:53 WARNING[2343]: loader.c:509
load_modules: Loading module chan_zap.so failed!


Sep 30 16:16:22 ERROR[3091]: Unable to open channel
1: No such device or address
here = 0, tmp-channel = 1, channel = 1
Sep 30 16:16:22 ERROR[3091]: Unable to register
channel '1'


I use a card Wildcard X100P, Wildcard X101P: wcfxo.
I use a configuration from database.

I really do not know that can be the problem.

If somebody on the list has an idea It would be
with pleasure

Thanks,

Cyril







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[Asterisk-Users] No ringback tone generated by Asterisk with OH323 connections

2005-09-30 Thread Juan Jose Comellas

I am using Asterisk (Debian unstable packages) with an OH323 connection to my 
provider. Everything is working except for the generation of ringback tones 
when I receive inbound calls from the PSTN. My provider tells me that we're 
sending call progress indications and that because of this they're expecting 
us to generate the ringback tone. Does anybody know how to configure this in 
Asterisk? The relevant settings in oh323.conf are:

[general]
listenAddress=0.0.0.0
listenPort=1720
tcpStart=20001
tcpEnd=3
udpStart=20001
udpEnd=3
fastStart=yes
h245Tunnelling=yes
h245inSetup=yes
inBandDTMF=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
simultaneousMax=10
bandwidthLimit=2000
gatekeeper=DISABLE
gatekeeperTTL=600
userInputMode=RFC2833

The package versions I'm using are:

asterisk1.0.9.dfsg-5
asterisk-oh323  0.6.6pre3-4
libopenh323-1.15.3c21.15.3-4

-- 
Juan Jose Comellas
([EMAIL PROTECTED])

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Re: [Asterisk-Users] zttest - 100% ?

2005-09-30 Thread Kevin Bockman


[EMAIL PROTECTED] wrote:

just as an (bad) example:
we are using an x206 and couldn't get the zttest above 99.975 
equal what we were doing
single irq, w/o acpi, w/o apic, different kernels, w/o 
hyperthreading, different slots, a.s.o.
for an Digium wildcard TE110P 

so if someone got such a board to zttest 100% maybe could give some 
information if where's something

special to run asterisk on such boards...
otherwise I think there are production differences on the ibm-mainboards 
or the used chipsets


we'll change hardware next...
You don't have to have 100% on zttest.  You probably won't get it.  I 
get the same results on one of my servers and  it runs perfectly.


Kevin
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[Asterisk-Users] No Incoming Calls on Asterisk

2005-09-30 Thread Zeeshan









Hi,



My VoIP service provider has provided me
with a Sipura adapter and it works perefect. But I want to receive calls on my Asterisk
server. Ive tried everything but no success. I can dial successfully
from Asterisk but it doesnt receive calls. Dialing phone hears a busy
tone and cell phone says Call Failed. What am I missing in settings,
or what should I need to ask from service provider to add in my Asterisk.



My SIP is:

[general]

port = 5060

bindaddr =
0.0.0.0

disallow=all

allow=ulaw

allow=alaw

context = sip-external

callerid =
Unknown

useragent =
Sipura/SPA2000-2.0.10(e)



[siptrunk]

username=username

type=peer

secret=password

insecure=very

host=x.x.x.x

disallow=all

canreinvite=no

canredirect=no

allow=alaw

allow=ulaw



[incoming]

username=username

type=user

secret=password

qualify=no

fromuser=username

context=from-pstn

canreinvite=no






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Re: [Asterisk-Users] chan_zap.so ?

On Friday 30 September 2005 10:26, cyril SIMON wrote:
 I've a little problem with my asterisk server.

Yes, it is a little problem.

 Today, I wanted to restart it and when I did it, my
 asterisk server didn't want to start again.

It's telling you the problem pretty damn clearly:

 Sep 30 16:51:53 WARNING[2343]: chan_zap.c:816 zt_open:
 Unable to specify channel 1: No such device or address

It can't find the zaptel hardware.  There are a few causes for this:

1) You took the hardware out
2) You changed the configuration of the hardware
3) You didn't load the drivers for the hardware

 I really do not know that can be the problem.

Please review the potential causes above and report back.  I'm curious as to 
what the actual problem was.  We can only improve our ability to help if you 
provide good feedback.

-A.
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RE: [Asterisk-Users] No ringback tone generated by Asterisk with OH323connections

2005-09-30 Thread Brian C. Fertig

are you giving answer()?

..o---o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juan Jose
Comellas
Sent: Friday, September 30, 2005 10:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] No ringback tone generated by Asterisk with
OH323connections

I am using Asterisk (Debian unstable packages) with an OH323 connection
to my 
provider. Everything is working except for the generation of ringback
tones 
when I receive inbound calls from the PSTN. My provider tells me that
we're 
sending call progress indications and that because of this they're
expecting 
us to generate the ringback tone. Does anybody know how to configure
this in 
Asterisk? The relevant settings in oh323.conf are:

[general]
listenAddress=0.0.0.0
listenPort=1720
tcpStart=20001
tcpEnd=3
udpStart=20001
udpEnd=3
fastStart=yes
h245Tunnelling=yes
h245inSetup=yes
inBandDTMF=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
simultaneousMax=10
bandwidthLimit=2000
gatekeeper=DISABLE
gatekeeperTTL=600
userInputMode=RFC2833

The package versions I'm using are:

asterisk1.0.9.dfsg-5
asterisk-oh323  0.6.6pre3-4
libopenh323-1.15.3c21.15.3-4

-- 
Juan Jose Comellas
([EMAIL PROTECTED])

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Re: [Asterisk-Users] No Incoming Calls on Asterisk

I use UNIVOICE provider, therefore you change
sip.uni.it with your provider.

View files






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extensions.conf
Description: 3949034846-extensions.conf


sip.conf
Description: 3455877249-sip.conf
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RE: [Asterisk-Users] 911 Q

2005-09-30 Thread Alexander Lopez

 With hotel systems When some places a 911 call it is printed on the
printer in the Front Desk, Hwen help arrives they usually go to the
Frount Dsek anyway.  I would set up a System() that would not only
printout he romm number on the Front Desk Printer but also drop a call
file in to trigger a call to the Front Desk with a prerecorded message
of wht extention just called 911. That way the Hotel can send someone to
the room to act as first response and the Frount Desk can direct the 911
team to the correct location.



 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Joel Newkirk
 Sent: Friday, September 30, 2005 10:20 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] 911 Q
 
 OK, got a question on 911.
 
 Looking into setting up a couple asterisk servers at a 
 country club, with VOIP phones in each of 100 short-term 
 residential rental units.
 Approx 100 extensions, approx 24 outside lines.
 
 Since everything is geographically at one location, reaching 
 911 correctly shouldn't present a problem.  However, the club 
 wishes to ensure that 911 authorities are able to identify 
 the precise rental unit placing the call.
 
 How can we achieve this, short of 'reciting' the unit number 
 aloud at the beginning of the placed call?
 
 Thanks for any advice/tips.
 
 j
 
 
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[Asterisk-Users] Asterisk and telephone volume


Hello

I am using a Snom 190 and the quality seems OK.  Trouble is the volume is 
quite low and full volume on the Snom is still too low.  Is there something 
I can do on the asterisk to increase the volume?


Angus 



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[Asterisk-Users] Asterisk::AGI - What license ???

2005-09-30 Thread Jean-Michel Hiver


Hi,

Asterisk::AGI is a fantastic piece of software. Unfortunately it comes 
with NO LICENSE WHATSOEVER. That's very annoying when you want to write 
GPL stuff that depends on it.


I have tried mailing the author some time ago with no response. Does 
anybody know what the software license for this module is?


Cheers,
Jean-Michel.
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Re: [Asterisk-Users] Canada VOIP provider quality

2005-09-30 Thread Jean-Michel Hiver



When you say sufficient capacity what, exactly, do you mean? We 
monitor our B channel utilisation and add PRIs whenever we see peak 
usage above 90% (note -- peak, not average). We aim for less than 0.1% 
blocking factor and have not yet come close.


Pretty cool. What tool do you use to monitor channel usage?

Cheers,
Jean-Michel.

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Re: [Asterisk-Users] G.729 patent in France

2005-09-30 Thread Jean-Michel Hiver


Amaury BOSSE a écrit :


Hi all,

I am building an Asterisk PBX with voicemail and music on hold functions.

An ISDN BRI line will also be available and G.729 IP-phones will be used.

 


Are there patents rights applicable to France?

 

Which licence could I use and how many ones are required (only one per 
phone or also for voicemail and MOH)?


You can buy them from Digium. It costs $10 per channel. Which means that 
if you buy five, asterisk will transcode at maximum 5 channels at a time.

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[Asterisk-Users] strange wave like noise on sip handset


Hello

On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like 
sound - it gets louder then softer and continually repeats.


I don't remember hearing this when using other handsets.  But what is this 
effect?  How can I reduce it?


Angus


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Re: [Asterisk-Users] maximum retries exceeded on call

2005-09-30 Thread Michael Häberle


has somebody an advise.
Do I need to provide more information?

Regards
Michael

Michael Häberle wrote:

Hi,

I phone with phpagi and/or x-pro.
Sometimes I get this warning in the asterisk-console:
maximum retries exceeded on call.
I noticed when this message shows up, asterisk hangs up the call (even 
when i'am in the middle of a call, according to our employess)


When they restart x-pro it seems to work properly again (at least some 
time).


Asterisk and the clients are in the same LAN.

I read the FAQ at voip-info.org but it didn't help.

Here is my sip.conf
--
[general]
context=telin
port=5060
bindaddr=0.0.0.0
srvlookup=yes
toos=lowdelay

allow=g726
allow=ulaw

rtptimeout=60
rtpholdtimeout=300

useragent=EASYCOM
nat=yes
-
after that comes the whole register-thing

here comes a sample user (all are the same)
-
[user]
context=telout
type=friend
secret=XXX
dtmfmode=rfc2833
host=dynamic
allow=all
canreinvite=no
-

in x-pro everything is standard (nothing changend but the 
network-settings and sip-proxy)


Since Iam neither a linux nor a asterisk-crack, I don't really have a 
clue what's going on.


Hope you can help me :)

Regards
Michael




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RE: [Asterisk-Users] Asterisk and telephone volume

2005-09-30 Thread Colin Anderson

Funny, I find it just fine, but I have had a few users complain about it.
One lady said her phone wasn't working at all, so I checked it out, worked
fine, and then she admitted to me that she was 80% deaf in her one ear, and
on her old Vista 390 she had the volume cranked so it was ridiculously loud.
Solution? Train the user to put the phone in the other ear! 

The volume I think is fine, but it's true that Snoms do not have an 11
Spinal-tap syle setting. A setting like this will lead to echo on the call
because the microphone is very sensitive. For users that have problems like
this, a headset is the way to go. This will probably become an issue more
and more in the future as Ipod users slowly destroy their hearing. 

You could try cranking Rx gain in your zapata.conf but again that's going to
lead to echo. 

-Original Message-
From: Angus Comber [mailto:[EMAIL PROTECTED]
Sent: Friday, September 30, 2005 8:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk and telephone volume


Hello

I am using a Snom 190 and the quality seems OK.  Trouble is the volume is 
quite low and full volume on the Snom is still too low.  Is there something 
I can do on the asterisk to increase the volume?

Angus 


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[Asterisk-Users] Question about 3Com(r) 3101 Basic Phone

2005-09-30 Thread Jorge Cisneros



 Hi, i have one question, the 3Com
3101 Basic Phone work with asterisk, if so i any a especial firmware o
another thing. And wath other 3com ip phone product work with asterisk.
I think is a good idea to create a list with the all voip device and
the status with asterisk.


 Thanks.

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Re: [Asterisk-Users] Any way to not overwrite sound files on compile?

2005-09-30 Thread Kevin P. Fleming


Matt wrote:


I end up with the version of Asterisk I wanted installed, my sound
files get over written, and my config files stay in place =\  
very odd and slightly frustraighting!


That is correct. 'make install' installs the standard sound files along 
with the binaries; if we did not do that, then when the code had been 
changed to required new sound files they would not be present...


However, we have been working on a simpler method to handle this, where 
the sound files directory would be version-tagged, and we wouldn't 
overwrite anything unless the new version was needed. This would still 
overwrite your files though, if a new version of the sound files was 
needed with an Asterisk upgrade. The only solution for that problem is 
to version-tag every single sound file, and I don't think it's worth the 
hassle for that.


How about if we add a 'post-install' step in the Makefile, that would 
run a local script/program if specified, which could copy your sound 
files back into place?

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Re: [Asterisk-Users] Re:Any way to not overwrite sound files on compile?

What about during an upgrade though? From one version to another where
you actually want to replace asterisk?  I mean it isn't just a few
files (I don't think?) that get copied when you do make install is it?
 Aren't there a fair amount?

On 9/30/05, Gustavo A. Gonzalez [EMAIL PROTECTED] wrote:
 When compile only type 'make'  and copy manually your module/s from asterisk
 apps directory into your asterisk modules directory.

 regards.
 G.


 Matt wrote:

 Every time I recompile Asterisk (or upgrade to a new CVS-HEAD,
 whatever) asterisk overwrites custom files I have made.  Granted,
 these files are named the same as the asterisk default files
 (vm-login.gsm, etc) because we had a person here record them to
 customize them a bit more for our application.
 
 Short of keeping them somewhere and copying them back every time
 (which isn't all that often) I do a re-compile.  Is there some flag or
 something to tell Asterisk not to install sound files, or at the very
 least not to overwrite ones already existing?

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Re: [Asterisk-Users] Any way to not overwrite sound files on compile?

A post-install would be great (or I myself can write a script)... it
isn't that big of a deal.. I just wanted to see if I was over looking
something.   Tagging the sound directory for a version would also be
good but if there is no way (and I do understand the reasoning)
then I can just write a simple shell script to copy my files back and
keep them safe elsewhere.

On 9/30/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
 Matt wrote:

  I end up with the version of Asterisk I wanted installed, my sound
  files get over written, and my config files stay in place =\
  very odd and slightly frustraighting!

 That is correct. 'make install' installs the standard sound files along
 with the binaries; if we did not do that, then when the code had been
 changed to required new sound files they would not be present...

 However, we have been working on a simpler method to handle this, where
 the sound files directory would be version-tagged, and we wouldn't
 overwrite anything unless the new version was needed. This would still
 overwrite your files though, if a new version of the sound files was
 needed with an Asterisk upgrade. The only solution for that problem is
 to version-tag every single sound file, and I don't think it's worth the
 hassle for that.

 How about if we add a 'post-install' step in the Makefile, that would
 run a local script/program if specified, which could copy your sound
 files back into place?

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[Asterisk-Users] Calls Dropping w/ Cisco 7960 Phones

2005-09-30 Thread Jon Dahl

Hello,

I have scoured google for the last couple of days, implemented some changes but my issue is still occuring.

My company uses a hardware Bridge System for conferencing. Typically,
users will call in from cell phones but three always call from the VoIP
system.

Once or twice a day, one of the VoIP phones will just drop.
Subsequently, we will hear a modem like sound through the bridge system
when this happens.

The only way to make the sound disappear is to reset the Bridge or unplug the POTS line to which the VoIP phone came in on.

We have Asterisk CVS-v1-0-05/30/05-19:08:10 running on a Dual Xeon
3.0Ghz 2GB RAM machine. OS is Redhat 4.0 with updates applied regularly.
Our provider is SIP only, no IAX. Our phones (Cisco 7960 - SIP) are
connected to a Cisco 3560 POE switch. IP Addresses are received through
DHCP and link speed
is auto.

Has anyone else experienced this situation before? It's hard for me to
figure out if Asterisk or the phones are sending the modem sound or
not. The phone
company has checked each POTS line into the Bridge and they are all
fine. The Bridge is new and recently installed. The vendor had checked
everything
out and there are no issues with it. Additionally, the Bridge requires
a security code, so there is no way a fax machine could get in as the
lines(Phone Numbers)
are new.

I'm not seeing anything in the /asterisk/messages file stating there
were any problems. No disconnects are listed, or unable to register
with our provider.

TIA,

Jon Dahl
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RE: [Asterisk-Users] G.729 patent in France

2005-09-30 Thread Kevin Walsh

Steve Underwood [EMAIL PROTECTED] wrote:
 A large percentage of the patents applicable to G.729 are held by France
 Telecom. Now guess whether they bothered to get those patents in France.

British Telecom has a large number of patents in North America.  It can't
use its software-only patents in England, of course.

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[Asterisk-Users] It is possible to have 2 AVM Fritz! USB for multiple BRI access?

2005-09-30 Thread Amaury BOSSE

Hello Asterisk users, 

I would like to use 2 BRI lines on my * box but I haven't any PCI slot.
Is it to possible to use 2 two AVM Fritz! USB.
If not, what other solution can I use?

Thanks

Amaury



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RE: [Asterisk-Users] OT: SIPSAK usage

2005-09-30 Thread Christian Stredicke

snom phones by default do not accept SIP messages from other
destinations that the registrar (in this case they send a error
response) and they dont listen on port 5060 by default. Reason:
SECURITY!!!

If you want to lower your security, you can manually specify the SIP
port to 5060 and manually disable the filering from the proxy/registrar.
But then dont complain if people make a fun out of themselves by making
your phone ring with funny SIPSAK requests!!!

I think the best practice on this is to send the requests to the proxy
which then will forward the packets depending on the proxy's security
policy. Replace proxy with Asterisk!

Christian

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Colin Anderson
 Sent: Friday, September 30, 2005 4:31 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] OT: SIPSAK usage 
 
 I'm using sipsak to send messages to Snoms in my subnet. At 
 work, works
 fine:
 
 sipsak -M -O desktop -B foo -s sip:[EMAIL PROTECTED] -H 
 192.168.1.46
 
 displays foo on the Snom display
 
 On my home LAN (AAH 1.5, Snom 190 3.60s, switched 100, no 
 VLAN, no routing) the same command (modified for my LAN) 
 always yields:
 
 (type: 3, code: 3): from 192.168.171.8 
 
 at the console of the sending machine. Same if I use FQDN.  
 Type 3 Code 3 means ICMP port unreachable 
 
 Doing a PCAP from the phone indicates that the Snom gets the 
 message, but nothing shows up in the SIP log. Doing tcpdump 
 on the originating machine yields something like Reply from 
 192.168.171.8  192.168.171.10 UDP port 5060 unreachable
 
 Same phones, same firmware rev, same version of sipsak, no 
 IPTABLES on the originating machine, DNS lookups work, call 
 behavior is normal, other SIP behavior like MWI works fine. I 
 got nuthin here, anyone got a tip?
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Re: [Asterisk-Users] Any way to not overwrite sound files on compile?

On Friday 30 September 2005 11:39, Kevin P. Fleming wrote:
 That is correct. 'make install' installs the standard sound files along
 with the binaries; if we did not do that, then when the code had been
 changed to required new sound files they would not be present...

So wrap the install binary such that it checks for the existence first.

 How about if we add a 'post-install' step in the Makefile, that would
 run a local script/program if specified, which could copy your sound
 files back into place?

Sounds tedious.  Why not simply emit cowardly refusing to overwrite existing 
sound file 'chilliconcarneexplosivegastrointestinalnoise.gsm' with a use 
make install-force to overwrite everything message?

-A.
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Re: [Asterisk-Users] zttest - 100% ?

2005-09-30 Thread DRi

Digium itself is saying their cards may work not properly with zttest 
results below 99,98
the card itself is working  the way that we can call out and receive 
calls, but we encountered massive echo-problems - sometimes more,
sometimes less even on lines within the same phone-provider and be sure 
that we've been messing around with all other possible
parameters for weeks without any result. Until now we've got a setup that 
we can live with at least until we get different hardware.
It's really worse calling someone and missing the name the called person 
said then picking up the phone in cause of echo-cancelling
parameters or even think the line is dead, or if you've got massive echoes 
and it takes about 30 seconds to filter them out if at all.

Dirk

[EMAIL PROTECTED] wrote on 30.09.2005 16:34:18:

 [EMAIL PROTECTED] wrote:
  just as an (bad) example:
  we are using an x206 and couldn't get the zttest above 99.975 
  equal what we were doing
  single irq, w/o acpi, w/o apic, different kernels, w/o 
  hyperthreading, different slots, a.s.o.
  for an Digium wildcard TE110P 
  
  so if someone got such a board to zttest 100% maybe could give some 
  information if where's something
  special to run asterisk on such boards...
  otherwise I think there are production differences on the 
ibm-mainboards 
  or the used chipsets
  
  we'll change hardware next...
 You don't have to have 100% on zttest.  You probably won't get it.  I 
 get the same results on one of my servers and  it runs perfectly.
 
 Kevin
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Re: [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality

2005-09-30 Thread steve



On Fri, 30 Sep 2005, Stephen Bosch wrote:

 How ironic that Allison (the woman who did the Digium prompts) is
 Canadian...


Heh.  Well, I didn't notice a prompt where she said aboot, eh?

I'll take my foot out my mouth now...

Steve

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Re: [Asterisk-Users] Any way to not overwrite sound files on compile?

2005-09-30 Thread Kevin P. Fleming


Matt wrote:

A post-install would be great (or I myself can write a script)... it
isn't that big of a deal.. I just wanted to see if I was over looking
something.   Tagging the sound directory for a version would also be
good but if there is no way (and I do understand the reasoning)
then I can just write a simple shell script to copy my files back and
keep them safe elsewhere.


It's in CVS HEAD now, it will look for /usr/sbin/asterisk-post-install 
and execute it.

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Re: [Asterisk-Users] Re:Any way to not overwrite sound files on compile?

2005-09-30 Thread Ron Wellsted

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Matt wrote:
 What about during an upgrade though? From one version to another where
 you actually want to replace asterisk?  I mean it isn't just a few
 files (I don't think?) that get copied when you do make install is it?
  Aren't there a fair amount?
 
 On 9/30/05, Gustavo A. Gonzalez [EMAIL PROTECTED] wrote:
 
When compile only type 'make'  and copy manually your module/s from asterisk
apps directory into your asterisk modules directory.

regards.
G.


Matt wrote:


Every time I recompile Asterisk (or upgrade to a new CVS-HEAD,
whatever) asterisk overwrites custom files I have made.  Granted,
these files are named the same as the asterisk default files
(vm-login.gsm, etc) because we had a person here record them to
customize them a bit more for our application.

Short of keeping them somewhere and copying them back every time
(which isn't all that often) I do a re-compile.  Is there some flag or
something to tell Asterisk not to install sound files, or at the very
least not to overwrite ones already existing?

Why not add a separate make sounds-install or make sounds-upgrade
step post-install.

It will never be possible to come up with a solution that will suit
everybody, but at least that would prevent customised files being
overwritten.

- --
Ron Wellsted
http://www.wellsted.org.uk
[EMAIL PROTECTED]
FWD:519961
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Re: [Asterisk-Users] Correction: Asterisk sound files, audio bandwidth, and sound quality

2005-09-30 Thread Stephen Bosch

Hi, Steve:

Thanks for your comments!

[EMAIL PROTECTED] wrote:
 The recorded prompts supplied with Asterisk are encoded with the .gsm 
 codec.  That makes them sound like audio sounds on your GSM cellphone.  
 Which is noticably worse than true PCM audio.
 
 Now in the telephone world best quality still isn't very good - its ulaw 
 or alaw encoded 8kHz audio.  That's frequency response up to 3.5kHz and 
 about 12 or 13 bits of dynamic range.
 
 But the fuzzyness you hear on the standard Asterisk prompts is due, I'm 
 sure, to the use of gsm compression.

snip

 Now Digium hasn't made the standard prompts available in a format other 
 than gsm.  I don't know why.
 
 For us we recorded the prompts in South African voice and so we have 
 those.  You need to either extract the original non-compressed prompts 
 from Digium (if they have them), or take it as an opportunity and record 
 your own set in Canadian accent.

How ironic that Allison (the woman who did the Digium prompts) is
Canadian...

We are planning to do our own prompts anyway, so that's not a problem --
your input has been most helpful; at least now I have some idea what our
options are.

Thanks!

-Stephen-

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[Asterisk-Users] CRITICAL PROBLEM

2005-09-30 Thread Tim McKee

I'm running a large number (125) remote sip phones for FEMA in the Gulf area
over satellite.  I've run into a major problem and need some assistance.

When dialing the FEMA voice response system, it appears that it never
actually answers the phone.  I never get audio when dialing via SIP through
a provider and when dialing over my PRI it actually times out with a phone
not ansered message, though the audio is passed.  Apparently the FEMA system
does not issue an 'ANSWERED' or 'CONNECTED' code back to the PSTN as it
should.  The link stays in an in-progress state until timeout occurs or the
user hangs up.

Is there any way to get SIP to pass audio prior to getting a call complete
message?  This is Asterisk CVS-HEAD 08-01-2004.

Please respond to [EMAIL PROTECTED] as I don't have good access to the
email account serving this list during the day.

Many thanks,

Tim McKee
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[Asterisk-Users] mISDN, HFC, W6692, one-way-voice problem

2005-09-30 Thread Kovács Attila



Hi All,

I'm trying to use a HFC chip ISDN modem with mISDN and chan_misdn.
The card is configured to NT, PmP mode.
One Siemens ISDN phone connected to the modem.
When I call the ISDN phone is called everything is just fine, but when I 
call from the ISDN phone I face some problems.
- There is no dial tone when I pick up the handset.
- I call a number but there is no ringtone and after I accept that call 
there is no voice on the channel.
I also tried to use PBX instead of the ISDN phone but exactly the same 
result.

I tried to use a W6692 chip based modem in TE, PmP mode connected 
to the phone company ISDN line.
When I initiated an outgoing call there was no problem, but when I 
received a call there was only silence on the line.

Thanks for Your help!

Attila


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http://zenearuhaz.t-online.hu/
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Re: [Asterisk-Users] Any way to not overwrite sound files on compile?

2005-09-30 Thread Kevin P. Fleming


Andrew Kohlsmith wrote:


So wrap the install binary such that it checks for the existence first.


Existence of what? The issue is that if we have a new version of a sound 
file, there's no way to know whether the one currently in place is 
'original' or modified.


Sounds tedious.  Why not simply emit cowardly refusing to overwrite existing 
sound file 'chilliconcarneexplosivegastrointestinalnoise.gsm' with a use 
make install-force to overwrite everything message?


Again, how would we decide when to generate this message? I think it's 
much easier (and already done G) to just let the user provide their 
own override script to put their files into place.

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