Re: [Asterisk-Users] Update: Welltech Wellgate 3504A registration problem

[Leo Ann Boon]

essentially, by not keeping to the same Call-ID or tag, asterisk has 
no way of matching the wellgate's register with the past proxy auth 
packet (and thus the hashed md5 token).
I've just gotten the box to register all 4-ports with an external SIP 
provider. The provider is running an old release of Broadsoft backend. 
Seems like Broadsoft supports this strange way of authentication.

this has been reported to WellTech, but we've yet to get a response 
from them.
They just sent me a new version of the 107a firmware. Have yet to check 
what's the difference from my running one.

i've fashioned a patch which solves this behaviour and it works fine 
under asterisk 0.9.0 on freebsd and 1.0-RC1 on linux.

i've got the patches for 0.9.0 and 1.0-RC1, and will be glad to email 
them if anyone wants them. the patch adds a new option under 
[general], 'usenonce'. if usenonce=yes (default is no, normal asterisk 
behaviour), then asterisk will use the nonce tag as sent by the 
wellgate to double check it's credentials and let it thru. this allows 
all 4 ports on the 3504A to be registered with a password.
Any other gateways with the same problem?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] panasonic KX-TD1232

[el Flynn]

P. K. wrote:
hi everyone,
Is it possible to serially connect my panasonic KX-TD1232 with a Linux
box(Asterisk installed) and have it working?
I've done just that - got it working with an X100 clone card, so I 
figured it'll work just fine with the Digium cards too.

BTW I was also able to fax from another * box, over IP, connected via 
IAX2 to the * and Panasonic combo. The TD1232 just thinks * is another 
extension.

Cheers.
Flynn
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] panasonic KX-TD1232

[Peter Svensson]

On Tue, 21 Sep 2004, P. K. wrote:

> Is it possible to serially connect my panasonic KX-TD1232 with a Linux
> box(Asterisk installed) and have it working?

We have done it with the E1 PRI card in the Panaconic. The Asterisk box
sits between the pstn and the kx-td1232 with two E1 connections.

Peter


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] panasonic KX-TD1232

[el Flynn]

Peter Svensson wrote:
On Tue, 21 Sep 2004, P. K. wrote:

Is it possible to serially connect my panasonic KX-TD1232 with a Linux
box(Asterisk installed) and have it working?

We have done it with the E1 PRI card in the Panaconic. The Asterisk box
sits between the pstn and the kx-td1232 with two E1 connections.
Peter
Actually, my installation was like:
PSTN -> Panasonic -> Asterisk
I realize that you'd have to do some programming on the TD1232 to 
emulate DID or something like that, but it just goes to show how 
flexible Asterisk is.

So you could hook up a 4-port FXO card from Digium, and program the 
TD1232 to treat those four "extensions" as one "number" I suppose. 
Haven't had time to look at the KX-TD1232 Programmator application to 
see if this really would work, but theoretically yes I guess.

Flynn
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] panasonic KX-TD1232

[Peter Svensson]

On Tue, 21 Sep 2004, el Flynn wrote:

> PSTN -> Panasonic -> Asterisk
> 
> I realize that you'd have to do some programming on the TD1232 to 
> emulate DID or something like that, but it just goes to show how 
> flexible Asterisk is.
> 
> So you could hook up a 4-port FXO card from Digium, and program the 
> TD1232 to treat those four "extensions" as one "number" I suppose. 
> Haven't had time to look at the KX-TD1232 Programmator application to 
> see if this really would work, but theoretically yes I guess.

To get caller id on the analog extension ports of the kx-td1232 you need
extra modules and you can not get did information at all. That information 
is available on the SDMR port so one could write a program that receives 
the data and passes it to asterisk. 

Another option would be to use E&M tie lines. I think the kx-td1232 
supports doing more advanced stuff with that interface. 

Doing it via isdn was a lot easier though. :-)

Peter

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Faxing thru freshtel

[Shaun Dwyer]

Hi,
I'm looking at connecting an analog fax to asterisk via an FXO card.
The plan is to send faxes thru freshtel.
Has anyone done faxing with freshtel?
Cheers,
-Shaun
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Update: Welltech Wellgate 3504A registration problem

[Dinesh Nair]

On 21/09/2004 15:07 Leo Ann Boon said the following:
I've just gotten the box to register all 4-ports with an external SIP 
provider. The provider is running an old release of Broadsoft backend. 
Seems like Broadsoft supports this strange way of authentication.
the 3504As do work with welltech's SIP proxy servers as well.
i checked the authentication code in chan_sip.c, and discovered that 
asterisk keeps track of the nonce (aka md5 hashed token) it sends out to 
registering users. when the users resend the register packet with the auth 
credentials, asterisk uses the Call-ID (or Call-ID and tag attribute if 
pedantic=yes) to match the second register packet with it's kept state to 
get the nonce it sent out the first time.

according to the RFC, different SIP users should be using different 
Call-IDs. as each of the 3504A's FXS ports register individually, they thus 
should be using different Call-IDs, however they don't but rather reuse the 
same Call-IDs. this is the behaviour which confuses asterisk as the 
wellgate registers.

(like i mentioned before, this is based on my reading of the RFC. i'm no 
SIP guru, so i may well be wrong in this analysis)

--
Regards,   /\_/\   "All dogs go to heaven."
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo "The opinions here in no way reflect the opinions of my $a $b."  |
| done; done  |
+=+
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] ./zaprtc.o: unresolved symbol ??

[Atuc]

hallo,
i tried to install the zaprtc.o module but get errors when i try to insmod it?
i use a th kernel 2.4.27, rtc is compiled as module but not loaded (see 
below lsmod),
does anybody has an idea what is wrong?

many thanks for help,
alex
snd:/usr/src/zaptelrtc# make
cc -c zaprtc.c -D__KERNEL__ -DMODULE -DEXPORT_SYMTAB -fomit-frame-pointer 
-O2 -Wall -I/usr/src/linux/include  -Wall -DMODVERSIONS -include 
/usr/src/linux/include/linux/modversions.h
gcc -s -Wall -Wstrict-prototypes rtctest.c -o rtctest
gcc -s -Wall -Wstrict-prototypes rtcsetup.c -o rtcsetup
rtcsetup.c: In function `main':
rtcsetup.c:25: warning: unused variable `rtc_tm'
rtcsetup.c:24: warning: unused variable `data'
rtcsetup.c:24: warning: unused variable `tmp'
rtcsetup.c:23: warning: unused variable `irqcount'
rtcsetup.c:23: warning: unused variable `i'
sync

snd:/usr/src/zaptelrtc# insmod ./zaprtc.o
./zaprtc.o: unresolved symbol zt_transmit
./zaprtc.o: unresolved symbol zt_receive
./zaprtc.o: unresolved symbol zt_register
snd:/usr/src/zaptelrtc# lsmod
Module  Size  Used byNot tainted
8139too11808   1
crc32   2848   0  [8139too]
unix   13924   2  (autoclean)
snd:/usr/src/zaptelrtc#
snd:/usr/src/zaptelrtc# cat /proc/ioports
-001f : dma1
0020-003f : pic1
0040-005f : timer
0060-006f : keyboard
0080-008f : dma page reg
00a0-00bf : pic2
00c0-00df : dma2
00f0-00ff : fpu
0170-0177 : ide1
01f0-01f7 : ide0
02f8-02ff : serial(set)
0320-0323 : wd7000
0350-0353 : wd7000
0376-0376 : ide1
03c0-03df : vga+
03f6-03f6 : ide0
03f8-03ff : serial(set)
0cf8-0cff : PCI conf1
b400-b4ff : PCI device 10ec:8139
  b400-b4ff : 8139too
d000-d0ff : PCI device 13f6:0111
d400-d40f : PCI device 10b9:5229
  d400-d407 : ide0
  d408-d40f : ide1
e400-e43f : PCI device 10b9:7101
e800-e81f : PCI device 10b9:7101
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] ./zaprtc.o: unresolved symbol ??

[Atuc]

At 10:05 21.09.2004, you wrote:
hallo,
i tried to install the zaprtc.o module but get errors when i try to insmod it?
i use a th kernel 2.4.27, rtc is compiled as module but not loaded (see 
below lsmod),
does anybody has an idea what is wrong?

many thanks for help,
alex

sorry, i found teh problem,
i had to recompile also the zaptel module for my new kernel first, no its 
loading,

alex 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re[2]: [Asterisk-Users] Update: Welltech Wellgate 3504A registration problem

[Danny Zak]

Hello Leo,

3802 is doing the same (it is the fxo one)


-- 
Best regards,
 Dannymailto:[EMAIL PROTECTED]

belGOnet.com  a  Euro-pictures  division - internet solutions
place princesse elisabeth 9/11   -   1030 Brussels  - Belgium
Tel : +32-(0)2-215.67.65  -  Fax : +32-(0)2-215.66.65

domains - hosting - hardware - VoiP - consultancy - backuping
CISCO - HP/COMPAQ - SUN - EMC - JUNIPER - IBM - DELL - NORTEL


No  legal  consequences  can be derived from the contents of the email
neither  is  belGOnet.com committed to them. The content of this email
is  exclusively  intended  for  adressee(s)  and information purposes.
belGOnet.com  accepts  no  liability for any damage resulting from the
use and/or acceptation of the content of this email.


Tuesday, September 21, 2004, 9:07:00 AM, you wrote:


>>
>> essentially, by not keeping to the same Call-ID or tag, asterisk has
>> no way of matching the wellgate's register with the past proxy auth
>> packet (and thus the hashed md5 token).

LAB> I've just gotten the box to register all 4-ports with an external SIP
LAB> provider. The provider is running an old release of Broadsoft backend.
LAB> Seems like Broadsoft supports this strange way of authentication.

>>
>> this has been reported to WellTech, but we've yet to get a response
>> from them.

LAB> They just sent me a new version of the 107a firmware. Have yet to check
LAB> what's the difference from my running one.

>>
>> i've fashioned a patch which solves this behaviour and it works fine
>> under asterisk 0.9.0 on freebsd and 1.0-RC1 on linux.
>>
>> i've got the patches for 0.9.0 and 1.0-RC1, and will be glad to email
>> them if anyone wants them. the patch adds a new option under 
>> [general], 'usenonce'. if usenonce=yes (default is no, normal asterisk
>> behaviour), then asterisk will use the nonce tag as sent by the 
>> wellgate to double check it's credentials and let it thru. this allows
>> all 4 ports on the 3504A to be registered with a password.

LAB> Any other gateways with the same problem?

LAB> ___
LAB> Asterisk-Users mailing list
LAB> [EMAIL PROTECTED]
LAB> http://lists.digium.com/mailman/listinfo/asterisk-users
LAB> To UNSUBSCRIBE or update options visit:
LAB>http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] ./zaprtc.o: unresolved symbol ??

[Steven Critchfield]

On Tue, 2004-09-21 at 03:12, Atuc wrote:
> At 10:05 21.09.2004, you wrote:
> >hallo,
> >
> >i tried to install the zaprtc.o module but get errors when i try to insmod it?
> >
> >i use a th kernel 2.4.27, rtc is compiled as module but not loaded (see 
> >below lsmod),
> >does anybody has an idea what is wrong?
> >
> >many thanks for help,
> >alex
> 
> 
> sorry, i found teh problem,
> 
> i had to recompile also the zaptel module for my new kernel first, no its 
> loading,

BTW, insmod only loads the requested module. modprobe will load any
dependencies as well as the requested module. Fairly redimentary linux
administivia.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Faxing thru freshtel

[Youness El Andaloussi]

Not sure about freshtel, but if it is a sip provider, make sure the 
protocol is lossless, such as ulaw

Has anyone done faxing with freshtel?
Youness 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] TDM400P: RJ45 to RJ11

[Nguyen Quang Hoa]

Hello

 

I am using a TDM400P-4FXO to connect my Asterisk to
telephone line. However, this TDM400P uses RJ45 connection while our telephone
standard uses RJ11. How can I wire the cable for the connection?

 

Thanks

Hoa

 








---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.752 / Virus Database: 503 - Release Date: 9/3/2004
 
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Failed to authenticate on INVITE

[Whisker, Peter]

For 
info
 
The 
new chan_sip2.c and recent CVS (yesterday) fix this and I can now use Asterisk 
to make calls on the sip.btcommunicator.bt.net service. If anyone wants help 
with the settings, e-mail me off list.
 
:)
 
Peter
 
-Original Message-From: Whisker, Peter 
[mailto:[EMAIL PROTECTED]Sent: 17 September 2004 
14:40To: 'Asterisk Users Mailing List - Non-Commercial 
Discussion'Subject: RE: [Asterisk-Users] Failed to authenticate on 
INVITE
I am 
getting this also.
 
I am 
trying to get Asterisk to talk similarly to BT Communicator to the BT server. I 
can register but then the INVITE fails.
 
BT are 
mixed up with their domains (in fact in the INVITE their software has a To: 
header with @domain1 and an auth URI referencing 
@domain2. The realm is domain1.) This can't be done in Asterisk 
where it is consistent about the URI.
 
I had 
been blaming this, but if you are having problems too...
 
I get 
the standard 407 header requesting Proxy Auth for the call. Asterisk submits the 
INVITE with auth and after the usual "Trying" I just get another 407. I have 
traces of Asterisk and the client which works and they seem so similar in what 
they do. I have made all the port ranges the same too. BT Communicator fails if 
you use port 5060 for the SIP client - they use 5052.
 
Peter
 
 
 
-Original Message-From: Stig Thune 
[mailto:[EMAIL PROTECTED]Sent: 17 September 2004 
12:55To: [EMAIL PROTECTED]Subject: 
[Asterisk-Users] Failed to authenticate on INVITE
NOTICE[98310]: chan_sip.c:6638 handle_response: 
Failed to authenticate on INVITE to 
'sip:[EMAIL PROTECTED];tag=as0f1d3429'
 
 
 
sip.conf

 
register => 
1234:[EMAIL PROTECTED]

 
 
 
 
extension.conf
--
 
;; Own extensions;exten => 
0852509516,1,Goto(resepsjon-own,s,1)
 
;[resepsjon-own]; exten => 
s,1,Answer exten => s,2,SetMusicOnHold(default) exten => 
s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten 
=> 
s,5,Background(own/choose) 
; Meny, 1 for support, 2 for support, 3 for wx3 exten => 
s,6,Wait(1) exten => 
s,7,Background(own/choosenumber)   
; dialer pushes a # ,and being sent 
to..    
; ip-phone must be picked up in ,2ms,tr   or hangup exten 
=> 1,1,Goto(privatanslutningar,s,1) exten => 
2,1,Goto(foretagsanslutningar,s,1)
 
 ; #=hangup exten => 
#,1,Playback(custom/no-key-registered) exten => 
#,2,Hangup
 
 exten => 
t,1,Goto(#,1) ; If they take too 
long, give up exten => i,1,Playback(invalid) ; "That's not valid, 
try again" inmenu]
 
;[privatanslutningar]; exten => 
s,1,Answer exten => s,2,SetMusicOnHold(default) exten => 
s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten 
=> 
s,5,Background(own/privatanslutningar) 
; Meny, 1 for support, 2 for support, 3 for 
wx3    
; dialer pushes a # ,and being sent to..
 
 exten => 1,1,Answer exten => 
1,2,Queue(help-privatanslutningar-queue) exten => 
2,1,Answer exten => 
2,2,Queue(order-privatanslutningar-queue) exten => 
3,1,Answer exten => 
3,2,Queue(info-privatanslutningar-queue)
 
 ; #=hangup ;exten => 
#,1,Playback(custom/no-key-registered) ;exten => 
#,2,Hangup
 
 exten => 
t,1,Queue(general-privatanslutningar-queue)   
; If they take too long, give up exten => 
i,1,Playback(invalid) 
; "That's not valid, try again" inmenu]
 
;[foretagsanslutningar]; exten => 
s,1,Answer exten => s,2,SetMusicOnHold(default) exten => 
s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,15 exten 
=> 
s,5,Background(own/foretagsanslutningar)   
; Meny, 1 for support, 2 for support, 3 for 
wx3    
; dialer pushes a # ,and being sent 
to..    
; ip-phone must be picked up in ,2ms,tr   or hangup exten 
=> 1,1,Answer exten => 
1,2,Queue(info-bedriftsanslutningar-queue) exten => 
2,1,Answer exten => 
2,2,Queue(help-bedriftsanslutningar-queue) exten => 
3,1,Answer exten => 
3,2,Queue(error-bedriftsanslutningar-queue)
 
 ; #=hangup ;exten => 
#,1,Playback(custom/no-key-registered) ;exten => #,2,Hangup
 
 exten => 
t,1,Queue(general-bedriftsanslutningar-queue) 
; If they take too long, give up exten => 
i,1,Playback(invalid) 
; "That's not valid, try again" inmenu]
--
 
 
The call gets into queue, then... the other phone 
rings.. and when I pick up - I get this message:
NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate 
on INVITE to 'sip:[EMAIL PROTECTED];tag=as0f1d3429'
 
I know that the register => works.. I have checked with my SIP-provider, 
and they say that it is logged in.
 
What else can be wrong ?
 
/ Stig He

Re: [Asterisk-Users] TDM400P: RJ45 to RJ11

[Dave Cotton]

On Tue, 2004-09-21 at 10:55 +0200, Nguyen Quang Hoa wrote:
> Hello
> 
>  
> 
> I am using a TDM400P-4FXO to connect my Asterisk to telephone line.
> However, this TDM400P uses RJ45 connection while our telephone
> standard uses RJ11. How can I wire the cable for the connection?

The horrible answer is just plug the RJ11 into te RJ45 socket the
locating tag will line them up.

A better solution is to note the order of the colours of the cable and
remove the RJ11 and replace with an RJ45 using only the centre
connections (3-6).

-- 
Dave Cotton <[EMAIL PROTECTED]>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] uk caller id

[Kevin Walsh]

Graham Turner [EMAIL PROTECTED] lazily top-posted:
> i have installed asterisk / zaptel from cvs distribution as of 17/09/04
> so i assume this does it 
> 
If you have a TDM/FXO then you'll need the latest CVS code.  If you
have a X100P then you'll need any CVS (the latest is usually a good
choice) and some patches.  I have the X100P running with today's CVS
version and with UK (BT) Caller*ID support.

>
> have configured zapata.conf as per instruction but i would have expected
> to have seen the callerid on the asterisk console as it receives the call
> but then may be not ?? 
> 
> the relevant my extensions.conf is ;
> 
> exten => s,1,answer
> exten => s,2,Dial(SIP/1001|10)
> 
> it is quite possible that callerid is being seen by * but i would have
> expected it to have been echoed to the console or at least written to the
> CDR entries ??? 
>
There's no need to answer before dialling, btw.  The SIP phone's
answer will filter through the system.

Aside from that, the (UK) Caller*ID will only be available if you have
one of the setups described above (TDM with latest CVS code or X100P
with patches).

> 
> going a bit further on, the whole point of this exercise is to allow this
> CALLERID to be displayed on the console of a SIP peer (7940 ip phone) that
> is defined as an asterisk extension
> 
That will happen once you're set up, yes.

-- 
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Suppressing CallerID in .call files

[Christian Victor]

Christian Victor schrieb:
I am trying to suppres the transmission of my CallerID when I place a 
call using a .call file in /var/spool/asterisk/outgoing
Okay - now I have a little Progress. :-) Suppressing CallerID on a PRI 
is done by setting the CallingPres parameter. But unfortunately this 
seems to be impossible in .call files. At least setting

CallingPres: 32
in the .call file leads to not processing the file.
Has anybody an Idea how I could solve this problem?
Thanks,
Christian
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] CallerID in Queue

[Spoljar, Mario]

I suppouse that you are using AgentLogin application, try instead to use
AgentCallbackLogin. Here is example how I do call center application on
my site:

[extension.conf]
---

[macro-agent-login]
; Agent login
; ${ARG1} - Caller nubmber - same as agent number
exten => s,1,AgentCallbackLogin(${ARG1}|[EMAIL PROTECTED])

[macro-agent-logoff]
; Agent logoff
; ${ARG1} - Caller nubmber - same as agent number
exten => s,1,AgentCallbackLogin(${ARG1})

[macro-SetCLIDprefix]
; ${ARG1} - A number
; add prefix for recognision of called destination when sharing agents
; 
; ex. ** before CLID user called Helpdesk call queue
;
exten => s,1,NoOp(OriginalCLID:${CALLERIDNUM})

exten => s,2,GotoIf($["${CALLERIDNUM:0:1}" = "*"]?s|4:s|3)

exten => s,3,SetCallerID(${ARG1}${CALLERIDNUM})
exten => s,4,NoOp(ChangedCLID:${CALLERIDNUM}) 
exten => s,5,Goto(${MACRO_CONTEXT},${MACRO_EXTEN},$[${MACRO_PRIORITY} +
1]);


[callcenter]

; define how to call Zap agents through PRI 
; there should be listed all agent destitantions

exten => 3991,1,Dial(Zap/g1/3991)

exten => 3992,1,Dial(Zap/g1/3992)

exten => 3993,1,Dial(Zap/g1/3993)  

[CC_HelpDesk]
;
; before A number add ** like identifier that call is originated to
call center - 
; usefull when you share agents with more call queues, then agent can
see prefix 
; so it can recognise to whome is call placed 
;

exten => s,1,ResponseTimeout(15)  
exten => s,2,Wait(2)
exten => s,3,Answer
exten => s,4,Playback(GBS-CC/10); Play 'You
reached GBS IT call center'
exten => s,5,SetMusicOnHold(default)   
exten => s,6,Macro(SetCLIDprefix,**)

exten => s,7,DigitTimeout(5)
exten => s,8,Queue(hd-q|tn|30)  ; r- ring
instead of moh,
; t- transfer alowed, 
; n- after
timeout will exit this application and go to the next step

exten => s,9,Background(GBS-CC/2)   ; Play 'All
operators are busy...press 1 to leave message, 2 to keep waiting'

exten => 1,1,Voicemail(s5666) 

exten => 2,1,Goto(s,8)

exten => t,1,Background(Attendant/zauzeti)  ;   What to
do after Timout set in s,1

exten => i,1,Goto(CC_HelpDesk,s,9)  ;




[default]
; 5670 - prefix to agent logon
exten => 5670,1,Macro(agent-login,${CALLERIDNUM:3}) ; you can log in
yust from your own station, exaple My CLID= 3723991, strip first 3
digits,
; pass to macro
just 3991 (agent & station number)
; 5671 - prefix to logoff
exten => 5671,1,Macro(agent-logoff,${CALLERIDNUM:3}); call logoff
macro with my number equal to agent number, after prompt input password,
press twice #

; -
; HD call queue 
; extension to call my Help desk call queue
; -
exten => 5666,1, Goto(CC_HelpDesk,s,1)


[queues.conf]
-
[hd-q]

; managed calls from group HD

;
; Notice - if using following notationa:
;  member => Agent/@1 - roundrobin strategy doesnot work, you should put
each agent in configuration separately
;
!!
; 
music = default

strategy = random

timeout = 15

maxlen = 5

; group 1 (hd-main)
member => Agent/3991
member => Agent/3992
member => Agent/3993
member => Agent/3994
member => Agent/3999
; group 4 (hd-backup)
member => Agent/2170
member => Agent/2314
member => Agent/2603
member => Agent/2858
member => Agent/2216
member => Agent/2864
member => Agent/2688
member => Agent/2701
member => Agent/2952

[agents.conf]
;
;
..
group=1

agent => 3991,1234,HD-3991 ; 3991
agent => 3992,1234,HD-3992 ; 3992
agent => 3993,1234,HD-3993 ; 3993
agent => 3994,1234,HD-3994 ; 3994
agent => 3999,1234,HD-3999 ; 3999
..

Hope it will help you...



Mario Spoljar
IT TO Telecommunications
GBS IT
Croatia
 
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rolland
Wong
Sent: Monday, September 20, 2004 10:37 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] CallerID in Queue

How can I bring the Caller ID when the calls enter call queue and answer
by X- lite or kphone?

I've tried many configuration but no luck that it only shows the
AgentLogin's exten..

Thanks!

R Wong

The information transmitted is intended only for the person or entity to
which it is addressed and may contain confidential and/or privileged
material.
Any review, retransmission, dissemination or other use of, or taking of
any action in reliance upon, this information by persons or entities
other than the intended recipient is prohibited.
If you received this in error, please contact the sender and delete the
material from any computer.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



--

[Asterisk-Users] Re: passing octothorpe

[Randy Bush]

> The standard way to get around this is to use the doublehash (or 
> maybe doublepound but unlikely to be doubleoctothorpe) patch which 
> will allow you to press hash twice for transfer or once to send it to 
> the remote end.  IIRC you can also specify the timeout for it to wait 
> for the second hash.

aha!  and the latest is the year old one at
?

randy

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] spandsp / fax partially received

[Maurizio Marini]

at last (using libtiff 3.5.7 as Mike Machado suggested) i was able to get 
spandsp working on my debian sid 2.6.8.1, with * up to 18-08-2004 and 
bristuff 4a
i've tested only fax receiving and the file on /var/spool/asterisk-fax/ 
contains only the first 3-4 cm. of page sent

sending fax notified an 'ok' fax status

Maurizio 
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Update: Welltech Wellgate 3504A registration problem

[Dinesh Nair]

On 21/09/2004 16:13 Danny Zak said the following:
Hello Leo,
3802 is doing the same (it is the fxo one)
danny,
i've sent you the patch for 1.0-RC1 in a private email. could you apply 
that, rebuild asterisk and the test it with the 3802 to see if the problem 
goes away ?

--
Regards,   /\_/\   "All dogs go to heaven."
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo "The opinions here in no way reflect the opinions of my $a $b."  |
| done; done  |
+=+
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] zaptelrtc for 2.6.x

[asterisk]

Hello,

i would like to use Meetme and i need zaptelrtc for that, since i dont
have any USB devices or a card from digium.
I compiled it on Llinux 2.6.8 and all i got was a zaprtc.o which obviously
wont work with a 2.6 kernel:
~/zaptelrtc# make load
sync
modprobe zaptel
insmod ./zaprtc.o
insmod: error inserting './zaprtc.o': -1 Invalid module format
make: *** [load] Error 1


Is there a zaptelrtc version which will compile for 2.6?

Thanks, Mario


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Problems compiling CAPI module

[igil]

Hello all,

I a'm having a lot of troubles compiling
the CAPI driver for mi AVM card, model C2 with two ports.

I´m using Debian stable with kernel
2.4.18 (bf24), but i can't compile this driver.

I just followed the steps from http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI

and there we can read that is necesary
to select [*]CAPI2.0 filesystem support, but in my kernel, doing make menuconfig.

I can`t select this option, this option
it is not present on my kernel 2.4.18




 CAPI2.0 support 
[*] Verbose reason code reporting (Kernel size +=7K) 
[*] CAPI2.0 Middleware support (EXPERIMENTAL) 
 CAPI2.0 /dev/capi support 
[*] CAPI2.0 filesystem support 
 CAPI2.0 capidrv interface support 

The next thing i tried to do, is to
compile the package chan_capi, but it fails too.
a lot of errors apears when i do make,


debian-asterisk:/home/ismaelg/chan_capi-0.3.5#
make
gcc -pipe -Wall -Wmissing-prototypes
-Wmissing-declarations -g  -I/usr/src/aster
isk      -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686  -DCAPI_ES -DCAPI_GAIN -DCAP
I_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes
-Wno-missing-declarations -DCRYPTO
   -c -o chan_capi.o chan_capi.c
chan_capi.c:23: asterisk/features.h:
No such file or directory
chan_capi.c:24: asterisk/utils.h: No
such file or directory
chan_capi.c:62: warning: type defaults
to `int' in declaration of `AST_MUTEX_DEF
INE_STATIC'
chan_capi.c:62: warning: parameter names
(without types) in function declaration
chan_capi.c:62: warning: data definition
has no type or storage class
chan_capi.c:63: warning: type defaults
to `int' in declaration of `AST_MUTEX_DEF
INE_STATIC'
chan_capi.c:63: warning: parameter names
(without types) in function declaration
chan_capi.c:63: warning: data definition
has no type or storage class
chan_capi.c:64: warning: type defaults
to `int' in declaration of `AST_MUTEX_DEF
INE_STATIC'
chan_capi.c:64: warning: parameter names
(without types) in function declaration
chan_capi.c:64: warning: data definition
has no type or storage class
chan_capi.c:65: warning: type defaults
to `int' in declaration of `AST_MUTEX_DEF
INE_STATIC'
chan_capi.c:65: warning: parameter names
(without types) in function declaration
chan_capi.c:65: warning: data definition
has no type or storage class
chan_capi.c:66: warning: type defaults
to `int' in declaration of `AST_MUTEX_DEF
INE_STATIC'
chan_capi.c:66: warning: parameter names
(without types) in function declaration
chan_capi.c:66: warning: data definition
has no type or storage class
chan_capi.c:67: warning: type defaults
to `int' in declaration of `AST_MUTEX_DEF
INE_STATIC'
chan_capi.c:67: warning: parameter names
(without types) in function declaration
chan_capi.c:67: warning: data definition
has no type or storage class
chan_capi.c:68: warning: type defaults
to `int' in declaration of `AST_MUTEX_DEF
INE_STATIC'
chan_capi.c:68: warning: parameter names
(without types) in function declaration
chan_capi.c:68: warning: data definition
has no type or storage class
chan_capi.c: In function `_capi_put_cmsg':
chan_capi.c:106: `capi_put_lock' undeclared
(first use in this function)
chan_capi.c:106: (Each undeclared identifier
is reported only once
chan_capi.c:106: for each function it
appears in.)
chan_capi.c: In function `capi_echo_canceller':
chan_capi.c:181: `contrlock' undeclared
(first use in this function)
chan_capi.c: In function `capi_detect_dtmf':
chan_capi.c:231: `contrlock' undeclared
(first use in this function)
chan_capi.c: In function `capi_send_digit':
chan_capi.c:309: `contrlock' undeclared
(first use in this function)
chan_capi.c: In function `remove_pipe':
chan_capi.c:481: `pipelock' undeclared
(first use in this function)
chan_capi.c: In function `capi_hangup':
chan_capi.c:613: `usecnt_lock' undeclared
(first use in this function)
chan_capi.c: In function `capi_call':
chan_capi.c:685: `pipelock' undeclared
(first use in this function)
chan_capi.c: In function `capi_read':
chan_capi.c:826: structure has no member
named `delivery'
chan_capi.c:827: structure has no member
named `delivery'
chan_capi.c: In function `capi_write':
chan_capi.c:899: `capi_send_buffer_lock'
undeclared (first use in this function)
chan_capi.c: In function `capi_new':
chan_capi.c:1022: structure has no member
named `delivery'
chan_capi.c:1023: structure has no member
named `delivery'
chan_capi.c:1078: `usecnt_lock' undeclared
(first use in this function)
chan_capi.c: In function `capi_request':
chan_capi.c:1130: `iflock' undeclared
(first use in this function)
chan_capi.c:1146: `contrlock' undeclared
(first use in this function)
chan_capi.c: In function `find_pipe':
chan_capi.c:1181: `pipelock' undeclared
(first use in this function)
chan_capi.c: In function `pipe_frame':
chan_capi.c:1214: too few arguments
to function `ast_dsp_process'
chan_capi.c: In function `pipe_msg':
chan_capi.c:1347: `contrlock' undeclared
(first use in this function)
chan_capi.c:1499: structure has no member
named `delivery'
chan_capi.c:1500: structure has no membe

Re: [Asterisk-Users] zaptelrtc for 2.6.x

[William Suffill]

why not use ztdummy which doesn't require USB on 2.6.x? Uncomment it
in the zaptel make file and away you go =)
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] FreeBSD 100% cpu

[Jan Baggen]

Compiled Asterisk from FreeBSD port (0.9.0_2) 
When I start asterisk it uses 100% cpu. Searches on Google
say to comment the noload => chan_oss.so in modules.conf
But this is already commented. Make.conf contains some 
optimizations. 

modules.conf:
; Asterisk configuration file
;
; Module Loader configuration file
;

[modules]
autoload=yes
;
; If you want, load the GTK console right away.
; Don't load the KDE console since
; it's not as sophisticated right now.
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
;
; Intercom application is obsoleted by
; chan_oss.  Don't load it.
;
noload => app_intercom.so
;
; Explicitly load the chan_modem.so early on to be sure
; it loads before any of the chan_modem_* 's afte rit
;
load => chan_modem.so
load => res_musiconhold.so
;
; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
;noload => chan_oss.so
;
; Module names listed in "global" section will have symbols globally
; exported to modules loaded after them.
;
[global]
chan_modem.so=yes


/etc/make.conf
CPUTYPE=i686
CFLAGS= -O2 -pipe -funroll-loops


---
Jan Baggen - [EMAIL PROTECTED]
IP2 Internet BV / http://www.ip2.nl  

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] SIP termination in Brazil

[Johannes van Hulst]

Is there an up and running provider of SIP termination in Brazil?

I know that there are some people building on a SIP
termination solution.

 

But who as it up and running ?

 

Best regards,

 

Han






___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] ISDN problem: lacking dialtone

[Martin Mielke]

Hi all,
this is a rather "newbie-oriented" question, so please bear with me...
The system running Asterisk has been provided with an AVM FRITZ!Card 
PnP. SuSE Linux 9.0 recognizes it right after booting the system and it 
seems to be configured (MSN) correctly...

The hwinfo looks like this:
---
pbx:/etc/asterisk # hwinfo --isapnp
11: ISA(PnP) 01.0: 10300 ISDN Adapter
 [Created at isapnp.193]
 Unique ID: QQNm.4JPVYg4a1y4
 Hardware Class: isdn adapter
 Model: "AVM FRITZ!Card PnP"
 Vendor: AVM "AVM"
 Device: eisa 0x0900 "AVM ISDN-Controller FRITZ!Card"
 I/O Ports: 0x220-??? (rw,disabled)
 IRQ: 5 (disabled)
 Requires: capi4linux, i4l-base, i4l-isdnlog
 Driver Info #0:
   I4L Type: 8002/7 [AVM FRITZ!Card PnP]
 Driver Info #1:
   I4L Type: 27/2 [AVM FRITZ!Card PnP]
 Config Status: cfg=yes, avail=yes, need=no, active=unknown
---
My extensions.conf has a the following relevant lines:
---
TRUNK=Modem/g1  ; Trunk interface
TRUNKMSD=1  ; MSD digits to strip 
(usually 1 or 0)
.
.
ignorepat => 8
.
.
exten => _8.,1,Dial(${TRUNK}:${EXTEN})
---

The modems.conf defines the group like this:
---
group=1
msn=987654321
incomingmsn=987654321
device => /dev/ttyI0
device => /dev/ttyI1
.
.
.
---
Asterisk shows the following message when coming up:
---
 == Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulated 
Modem Driver)
   -- Configured modem /dev/ttyI0 with driver i4l (Linux ISDN)
   -- Configured modem /dev/ttyI1 with driver i4l (Linux ISDN)
 == Registered channel type 'Modem' (Generic Voice Modem Channel Driver)
---

so the isdn4linux drivers are correctly loaded. I know, CAPI should do 
better but I can't compile from the tarball (see my post about it)

When trying to dial the PSTN using the ISDN interface I get:
---
*CLI> -- Executing Dial("SIP/mmielke-8c8e", "Modem/g1:8123456789") 
in new stack
   -- Called g1:8123456789
Sep 21 14:20:53 WARNING[229391]: chan_modem_i4l.c:355 i4l_read: Device 
'/dev/ttyI1' lacking dialtone
   -- Hungup 'Modem[i4l]/ttyI1'
 == No one is available to answer at this time
---

..."lacking dialtone" is false. I just plugged an ISDN-phone to the line 
and tested it works perfectly...

So, any ideas? what am I doing wrong this time? ;)
TIA,
Martin
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Problems compiling CAPI module

[asterisk]

> Hello all,
>
> I a'm having a lot of troubles compiling the CAPI driver for mi AVM
> card,  model C2 with two ports.
>
> I´m using Debian stable with kernel 2.4.18 (bf24), but i can't compile
> this driver.
>
> I just followed the steps from
> http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI
>
> and there we can read that is necesary to select [*]CAPI2.0 filesystem
> support, but in my kernel, doing make menuconfig.
>
> I can`t select this option, this option it is not present on my kernel
> 2.4.18

Enable experimental at the top of your configuration file. Its explained
at onthe wiki page.
>
>
>  CAPI2.0 support
> [*] Verbose reason code reporting (Kernel size +=7K)
> [*] CAPI2.0 Middleware support (EXPERIMENTAL)
>  CAPI2.0 /dev/capi support
> [*] CAPI2.0 filesystem support
>  CAPI2.0 capidrv interface support
>
> The next thing i tried to do, is to compile the package chan_capi, but
> it  fails too.
> a lot of errors apears when i do make,
>
> debian-asterisk:/home/ismaelg/chan_capi-0.3.5# make
> gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g
> -I/usr/src/aster
> isk  -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686  -DCAPI_ES
> -DCAPI_GAIN  -DCAP
> I_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations
>  -DCRYPTO
>   -c -o chan_capi.o chan_capi.c
> chan_capi.c:23: asterisk/features.h: No such file or directory
> chan_capi.c:24: asterisk/utils.h: No such file or directory

Have you installed the asterisk sources? Or the devel package?


I am running debian, too.
I used my own 2.6 vanilla kernel, upgraded to unstable and used the cvs
asterisk release.Just make sure you have your kernel links set correctly and you 
should get
it up and running fairly easily.
Mario


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] TDM400P: RJ45 to RJ11

[Lyle Giese]

Carefull plug in your RJ11 and it will work.  
If you want to rewire to RJ45, use the middle two pins, 4&5.
 
Lyle
 

  - Original Message - 
  From: 
  Nguyen Quang Hoa 
  To: [EMAIL PROTECTED] 
  
  Sent: Tuesday, September 21, 2004 3:55 
  AM
  Subject: [Asterisk-Users] TDM400P: RJ45 
  to RJ11
  
  
  Hello
   
  I am using a TDM400P-4FXO to 
  connect my Asterisk to telephone line. However, this TDM400P uses RJ45 
  connection while our telephone standard uses RJ11. How can I wire the cable 
  for the connection?
   
  Thanks
  Hoa
   
  ---Outgoing mail is certified Virus Free.Checked by 
  AVG anti-virus system (http://www.grisoft.com).Version: 6.0.752 
  / Virus Database: 503 - Release Date: 9/3/2004
  
  

  ___Asterisk-Users 
  mailing 
  list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo 
  UNSUBSCRIBE or update options visit:   
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] zaptelrtc for 2.6.x

[asterisk]

> why not use ztdummy which doesn't require USB on 2.6.x? Uncomment it in
> the zaptel make file and away you go =)

Zaptel Makefile:
MODULES=zaptel tor2 torisa wcusb wcfxo wcfxs \
ztdynamic ztd-eth wct1xxp wct4xxp ztdummy


~# lsmod
Module  Size  Used by
zaptel221764  0
crc_ccitt   2144  1 zaptel
md5 4000  1
ipv6  254980  12
fcpci 502616  0
capi   18496  0
capifs  5864  2 capi
kernelcapi 48128  2 fcpci,capi
8250   30176  0
serial_core22432  1 8250
unix   27092  14


Asterisk error:
---
-- Goto (meeting,s,1)
Urgent handler
Sep 21 14:52:51 WARNING[1113521072]: chan_zap.c:757 zt_open: Unable to
open '/dev/zap/pseudo': No such device or addressSep 21 14:52:51 ERROR[1113521072]: 
chan_zap.c:6359 chandup: Unable to dup
channel: No such device or addressSep 21 14:52:51 WARNING[1113521072]: 
app_meetme.c:225 build_conf: Unable
to open pseudo channel - trying deviceSep 21 14:52:51 WARNING[1113521072]: 
app_meetme.c:228 build_conf: Unable
to open pseudo device-- Playing 'conf-invalid' (language 'en')


Since my module is loaded but not used, i guess the pseudo zaptel device
is still not active. Or is this asterisk error not reall zaptel related?
Thanks, Mario


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] FreeBSD 100% cpu

[Dinesh Nair]

On 21/09/2004 20:20 Jan Baggen said the following:
Compiled Asterisk from FreeBSD port (0.9.0_2) 
When I start asterisk it uses 100% cpu. Searches on Google
say to comment the noload => chan_oss.so in modules.conf
But this is already commented. Make.conf contains some 
optimizations. 
add 'noload => pbx_wilcalu' into modules.conf. that's what's causing the 
high CPU load.

--
Regards,   /\_/\   "All dogs go to heaven."
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo "The opinions here in no way reflect the opinions of my $a $b."  |
| done; done  |
+=+
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Basic ISDN Access

[Zara Trousk]

Hi All,

Here at my office we have two basic ISDN Access with two phone lines each.
We would like to install an Asterisk PBX. The question is: How can I connect the ISDN 
lines to Asterisk?
Is it possible to connect to a DIGIUM Wildcard TE410P card? Or this E1 card is only 
for PRI access? Any other suggested and tested hardware for a production box?

Let me know,

-Z
-- 
___
Sign-up for Ads Free at Mail.com
http://promo.mail.com/adsfreejump.htm

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Problems compiling CAPI module

[igil]

Hello all,

My asterisk works well, the problem
is how to configure the AVM c2 card.
To provide ISDN external calls.

When you say, "Just
make sure you have your kernel links set correctly",
what kind of links are you talking about?

Thanks.

Ismael Gil.







<[EMAIL PROTECTED]>

Enviado por: [EMAIL PROTECTED]
21/09/2004 14:36



Por favor, responda a
Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED]>





Para
<[EMAIL PROTECTED]>


cc



Asunto
Re: [Asterisk-Users] Problems compiling
CAPI module








> Hello all,
>
> I a'm having a lot of troubles compiling the CAPI driver for mi AVM
> card,  model C2 with two ports.
>
> I´m using Debian stable with kernel 2.4.18 (bf24), but i can't compile
> this driver.
>
> I just followed the steps from
> http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI
>
> and there we can read that is necesary to select [*]CAPI2.0 filesystem
> support, but in my kernel, doing make menuconfig.
>
> I can`t select this option, this option it is not present on my kernel
> 2.4.18

Enable experimental at the top of your configuration file. Its explained
at onthe wiki page.
>
>
>  CAPI2.0 support
> [*] Verbose reason code reporting (Kernel size +=7K)
> [*] CAPI2.0 Middleware support (EXPERIMENTAL)
>  CAPI2.0 /dev/capi support
> [*] CAPI2.0 filesystem support
>  CAPI2.0 capidrv interface support
>
> The next thing i tried to do, is to compile the package chan_capi,
but
> it  fails too.
> a lot of errors apears when i do make,
>
> debian-asterisk:/home/ismaelg/chan_capi-0.3.5# make
> gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g
> -I/usr/src/aster
> isk      -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686
 -DCAPI_ES
> -DCAPI_GAIN  -DCAP
> I_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations
>  -DCRYPTO
>   -c -o chan_capi.o chan_capi.c
> chan_capi.c:23: asterisk/features.h: No such file or directory
> chan_capi.c:24: asterisk/utils.h: No such file or directory

Have you installed the asterisk sources? Or the devel package?


I am running debian, too.
I used my own 2.6 vanilla kernel, upgraded to unstable and used the cvs
asterisk release.Just make sure you have your kernel links set correctly
and you should get
it up and running fairly easily.
Mario


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Can someone suggest

[m. smadi]

I am taking a course in nework resource management with a theoritical 
emphasis.  Can someone suggest an asterisk related theoritcal project?

Thanks
M. Smadi
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] zaptelrtc for 2.6.x

[Patrick]

On Tue, 2004-09-21 at 14:59, [EMAIL PROTECTED] wrote:
[snip]
> ~# lsmod
> Module  Size  Used by
> zaptel221764  0
> crc_ccitt   2144  1 zaptel
> md5 4000  1
> ipv6  254980  12
> fcpci 502616  0
> capi   18496  0
> capifs  5864  2 capi
> kernelcapi 48128  2 fcpci,capi
> 8250   30176  0
> serial_core22432  1 8250
> unix   27092  14
[snap]

Looks like a digium card driver, ztdummy or zaprtc is not loaded.

Regards,
Patrick

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Problems compiling CAPI module

[asterisk]

> Hello all,
>
> My asterisk works well, the problem is how to configure the AVM c2
> card. To provide ISDN external calls.
>
> When you say, "Just make sure you have your kernel links set
> correctly", what kind of links are you talking about?
>

The links in /usr/src/
Example:
lrwxrwxrwx   1 root src20 Sep 21 11:39 linux -> /usr/src/linux-2.6.8
lrwxrwxrwx   1 root src20 Oct 16  2004 linux-2.6 -> /usr/src/linux-2.6.8
drwxrwxr-x  19  500  500 4.0K Sep 21 13:08 linux-2.6.8


But in your case, it was missing the asterisk header files i guess?
Have you installed asterisk from cvs? If not, did you install asterisk-dev?


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Question/Future Request for Call Queues

[Paul van Brouwershaven]

I have some quetions/ideas for the Asterisk Call Queues system.
System information:
- Fedora Core 1
- Kernel 2.4.22-1.2115.nptl
- Asterisk CVS-HEAD-09/08/04-17:43:15
1.> I sould like it that if a user is in the que and the expected wait 
time is longer then xxx seconds or there are more then xxx callers. That 
there is played a sound from directory xxx with some product information 
(advertisement)

2.> You can specify a member sequence with an agument on the memeber 
function like this:

member => SIP/user,1 ;(ringing with first attempt)
member => SIP/someuser,2 ;(ringing with first attempt)
member => SIP/otheruser,3 ;(ringing with first attempt)
member => SIP/someotheruser,3 ;(ringing with first attempt)
But this sequence is not working as I aspected. I aspected that it's
working like:
member => SIP/user,1 ;(ringing with first attempt)
member => SIP/someuser,2 ;(ringing with second attempt)
member => SIP/otheruser,3 ;(ringing with third attempt)
member => SIP/someotheruser,3 ;(ringing with third attempt)
My strategy is currently set on ringall, beceuse the other options does 
not specify the option I like to have.

Regards,
Paul
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SIP termination in Brazil

[Daniel Bichara]

Hi Han,

Our company can offer you a SIP termination in Brazil up and
running.

Daniel


Johannes van Hulst wrote:

  
  
  
  
  
  
  Is there an up and
running provider of SIP termination in Brazil?
  I know that there are
some people building on a SIP
termination solution.
   
  But who as it up and
running ?
   
  Best regards,
   
  Han
  
  

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Agents on zap channels must acknowledge calls even with ackcall=no

[Patrick Conroy]

I sent this message last week, but it looks like it didn't go through.
 So, if anyone receives this more than once, you have my sincere
apologies.

Hello,

I upgraded to CVS-HEAD-09/10/04-19:07:18 over the weekend and now
agents that are logged
in on zap channels have to acknowledge ACD calls by pressing #, even
though I have

ackcall=no

in agents.conf.  This doesn't seem to be happening to agents on SIP
phones, and it this is the first time I have had an agent logged in on
a Zap channel since the upgrade.  Is there a new config param that I
am missing?

Thanks,
Patrick
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Problems compiling CAPI module

[igil]

Thanks,

My links,  are properly set, as
you told me, 

I just instaled Asterisk from de cvs,
first download it from the cvs and then compiling it.

Where could I find the asterisk header
files?

I just download asterisk in /usr/src/asterisk.

Regards from Madrid.

Ismael Gil.







<[EMAIL PROTECTED]>

Enviado por: [EMAIL PROTECTED]
21/09/2004 15:27



Por favor, responda a
Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED]>





Para
<[EMAIL PROTECTED]>


cc



Asunto
Re: [Asterisk-Users] Problems compiling
CAPI module








> Hello all,
>
> My asterisk works well, the problem is how to configure the AVM c2
> card. To provide ISDN external calls.
>
> When you say, "Just make sure you have your kernel links set
> correctly", what kind of links are you talking about?
>

The links in /usr/src/
Example:
lrwxrwxrwx   1 root src    20 Sep 21 11:39 linux -> /usr/src/linux-2.6.8
lrwxrwxrwx   1 root src    20 Oct 16  2004 linux-2.6
-> /usr/src/linux-2.6.8
drwxrwxr-x  19  500  500 4.0K Sep 21 13:08 linux-2.6.8


But in your case, it was missing the asterisk header files i guess?
Have you installed asterisk from cvs? If not, did you install asterisk-dev?


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Basic ISDN Access

[slwatts]

I dont know the card in question but
pri ISDN cards and BRI cards are two very different things

If you want to connect asterisk to an
ISDN 2e interface then you are going to need a BRI card (you can get dual
or quad interface cards)

You will also need to find out from
your telco how the ISDN 2 is presented - P2P or Point to multipoint? this
makes a difference with your card purchase (active as opposed to passive)
(or you could just request your telco to  change the presentation
to suit) I am no expert but have just gone through this process and only
partially understood it.

I am using the BT speedway card (Fritz)
and its working like a dream.

Sam

Zara Trousk <[EMAIL PROTECTED]> wrote
on 21/09/2004 14:01:13:

> Hi All,
> 
> Here at my office we have two basic ISDN Access with two phone lines
each.
> We would like to install an Asterisk PBX. The
question is: How can I connect 
> the ISDN lines to Asterisk?
> Is it possible to connect to a DIGIUM Wildcard
TE410P card? Or this E1 card is
> only for PRI access? Any other suggested and tested hardware for a
production box?
> 
> Let me know,
> 
> -Z
> -- 
> ___
> Sign-up for Ads Free at Mail.com
> http://promo.mail.com/adsfreejump.htm
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
Winckworth Sherwood Solicitors and 
Parliamentary Agents 
DX 148400 WESTMINSTER 5 : 35 Great Peter Street, London SW1P 3LR
Telephone 020 7593 5000 Fax 020 7593 5099

Confidentiality 
This email message and any attachments are confidential; they may be subject 
to legal professional privilege and are intended for the named recipient only. 
If you are not the named recipient, please return the message and enclosures 
immediately and delete them from your system.

Caution 
Before advice received only by email (whether by attachment or otherwise) may 
be relied on, the authenticity of the communication must be verified by means 
independent of email.

Regulation
The firm is regulated by the Law Society. 
Partners 
A list of partners is available for inspection at each office of the firm and 
on the firm's website at
www.winckworths.co.uk



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Basic ISDN Access

[hahnke]

Hi,
I don't know much about the Wilcard TE410P but I surely know it won't work
with a BRI Interface ... as you suppose it's only for PRI access.
For a BRI Interface you can find several solutions, the cheapest would be to
simply use a BRI (ISDN) Card but there are also some more special cards
with Quad- or Octo- BRI Connectors available, all depending on your demand.
For communication between * and your BRI cards you can use capi, misdn, i4l
or bristuff (zaphfc) drivers depending on your card. I would suggest to use 
a card
with a HFC - based chipset. These are cheap and work very well as far as I 
know.

You will find further information about it at http://www.voip-info.org and 
several other
websites.

bye



At 15:01 21.09.2004, you wrote:
Hi All,
Here at my office we have two basic ISDN Access with two phone lines each.
We would like to install an Asterisk PBX. The question is: How can I 
connect the ISDN lines to Asterisk?
Is it possible to connect to a DIGIUM Wildcard TE410P card? Or this E1 
card is only for PRI access? Any other suggested and tested hardware for a 
production box?

Let me know,
-Z
--
___
Sign-up for Ads Free at Mail.com
http://promo.mail.com/adsfreejump.htm
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] 1 extension entry for multiple purposes?

[Matthew Boehm]

Not to flame a respond, but I only count 13 lines, not 200.
Anyway, what you posted is exactly what I am trying to prevent.
Do you see how you had to put 2 SetCIDNum entries for 2 seperate
dial-out numbers? Why can I not make 1 SetCIDNum entry for all
outgoing numbers below it like I tried to do with the 's' extension?

All 200 of our extensions need to be seen to the outside world as the
same number (212-433-3344) but internally need to be seen as their
4 digit extension which has no outside world mapping (ie: no direct
number to extension).

Is it possible to have 1 SetCIDNum line for all outgoing calls?

Thanks,
Matthew
- Original Message - 
From: "Adam Goryachev" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Monday, September 20, 2004 8:59 PM
Subject: Re: [Asterisk-Users] 1 extension entry for multiple purposes?


> On Tue, 2004-09-21 at 08:14, Matthew Boehm wrote:
> > OK. So I removed all the callerid= from the sip.conf and Wiley's fix
works
> > perefectly. But I am back to where if I call out, the caller id shows up
as
> > my extension only.
> >
> > My fix, that didn't work:
> >
> > [global-outgoing]
> >  exten => s,1,SetCIDNum(212-433-3344)
> >  exten => _9212XXX,2,Dial(SIP/${EXTEN}@,15,tr)
> >  exten => _91XX,2,Dial(SIP/${EXTEN}@,15,tr))
> >
> > I figured that if I tacked an 's' extension before the pattern matching,
> > every outgoing pattern below the 's' would get that CID. But that didn't
> > work.
>
> Read on for the answer, but first, please trim your posts, you don't
> need to include 200 lines of quoted text, just the relevant portions.
>
> Now, you don't seem to understand how extensions work. Look on the wiki
> for some additional examples, but for now, once you choose an extension
> (s, i, or 243284 or _) then you will move through the priorities
> within that extension until the caller dials a new extension, or you hit
> a goto of some sort.
> Also, priority must be consecutive order starting at 1.
>  exten => _9212XXX,1,SetCIDNum(212-433-3344)
>  exten => _9212XXX,2,Dial(SIP/${EXTEN}@,15,tr)
>  exten => _91XX,1,SetCIDNum(212-433-3344)
>  exten => _91XX,2,Dial(SIP/${EXTEN}@,15,tr))
>
> That should work much better for you.
>
> PS, this post is not meant to flame, or be derogatory, but a push to
> encourage you to read/learn more.
>
> Regards,
> Adam
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] RDSI vs Analogic

[Rodolfo Grave]

Hi. I'm getting new lines for using with Asterisk. In my Telco they said 
I could choose between Analogic lines and RDSI lines... I've already 
bought a TDM400P with FXO modules. Can you give some hints on the 
differences between RDSI and normal Analogic lines? Would I have 
problems for using a RDSI line with the TDM? Any other issue in general?

Thanks in advance,
RODOLFO
---
avast! Antivirus: Outbound message clean.
Virus Database (VPS): 0439-0, 20/09/2004
Tested on: 21/09/2004 15:52:59
avast! - copyright (c) 2000-2004 ALWIL Software.
http://www.avast.com

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Queues & Transfers

[Ben Merrills]

If someone takes a call from a queue on a CISCO 7960,
then does a Transfer to another agent (using the transfer button), the queue
system seems to think they still have the call, and wont assign them another
call till the other agent finishes the transferred call. Is this a known bug? Is
it something that can be overcome?

 

Cheers,

 

Ben






___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SIP termination in Brazil

[Julio Arruda]

Daniel Bichara wrote:
Hi Han,
Our company can offer you a SIP termination in Brazil up and running.
Daniel
IAX2 Termination ? I'm looking for some in Campinas, Sao Paulo and Rio 
de Janeiro.

Johannes van Hulst wrote:
Is there an up and running provider of SIP termination in Brazil?
I know that there are some people building on a SIP termination solution.
But who as it up and running ?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] ZapRTC loading problems

[Matthew Boehm]

genius..pure genius..

ThomasNiesel=>karma = karma++

Thanks,
Matthew

- Original Message - 
From: "Thomas Niesel" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Tuesday, September 21, 2004 12:48 AM
Subject: Re: [Asterisk-Users] ZapRTC loading problems


> Hallo Matthew Boehm
> On Mon, 20 Sep 2004 16:54:33 -0500 you wrote:
>
> > I finally got 2.4 recompiled with RTC as a module:
> > Module  Size  Used byNot tainted
> > autofs 13684   0  (autoclean) (unused)
> > acenic241092   0  (unused)
> > iptable_filter  2412   0  (autoclean) (unused)
> > ip_tables  15864   1  [iptable_filter]
> > e100   62340   1
> > rtc 9084   0  (autoclean)
> >
> > Here is my compile/load output from zaptelrtc:
> >
> > [EMAIL PROTECTED] zaptelrtc]# make
> > cc -c
> > zaprtc.c -D__KERNEL__ -DMODULE -DEXPORT_SYMTAB -fomit-frame-pointer -O2
> > -Wal l -I/usr/src/linux-2.4/include -I../zaptel -Wall -DMODVERSIONS
> > -include/usr/src/linux-2.4/include/linux/modversions.h
> > gcc -s -Wall -Wstrict-prototypes rtctest.c -o rtctest
> > rtctest.c: In function `main':
> > rtctest.c:30: warning: implicit declaration of function `exit'
> > gcc -s -Wall -Wstrict-prototypes rtcsetup.c -o rtcsetup
> > rtcsetup.c: In function `main':
> > rtcsetup.c:31: warning: implicit declaration of function `exit'
> > rtcsetup.c:23: warning: unused variable `i'
> > rtcsetup.c:23: warning: unused variable `irqcount'
> > rtcsetup.c:24: warning: unused variable `tmp'
> > rtcsetup.c:24: warning: unused variable `data'
> > rtcsetup.c:25: warning: unused variable `rtc_tm'
> > sync
> >
> > [EMAIL PROTECTED] zaptelrtc]# ls
> > Makefile  README  rtcsetup  rtcsetup.c  rtctest  rtctest.c  zaprtc.c
> > zaprtc.o
> >
> > [EMAIL PROTECTED] zaptelrtc]# make load
> > sync
> > modprobe zaptel
> > insmod ./zaprtc.o
> > ./zaprtc.o: init_module: Input/output error
> > Hint: insmod errors can be caused by incorrect module parameters,
> > including invalid IO or IRQ parameters.
>
> IMHO you have to _replace_ rtc with zaprtc (unload rtc first, load
> zaprtc second)
>
> >   You may find more information in syslog or the output from dmesg
> > make: *** [load] Error 1
> >
> > [EMAIL PROTECTED] zaptelrtc]# insmod ./zaprtc.o
> > ./zaprtc.o: init_module: Input/output error
> > Hint: insmod errors can be caused by incorrect module parameters,
> > including invalid IO or IRQ parameters.
> >   You may find more information in syslog or the output from dmesg
> >
> > [EMAIL PROTECTED] zaptelrtc]# uname -a
> > Linux localhost.localdomain 2.4.20-8custom #1 SMP Mon Sep 20 14:39:43
> > CDT 2004 i686 i686 i386 GNU/Linux
> >
> > Any ideas on the I/O error? syslog and dmesg show the same error.
> > Nothing extra.
> >
> > Thanks,
> > Matthew
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> -- 
> Tho/\/\as
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SIP termination in Brazil

[Daniel Bichara]

Olá Julio,
Também oferecemos IAX2.
Daniel
Julio Arruda wrote:
Daniel Bichara wrote:
Hi Han,
Our company can offer you a SIP termination in Brazil up and 
running.

Daniel

IAX2 Termination ? I'm looking for some in Campinas, Sao Paulo and Rio 
de Janeiro.

Johannes van Hulst wrote:
Is there an up and running provider of SIP termination in Brazil?
I know that there are some people building on a SIP termination 
solution.
But who as it up and running ?

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP termination in Brazil

[Sebastian Nocetti]

I am interested too in termination using SIP to brazil, we need h.323 too...

Can you contact me?

Thanks

Sebastian. 

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Daniel Bichara
Enviado el: Martes, 21 de Septiembre de 2004 11:06 a.m.
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] SIP termination in Brazil

Olá Julio,

Também oferecemos IAX2.

Daniel

Julio Arruda wrote:

> Daniel Bichara wrote:
>
>> Hi Han,
>>
>> Our company can offer you a SIP termination in Brazil up and 
>> running.
>>
>> Daniel
>
>
> IAX2 Termination ? I'm looking for some in Campinas, Sao Paulo and Rio 
> de Janeiro.
>
>> Johannes van Hulst wrote:
>>
>>> Is there an up and running provider of SIP termination in Brazil?
>>>
>>> I know that there are some people building on a SIP termination 
>>> solution.
>>> But who as it up and running ?
>>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

---

Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.744 / Virus Database: 496 - Release Date: 2004-08-24
 

---

Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.744 / Virus Database: 496 - Release Date: 2004-08-24
 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Suppressing CallerID in .call files

[Jay Milk]

What about the "CallerID" parameter in the .call file?

http://www.voip-info.org/wiki-Asterisk+auto-dial+out

> -Original Message-
> From: Christian Victor [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, September 21, 2004 4:51 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Suppressing CallerID in .call files
> 
> 
> Christian Victor schrieb:
> 
> > I am trying to suppres the transmission of my CallerID when 
> I place a
> > call using a .call file in /var/spool/asterisk/outgoing
> 
> Okay - now I have a little Progress. :-) Suppressing CallerID 
> on a PRI 
> is done by setting the CallingPres parameter. But unfortunately this 
> seems to be impossible in .call files. At least setting
> 
> CallingPres: 32
> 
> in the .call file leads to not processing the file.
> 
> Has anybody an Idea how I could solve this problem?
> 
> Thanks,
> Christian
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED] 
> http://lists.digium.com/mailman/listinfo/aster> isk-users
> To 
> UNSUBSCRIBE or update options visit:
>
http://lists.digium.com/mailman/listinfo/asterisk-users



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] RDSI vs Analogic

[Marconi Rivello]

On Tue, 21 Sep 2004 15:52:57 +0200, Rodolfo Grave <[EMAIL PROTECTED]> wrote:
> Hi. I'm getting new lines for using with Asterisk. In my Telco they said
> I could choose between Analogic lines and RDSI lines... I've already
> bought a TDM400P with FXO modules. Can you give some hints on the
> differences between RDSI and normal Analogic lines? Would I have
> problems for using a RDSI line with the TDM? Any other issue in general?
> 
> Thanks in advance,
> 
> RODOLFO

I don't know the (practical) difference, but this will help others
answer your question:

RDSI (portuguese, and as you are from es i believe in spanish too :))
= ISDN (english)

Marconi.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk and Red Hat 9

[Gary Carr]

I am running a P4 2.8 with 1 gig of ram and 7200 
rpm IDE drives. No bottlenecks as yet.
 
 
 
 
Gary
 

  - Original Message - 
  From: 
  Henry Devito 
  
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Monday, September 20, 2004 8:25 
  PM
  Subject: RE: [Asterisk-Users] Asterisk 
  and Red Hat 9
  
  
  Thank you for all of 
  the replies.  I would like to build a PBX with a 16 channel pri and 36 
  phones.  What kind of processor and memory should I look at? 
  
   
  
  
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Gary CarrSent: Monday, September 20, 2004 11:16 
  AMTo: Asterisk Users Mailing 
  List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Asterisk 
  and Red Hat 9
   
  
  I am running RH9 with a 4 port and 
  1 port ISDN cards. Not problems that I am aware of 
  yet.
  
   
  
   
  
   
  
  Gary
  
   
  

- Original Message - 


From: Henry Devito 


To: [EMAIL PROTECTED] 


Sent: Sunday, 
September 19, 2004 6:30 PM

Subject: 
[Asterisk-Users] Asterisk and Red Hat 9

 
Hi everyone,  I’m a newbie 
to Asterisk.  Will Asterisk run on RH9, easily or does it have to run 
on FreeBSD?   Will the drivers for the Digium cards work on RH9? 




___Asterisk-Users 
mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo 
UNSUBSCRIBE or update options visit:   
http://lists.digium.com/mailman/listinfo/asterisk-users
  
  

  ___Asterisk-Users 
  mailing 
  list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo 
  UNSUBSCRIBE or update options visit:   
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] ISDN problem: lacking dialtone

[Thomas Niesel]

Hallo Martin Mielke
On Tue, 21 Sep 2004 14:32:34 +0200 you wrote:

...cut

> 
> so the isdn4linux drivers are correctly loaded. I know, CAPI should do 
> better but I can't compile from the tarball (see my post about it)
> 
> 
> When trying to dial the PSTN using the ISDN interface I get:
> ---
> *CLI> -- Executing Dial("SIP/mmielke-8c8e", "Modem/g1:8123456789") 
> in new stack
> -- Called g1:8123456789
> Sep 21 14:20:53 WARNING[229391]: chan_modem_i4l.c:355 i4l_read: Device 
> '/dev/ttyI1' lacking dialtone
> -- Hungup 'Modem[i4l]/ttyI1'
>   == No one is available to answer at this time
> ---
> 
> ..."lacking dialtone" is false. I just plugged an ISDN-phone to the line
> 
> and tested it works perfectly...

Does the phone had the same MSN?
Is there maybe a PBX needs a leading "Digit" to get outside line?
Try your settings by using minicom first.
There is a good manual from i4l to call yourself via ttyI0/ttyI1 with
minicom.
> 
> So, any ideas? what am I doing wrong this time? ;)
> 
-- 
Tho/\/\as
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Segmentation Fault TDM22B & TDM04B

[Carlos Medina]

Hi all, i have installed two digium cards on my asterisk box a TDM04B & TDM22B. The channels are configured as show below:
 
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)Channel 02: FXO Kewlstart (Default) (Slaves: 02)Channel 03: FXS Kewlstart (Default) (Slaves: 03)Channel 04: FXS Kewlstart (Default) (Slaves: 04)Channel 05: FXS Kewlstart (Default) (Slaves: 05)Channel 06: FXS Kewlstart (Default) (Slaves: 06)Channel 07: FXS Kewlstart (Default) (Slaves: 07)Channel 08: FXS Kewlstart (Default) (Slaves: 08)
8 channels configured.
When i load the cards everything its fine, and the status of both cards is OK. The leds are green except the two FXO ports.
 
The problem is when i try to load asterisk appears a segmentation fault, here is the error:
 
"Sep 21 10:40:47 WARNING[1074404032]: chan_zap.c:7658 setup_zap: Ignoring faxdetectSep 21 10:40:47 WARNING[1074404032]: chan_zap.c:665 zt_open: Unable to specify channel 1: No such deviceSep 21 10:40:47 ERROR[1074404032]: chan_zap.c:5340 mkintf: Unable to open channel 1: No such devicehere = 0, tmp->channel = 1, channel = 1Sep 21 10:40:47 ERROR[1074404032]: chan_zap.c:7377 setup_zap: Unable to register channel '1-2'Sep 21 10:40:47 WARNING[1074404032]: loader.c:313 ast_load_resource: chan_zap.so: load_module failed, returning -1  == Unregistered channel type 'Tor'  == Unregistered channel type 'Zap'Segmentation fault"
Thanks for your help.
 
Carlos Andres Medina
		Do you Yahoo!?
Yahoo! Mail Address AutoComplete - You start. We finish.___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Suppressing CallerID in .call files

[Christian Victor]

What about the "CallerID" parameter in the .call file?
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
Yes - that was what I thought too. But unfortunately leaving out the 
parameter or setting it to '' will cause transmission of the default 
number (usually subscriber number + 0)

On a PRI you have to actively deny the presentation of the CallerID. In 
extensions you can do this by setting CallingPres=32 but that seems to 
make the .call files fail.

Chris
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Suppressing CallerID in .call files

[Peter Svensson]

On Tue, 21 Sep 2004, Jay Milk wrote:

> What about the "CallerID" parameter in the .call file?
> 
> http://www.voip-info.org/wiki-Asterisk+auto-dial+out

I'm not the original poster, but I think this will not work. Just changing 
the Calling Number (where the callerid field ends up in the isdn setup 
message) to nothing will most of the time just make the pstn provider set 
it to the default. In fact, setting the Calling Number outside your did 
will normally make the pstn set the main number as the caller id.

Isdn provides a means to supress caller id with a similar effect to what 
you can get on an analog line by dialing a prefix. I think this is what 
the original poster is after.

Peter

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] 1 extension entry for multiple purposes?

[Adam Goryachev]

On Tue, 2004-09-21 at 23:49, Matthew Boehm wrote:
> Not to flame a respond, but I only count 13 lines, not 200.

That is 13 lines that *I* quoted from your email, you quoted a lot more
from the previous email (the entire email in fact)

> Anyway, what you posted is exactly what I am trying to prevent.
> Do you see how you had to put 2 SetCIDNum entries for 2 seperate
> dial-out numbers? Why can I not make 1 SetCIDNum entry for all
> outgoing numbers below it like I tried to do with the 's' extension?

No, I put 2 setcidnum entries, one for each destination prefix...

> All 200 of our extensions need to be seen to the outside world as the
> same number (212-433-3344) but internally need to be seen as their
> 4 digit extension which has no outside world mapping (ie: no direct
> number to extension).
> 
> Is it possible to have 1 SetCIDNum line for all outgoing calls?
> >  exten => _9212XXX,1,SetCIDNum(212-433-3344)
> >  exten => _9212XXX,2,Dial(SIP/${EXTEN}@,15,tr)
> >  exten => _91XX,1,SetCIDNum(212-433-3344)
> >  exten => _91XX,2,Dial(SIP/${EXTEN}@,15,tr))

Hmmm, thinking... perhaps something like this:
exten => _9.,1,SetCIDNum(212-433-3344)
exten => _9.,2,Dial(SIP/${EXTEN}@,15,tr)

The only advantage to the previous config is that you don't need to wait
DigitTimeout or whatever, for the dial to complete

Maybe this would work as well:
exten => _9,1,SetCIDNum(212444)
exten => _9,2,DISA(something??)


There are probably many other ways to accomplish this, I'm sure there
should be a simple method using a macro, and different contexts, but I
just can't think properly at the moment, (it's past my bedtime)...

PS, also, consider that finding the solution to your request might cost
you more than simply using the solution you already have. Though you
probably should use a variable instead of hard-coding the phone number
in multiple places... 

Regards,
Adam


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] RC1 still broken with Cisco 7960?

[Brian Cuthie]

After downloading the latest CVS head and testing it with the Cisco 7960 
(SIP v7.2), it appears that it's still necessary to patch rtp.c to avoid 
audio dropouts.

I'm quite sure my gateway provider is running an older version of 
Asterisk, and I suppose that this may be the root cause. But I mention 
the issue here because it seems like it would be a mistake to ship 
Asterisk 1.0 if it doesn't work properly with Cisco phones (as there are 
undoubtedly a lot of them out there).

-brian
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Suppressing CallerID in .call files

[Henry Devito]

Most LEC's & CLEC's, at least in our area, require sending a number (CSID)
before the call is completed.  This is do to E911 features and ANI.  If you
do not send a number the call will fail.  If you truly want to block caller
ID I would contact your carrier and they should be able to block it on their
level. 

Have a good day!

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christian
Victor
Sent: Tuesday, September 21, 2004 9:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Suppressing CallerID in .call files

> What about the "CallerID" parameter in the .call file?
> 
> http://www.voip-info.org/wiki-Asterisk+auto-dial+out

Yes - that was what I thought too. But unfortunately leaving out the 
parameter or setting it to '' will cause transmission of the default 
number (usually subscriber number + 0)

On a PRI you have to actively deny the presentation of the CallerID. In 
extensions you can do this by setting CallingPres=32 but that seems to 
make the .call files fail.

Chris
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Suppressing CallerID in .call files

[Henry Devito]

Most LEC's & CLEC's, at least in our area, require sending a number (CSID)
before the call is completed.  This is do to E911 features and ANI.  If you
do not send a number the call will fail.  If you truly want to block caller
ID I would contact your carrier and they should be able to block it on their
level.

Have a good Day!


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson
Sent: Tuesday, September 21, 2004 9:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Suppressing CallerID in .call files

On Tue, 21 Sep 2004, Jay Milk wrote:

> What about the "CallerID" parameter in the .call file?
> 
> http://www.voip-info.org/wiki-Asterisk+auto-dial+out

I'm not the original poster, but I think this will not work. Just changing 
the Calling Number (where the callerid field ends up in the isdn setup 
message) to nothing will most of the time just make the pstn provider set 
it to the default. In fact, setting the Calling Number outside your did 
will normally make the pstn set the main number as the caller id.

Isdn provides a means to supress caller id with a similar effect to what 
you can get on an analog line by dialing a prefix. I think this is what 
the original poster is after.

Peter

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] 1 extension entry for multiple purposes?

[Jerimiah Cole]

Matthew Boehm wrote:
> Not to flame a respond, but I only count 13 lines, not 200.
It's still obnoxious.
> All 200 of our extensions need to be seen to the outside world as the
> same number (212-433-3344) but internally need to be seen as their
> 4 digit extension which has no outside world mapping (ie: no direct
> number to extension).
>
> Is it possible to have 1 SetCIDNum line for all outgoing calls?
Here's a snippet from my dialplan that does exactly that:
; default context for 7960s
[pbx_tci_out]
include => pbx_tci_internal
include => pbx_tci_topstn
; internal extensions
[pbx_tci_internal]
exten => *101,1,Dial(SIP/pbx_tci_101,20,m)
exten => *101,2,Voicemail([EMAIL PROTECTED])
exten => *101,102,Voicemail([EMAIL PROTECTED])
exten => *102,1,Dial(SIP/pbx_tci_102,20,m)
exten => *102,2,Voicemail([EMAIL PROTECTED])
exten => *102,102,Voicemail([EMAIL PROTECTED])
exten => *103,1,Dial(SIP/pbx_tci_103,20,m)
exten => *103,2,Voicemail([EMAIL PROTECTED])
exten => *103,102,Voicemail([EMAIL PROTECTED])
exten => *104,1,Dial(SIP/pbx_tci_104,20,m)
exten => *104,2,Voicemail([EMAIL PROTECTED])
exten => *104,102,Voicemail([EMAIL PROTECTED])
exten => *109,1,SetVar(ALERT_INFO=)
exten => *109,2,Dial(SIP/pbx_tci_109,10,r)
exten => *109,3,voicemail([EMAIL PROTECTED])
exten => *109,103,voicemail([EMAIL PROTECTED])
exten => *110,1,Goto(pbx_tci_vm,s,1)
exten => *111,1,VoiceMailMain([EMAIL PROTECTED])
exten => *2,1,Voicemailmain(s${CALLERIDNUM})
exten => *2,2,Hangup()
exten => h,1,Hangup()
exten => i,1,Congestion()
; set clid for calls to pstn
[pbx_tci_topstn]
exten => _.,1,SetGroup(pbx_tci)
exten => _.,2,SetCallerID(Tularosa Communications <(505) 439-0200>)
exten => _.,3,SetVar(CALLFILENAME=${EXTEN}-${TIMESTAMP})
exten => _.,4,Monitor(wav,out-${CALLFILENAME},m)
exten => _.,5,Macro(tendigits,${EXTEN},dp_pbx_tci)
When a user dials an internal extension, the default clid info in the
sip.conf is used.  The _. pattern picks up all other extenstions
(external) and set's the clid appropriately.  I even use the internal
clid info to bypass the voicemail prompts.
Jerimiah
Tularosa Communications
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Suppressing CallerID in .call files

[Christian Victor]

Hi!
What about the "CallerID" parameter in the .call file?
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
I'm not the original poster, but I think this will not work. Just changing 
the Calling Number (where the callerid field ends up in the isdn setup 
message) to nothing will most of the time just make the pstn provider set 
it to the default. In fact, setting the Calling Number outside your did 
will normally make the pstn set the main number as the caller id.
Thats exactly my problem.
Isdn provides a means to supress caller id with a similar effect to what 
you can get on an analog line by dialing a prefix. I think this is what 
the original poster is after.
Wich is possible in Extensions by setting CallingPres=32. Now I am 
looking for a way of disabling presentation in .call files.

Chris
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Suppressing CallerID in .call files

[Christian Victor]

Henry Devito schrieb:
Most LEC's & CLEC's, at least in our area, require sending a number (CSID)
before the call is completed.  This is do to E911 features and ANI.  If you
do not send a number the call will fail.  If you truly want to block caller
ID I would contact your carrier and they should be able to block it on their
level. 
Right - I discussed this with my PSTN carrier lately. Now the Problem is 
that I want to decide on a per call basis if I send a CallerID or not.

Chris
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Voicemail forward to a remote server?

[Dominique Kull]

Anybody ever managed to implement a solution where one could forward a 
voicemail from one * server to another?

Dominique
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] ISDN problem: lacking dialtone

[Martin Mielke]

Thomas Niesel wrote:
[ snip ]
Does the phone had the same MSN?
 

I think so. It could dial outside without a problem...
Is there maybe a PBX needs a leading "Digit" to get outside line?
 

No, those are direct lines to the PSTN, so no "leading 0" (or whatever) 
is needed ...

Try your settings by using minicom first.
There is a good manual from i4l to call yourself via ttyI0/ttyI1 with
minicom.
 

The minicom-test doesn't work at all. I always get either "BUSY" or "NO 
CARRIER"...

More ideas?
Martin
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] more on spandsp and partially received fax

[Maurizio Marini]

more detailed output


Sep 21 15:54:31 DEBUG[1120357296]: pbx.c:1255 pbx_extension_helper: Launching 'RxFAX'
Sep 21 15:54:31 DEBUG[1120357296]: channel.c:1699 ast_set_read_format: Set channel 
Zap/1-1 to read format SLINR
Sep 21 15:54:31 DEBUG[1120357296]: channel.c:1666 ast_set_write_format: Set channel 
Zap/1-1 to write format SLINR
Changed from phase 0 to 1
Slow carrier up
Slow carrier down
Start receiving document
Changed from phase 1 to 4
Sending ident
>>> CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20
DIS:
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter and V.29
R8x7.7lines/mm and/or 200x200pels/25.4mm OK
2D coding OK
Scan line length: 215mm
Recording length: A4 (297mm)
Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85
R8x15.4lines/mm OK
Minimum scan line time for higher resolutions: T15.4 = T7.7
>>> DIS: 80 00 ce f0 80 80 01
HDLC underflow in state 9
Changed from phase 4 to 3
T4 timeout in state 9
Changed from phase 3 to 4
Sending ident
>>> CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20
DIS:
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter and V.29
R8x7.7lines/mm and/or 200x200pels/25.4mm OK
2D coding OK
Scan line length: 215mm
Recording length: A4 (297mm)
Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85
R8x15.4lines/mm OK
Minimum scan line time for higher resolutions: T15.4 = T7.7
>>> DIS: 80 00 ce f0 80 80 01
T2 timeout
Start receiving document
Sending ident
>>> CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20
DIS:
Preferred octets: 256
Can receive fax
Supported data signalling rates: V.27ter and V.29
R8x7.7lines/mm and/or 200x200pels/25.4mm OK
2D coding OK
Scan line length: 215mm
Recording length: A4 (297mm)
Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85
R8x15.4lines/mm OK
Minimum scan line time for higher resolutions: T15.4 = T7.7
>>> DIS: 80 00 ce f0 80 80 01
HDLC underflow in state 9
Changed from phase 4 to 3
Slow carrier up
<<< DCS: 83 00 c6 f0 80 80 00
DCS with final frame tag
In state 9
DCS:
Can receive fax
Selected data signalling rate: V.29, 9600bps
R8x7.7lines/mm and/or 200x200pels/25.4mm OK
2D coding OK
Scan line length: 215mm
Recording length: A4 (297mm)
Minimum scan line time: 0ms
Minimum scan line time for higher resolutions: T15.4 = T7.7
Get at 9600
Changed from phase 3 to 5
Fast carrier up
Coarse carrier frequency 1688.02 (84)
Fast carrier down
Fast carrier up
Coarse carrier frequency 1700.01 (66)
Training error 18.034745
Training succeeded (constellation mismatch 10.798613)
Fast carrier trained
Fast carrier down
Changed from phase 5 to 4
Start rx document - compression 2
Start rx page
>>> CFR: 84
HDLC underflow in state 5
Post trainability
Changed from phase 4 to 5
Fast carrier up
Coarse carrier frequency 1699.99 (66)
Training error 9.098502
Training succeeded (constellation mismatch 8.396236)
Fast carrier trained
Fast carrier down

Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 251 (got 3285, 
expected 1728).
Fax3Decode2D: (FakeInput): Uncompressed data (not supported) at scanline 252 (x 224).
Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 252 (got 224, expected 
1728).
Fax3Decode2D: (FakeInput): Bad code word at scanline 253 (x 1647).
Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 253 (got 1647, expected 
1728).
Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 254 (got 1776, 
expected 1728).
Fax3Decode2D: (FakeInput): Bad code word at scanline 255 (x 101).
Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 255 (got 101, expected 
1728).
Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 256 (got 2145, 
expected 1728).
Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 257 (got 1738, 
expected 1728).
Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 258 (got 2721, 
expected 1728).
Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 259 (got 1938, 
expected 1728).
Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 260 (got 2872, 
expected 1728).
Fax3Decode2D: (FakeInput): Bad code word at scanline 261 (x 555).
Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 261 (got 555, expected 
1728).
Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 262 (got 2526, 
expected 1728).
Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 265 (got 1729, 
expected 1728).
Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 266 (got 4882, 
expected 1728).
Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 268 (got 1885, 
expected 1728).
Fax3Decode2D: (FakeInput): Bad code word at scanline 270 (x 219).
Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 270 (got 219, expected 
1728).
Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 272 (got 4796, 
exp

Re: [Asterisk-Users] Suppressing CallerID in .call files

[Andrew Thompson]

Christian Victor wrote:
Wich is possible in Extensions by setting CallingPres=32. Now I am 
looking for a way of disabling presentation in .call files.
Can you not just have your .call file dial back through an extension 
passing in a parameter of the destination number, so that you can 
activate that setting as a part of the extension context?

Something like:
exten =>987NXXNXX, 1, (set calling pres, however you do that)
exten =>987NXXNXX, 2, Dial...
Or, are you doing something that I'm just missing?
--
Andrew Thompson
http://aktzero.com/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Suppressing CallerID in .call files

[Henry Devito]

Unfortunately in my area call-by-call blocking caller id is not an option on
ISDN PRI,  I did do a small work around for phones that I wanted to block
Caller ID from.  I took one of my DID numbers and made the CSID that number,
I had the LEC not send a name with that number.  I did not define that DID
in *,  by not defining the DID in * it caused the LEC to send a reject to
the calling party.  Depending on what LEC the calling party was using they
either got a busy signal or a recording saying that number is not valid. 

Hope this helps

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christian
Victor
Sent: Tuesday, September 21, 2004 9:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Suppressing CallerID in .call files

Henry Devito schrieb:

> Most LEC's & CLEC's, at least in our area, require sending a number (CSID)
> before the call is completed.  This is do to E911 features and ANI.  If
you
> do not send a number the call will fail.  If you truly want to block
caller
> ID I would contact your carrier and they should be able to block it on
their
> level. 

Right - I discussed this with my PSTN carrier lately. Now the Problem is 
that I want to decide on a per call basis if I send a CallerID or not.

Chris
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Need Help !!

[Daniel Eboa]

Hello to all,

 

I’m new user
of Asterisk. I’m running Asterisk on a RedHat 9 platform. Everything seems
to be ok but I got lot of error messages and I don’t know their meaning. Can
somebody help me ??

 

These are the
messages:

 

WARNING[163850]:
chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 102 (Non-critical
Request)

 

The server
displayed this message when the party I called picks up the phone. Immediately
after picking up the phone, the call drop.

 

-- Executing Dial("SIP/Daniel-7e41",
"SIP/[EMAIL PROTECTED]") in new stack

    -- Called [EMAIL PROTECTED]

    -- SIP/195.149.11.98-86d9 is ringing

    -- SIP/195.149.11.98-86d9 answered SIP/Daniel-7e41

    -- Attempting native bridge of SIP/Daniel-7e41 and
SIP/195.149.11.98-86d9

    -- Got SIP response 481 "Invalid CSeq
Number" back from 195.149.11.98

 

 

 

When a call
cannot be complete, this is what the server send as error.

 

-- Executing Dial("SIP/Daniel-3d52",
"SIP/[EMAIL PROTECTED]") in new stack

    -- Called [EMAIL PROTECTED]

    -- SIP/195.149.11.98-a4a6 is making progress
passing it to SIP/Daniel-3d52

    -- Got SIP response 500 "Internal Server
Error" back from 195.149.11.98

    -- SIP/195.149.11.98-a4a6 is circuit-busy

  == Everyone is busy/congested at this time

 

 

 

This error is
reporting also when the call cannot be completed. But the behave is different
with the previous error.

 

  -- Executing Dial("SIP/Daniel-57d4",
"SIP/[EMAIL PROTECTED]") in new stack

    -- Called [EMAIL PROTECTED]

    -- SIP/195.149.11.98-fda1 is making progress
passing it to SIP/Daniel-57d4

    -- Got SIP response 480 "Temporarily Not
Available" back from 195.149.11.98

    -- SIP/195.149.11.98-fda1 is circuit-busy

  == Everyone is busy/congested at this time

 

Thanks for your
help.

 

 



 






<>___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] sipura registration problem

[Mohammed Salim]

Hi everyone,

 

I’m having an odd problem with one of my sipura
boxes.  The box registers the first time with asterisk properly after being
plugged in.  After which, some of the subsequent registration tries fail and the
box becomes unregistered.  However, after a few hours, it finally successfully
re-registers and the cycle continues.  I have not been able to figure out the
problem but I’m running the same config for this sipura as I am for about
5 others.  I have not ruled out the possibility that the box might actually be
bad.  I am running software version 2.07e on all of the sipuras and this is the
only one that is causing problems.  

 

Any suggestions would be appreciated.

 





Regards,

Mohammed Salim

EZZI Telecom, Inc.

 

 





 






___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] chan_sccp/SEP.cnf.xml

[Jesse Tyler]

HI all:
I have spent a large amount of time configuring/installing phones 
connected to Asterisk. Halfway through the process I discovered that my 
Cisco7960 with 2 7914 expansions was not supported in the SIP protocol. 
After reverting to SCCP 6.0(4.0) I am now perplexed with the hassle of 
configuring SCCP to properly work with Asterisk.

So far I have gotten the phone to dial and receive calls from the other 
participating SIP 7.2 phones on the LAN. Obviously this verifies that 
chan_sccp is correctly installed and running. I have scoured the 
internet looking for documentation regarding the 7914 call modules with 
asterisk and turned up nothing.

So this is my question
Does anyone have any kind of documentation regarding 7960 phones with 
7914 expansion modules running on AsteriskPBX???

I also have the proper load file for the expansion module...
Any Help would be greatly appreciated,
Thanks,
Jesse Tyler
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: 1 extension entry for multiple purposes?

[Tom Ivar Helbekkmo]

"Matthew Boehm" <[EMAIL PROTECTED]> writes:

> Do you see how you had to put 2 SetCIDNum entries for 2 seperate
> dial-out numbers? Why can I not make 1 SetCIDNum entry for all
> outgoing numbers below it like I tried to do with the 's' extension?

You can, you just did it the wrong way.  ;-)

> Is it possible to have 1 SetCIDNum line for all outgoing calls?

Yup.  Here's one way:

[outgoing-init]
exten => _X.,1,SetCIDNum(212444)
exten => _X.,2,Goto(outgoing-doit,${EXTEN},1)

[outgoing-doit]
exten => _9212XXX,1,Dial(SIP/${EXTEN}@,15,tr)
exten => _91XX,1,Dial(SIP/${EXTEN}@,15,tr)

[outgoing]
include => outgoing-init

Then place your calls in the "outgoing" context.

-tih
-- 
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no  T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Voicemail forward to a remote server?

[Kris Boutilier]

I was having this thought also and I couldn't find any implementations.

Likely it could be done using the sendmail 'pipe to shell' facility,
combined with some kind of delivery receipt system and a few minor hacks on
app_voicemail.c

> -Original Message-
> From: Dominique Kull [mailto:[EMAIL PROTECTED]
> Sent: September 21, 2004 7:57 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Voicemail forward to a remote server?
> 
> 
> Anybody ever managed to implement a solution where one could 
> forward a 
> voicemail from one * server to another?
> 
> Dominique
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] New astGUIclient version released 1.0.4

[mattf]

Hello,

We've released another update to our Asterisk GUI Client suite: 1.0.4

http://astguiclient.sf.net/

Screen shots: http://astguiclient.sourceforge.net/screenshots.html

The client suite runs on both Windows and UNIX, includes the VICIDIAL
auto-dialer and is free as in GPL.
(the suite is not an asterisk configuration tool)
This package is geared towards Asterisk installations with SIP phones and
Zaptel devices.

In addition to many bug fixes, we've added a few new features to the
VICIDIAL application(time-zone restriction and another closer transfer
option) and we've streamlined and tuned many back-end scripts.

To upgrade from 1.0.3 you need to update all of the server apps and web
pages as well as use the new client apps, and run the update sql script and
a perl script to populate the timezone file(if you use vicidial).

Let me know what you think.

Thanks,

MATT---
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Voicemail forward to a remote server?

[John Baker]

Why not just use rsync or netcat?  There are about a dozen different 
ways to do this.

John
Kris Boutilier wrote:
I was having this thought also and I couldn't find any implementations.
Likely it could be done using the sendmail 'pipe to shell' facility,
combined with some kind of delivery receipt system and a few minor hacks on
app_voicemail.c

-Original Message-
From: Dominique Kull [mailto:[EMAIL PROTECTED]
Sent: September 21, 2004 7:57 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Voicemail forward to a remote server?
Anybody ever managed to implement a solution where one could 
forward a 
voicemail from one * server to another?

Dominique
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] T100P lost D channel

[Ross Donaldson]

Hi all,
I have a Wildcard that is flip floping between internally clocked and 
the PRI. It is showing Red Alarm/Recovering. After a long run around 
with the telco, they said I have lost the D channel on my side.  I am 
seeing this message:

 == Restart on requested on entire span 1
Sep 21 08:29:48 WARNING[1192491824]: chan_zap.c:5942 zt_pri_error: PRI: 
!! Got reject for frame 1, but we only have others!
 == D-Channel on span 1 down
 == D-Channel on span 1 up
 == Restart on requested on entire span 1
 == D-Channel on span 1 down

Has anybody seen anything similar? Is it possible I have a bad card?  
The card has been working perfectly for 6 months.  I don't want to buy a 
new card and still have the problem.

Here are my configs:
zaptel.conf
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
loadzone=us
defaultzone=us
zapata.conf
[channels]
switchtype=national
context=inbound-pri
signalling=pri_cpe
group=1
pickupgroup=1
rxgain=8.2
txgain=-1.0
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
channel => 1-23
Thanks


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] RDSI vs Analogic

[Michael Loftis]

I'm not sure if he means RDSI/ISDN and *ANALOG* (POTS)if 'analogic' 
means POTS then yes, he needs that ...  TDM400P is an POTS/Analog NOT ISDN 
device

--On Tuesday, September 21, 2004 11:28 -0300 Marconi Rivello 
<[EMAIL PROTECTED]> wrote:

On Tue, 21 Sep 2004 15:52:57 +0200, Rodolfo Grave <[EMAIL PROTECTED]>
wrote:
Hi. I'm getting new lines for using with Asterisk. In my Telco they said
I could choose between Analogic lines and RDSI lines... I've already
bought a TDM400P with FXO modules. Can you give some hints on the
differences between RDSI and normal Analogic lines? Would I have
problems for using a RDSI line with the TDM? Any other issue in general?
Thanks in advance,
RODOLFO
I don't know the (practical) difference, but this will help others
answer your question:
RDSI (portuguese, and as you are from es i believe in spanish too :))
= ISDN (english)
Marconi.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
Undocumented Features quote of the moment...
"It's not the one bullet with your name on it that you
have to worry about; it's the twenty thousand-odd rounds
labeled `occupant.'"
  --Murphy's Laws of Combat
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Voicemail forward to a remote server?

[Matthew Boehm]

Just have the two * servers login to eachother via IAX, then in your
extensions plan where you normally have:

 exten => 8899,1,Dial(SIP/8899,15,tr)
 exten => 8899,2,Voicemail([EMAIL PROTECTED])

change it to

 exten => 8899,1,Dial(SIP/8899,15,tr)
 exten => 8899,2,Dial(IAX2//)

We have two * servers setup, #1 as main and #2 as pure MeetMe. We use the
above IAX2 dial command to forward all calls on a paticular number to the
other server. Works great.

Matthew

- Original Message - 
From: "Dominique Kull" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 21, 2004 9:56 AM
Subject: [Asterisk-Users] Voicemail forward to a remote server?


> Anybody ever managed to implement a solution where one could forward a
> voicemail from one * server to another?
>
> Dominique
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Queue position and thankyou message plays even when queue is empty?

[Chris Icide]

I'm in the process of setting up a queue system where the position
message and thankyou message are required to play every 90 seconds. 
However, if a caller comes in to a queue with active agents logged in,
and no one else is in the queue, the messages play immediately, and
then the agents are polled.  Is there any configuration parameter that
will disable the playing of the messages if the queue is empty and
agents are available?

-Chris
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] ZAP problem / Strange State

[Brent Franks]

Hello,

I am receiving an error in my error logs any time I receive a call on
the third line in our hunt group.

Sep 20 13:15:03 WARNING[1116939584]: Ring/Off-hook in strange state 6 on
channel 3

The weird part is that the calls seem to work fine, just this error
message is logged.  Currently, I have a T100P connected to an Adtran TA
750.  I have punched down the lines again, reversed TIP and RING, and
still the same result.  Additionally, I was seeing the same exact error
on a different T100P connected to an Carrier Access ADIT 600.

There are no errors whenever it is an outgoing call.

No CallerID on the line, No Distinctive Ringing, etc?

I searched for this, and found some posts but there are no follow ups
offering resolution.

Does anyone know what this means or know how to resolve?

Thanks!

Brent D. Franks
Mindworks Internet Services



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] chan_sccp/SEP.cnf.xml

[Matthew Boehm]

If you are going to use the 7914 (which yes, unfortunatly isn't supported on
SIP, dammit Cisco) you might want to check out

http://chan-sccp.sourceforge.net

an alternative sccp module for *. Before we switched all our 7960's to SIP
we used this and it seemed alot better than the built in one.

Matthew
- Original Message - 
From: "Jesse Tyler" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 21, 2004 10:23 AM
Subject: [Asterisk-Users] chan_sccp/SEP.cnf.xml


> HI all:
>
> I have spent a large amount of time configuring/installing phones
> connected to Asterisk. Halfway through the process I discovered that my
> Cisco7960 with 2 7914 expansions was not supported in the SIP protocol.
> After reverting to SCCP 6.0(4.0) I am now perplexed with the hassle of
> configuring SCCP to properly work with Asterisk.
>
> So far I have gotten the phone to dial and receive calls from the other
> participating SIP 7.2 phones on the LAN. Obviously this verifies that
> chan_sccp is correctly installed and running. I have scoured the
> internet looking for documentation regarding the 7914 call modules with
> asterisk and turned up nothing.
>
> So this is my question
>
> Does anyone have any kind of documentation regarding 7960 phones with
> 7914 expansion modules running on AsteriskPBX???
>
> I also have the proper load file for the expansion module...
>
>
> Any Help would be greatly appreciated,
>
> Thanks,
>
>
> Jesse Tyler
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Voicemail forward to a remote server?

[Dominique Kull]

This certainly works, if you want to have a remote VM - but still does 
not forward a received VM to another server.

Dominique
Matthew Boehm wrote:
Just have the two * servers login to eachother via IAX, then in your
extensions plan where you normally have:
 exten => 8899,1,Dial(SIP/8899,15,tr)
 exten => 8899,2,Voicemail([EMAIL PROTECTED])
change it to
 exten => 8899,1,Dial(SIP/8899,15,tr)
 exten => 8899,2,Dial(IAX2//)
We have two * servers setup, #1 as main and #2 as pure MeetMe. We use the
above IAX2 dial command to forward all calls on a paticular number to the
other server. Works great.
Matthew
- Original Message - 
From: "Dominique Kull" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 21, 2004 9:56 AM
Subject: [Asterisk-Users] Voicemail forward to a remote server?


Anybody ever managed to implement a solution where one could forward a
voicemail from one * server to another?
Dominique
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
dominique kull
taridium.communications ltd
t: +44 207 731 1562
f: +44 207 900 6564
v: fwd 515151
w: http://taridium.com
e: [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Voicemail forward to a remote server?

[Kris Boutilier]

Agreed, however these rely on foreknowledge of the remote end configuration
and are non-transactional. I was thinking more along the lines of VPIM
(http://www.google.ca/search?q=%22Voice+Profile+for+Internet+Mail).

Consider a large-scale private networking scenario - it would be very nice
to have a fall back voicemail system running on the PSTN/Private Network
bridges that, in the event of a transport network or remote end outage,
could catch targeted incoming calls with a generic message, record a
voicemail and then pass it off to be delivered by some other existing
infrastructure. Ie. Large volumes of SMTP mail can be delivered in
circumstances where RTP protocols would be rendered useless such as over a
congested, temporary consumer-grade satellite connection.

> -Original Message-
> From: John Baker [mailto:[EMAIL PROTECTED]
> Sent: September 21, 2004 8:41 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Voicemail forward to a remote server?
> 
> 
> Why not just use rsync or netcat?  There are about a dozen different 
> ways to do this.
> 
> John
> 
> Kris Boutilier wrote:
> 
> > I was having this thought also and I couldn't find any 
> implementations.
> > 
> > Likely it could be done using the sendmail 'pipe to shell' facility,
> > combined with some kind of delivery receipt system and a 
> few minor hacks on
> > app_voicemail.c
> > 
{clip}
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] ISDN problem: lacking dialtone

[Thomas Niesel]

Hallo Martin Mielke
On Tue, 21 Sep 2004 17:03:54 +0200 you wrote:

> Thomas Niesel wrote:
> 
> [ snip ]
> 
> >
> >Does the phone had the same MSN?
> >  
> >
> 
> I think so. It could dial outside without a problem...
> 
> >Is there maybe a PBX needs a leading "Digit" to get outside line?
> >  
> >
> 
> No, those are direct lines to the PSTN, so no "leading 0" (or whatever) 
> is needed ...

Ok
> 
> >Try your settings by using minicom first.
> >There is a good manual from i4l to call yourself via ttyI0/ttyI1 with
> >minicom.
> >  
> >
> 
> The minicom-test doesn't work at all. I always get either "BUSY" or "NO 
> CARRIER"...

Same errror as on (*) !?
Cable/Card
> 
> More ideas?
Do you have isdnlog running? If so, call your self and watch syslog.
Even with minicom and configured MSN (at&e...) you should see something if
a call comes in on that MSN.
> 
> 
> Martin
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 


-- 
Tho/\/\as
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] cvs stable

[Michael Bielicki]

Stable seized to exist quite some time ago.


On Tue, 14 Sep 2004 16:35:28 +0500, Atif Rasheed
<[EMAIL PROTECTED]> wrote:
> on the asterisk site, it was stated while ago, how to download stable
> version. like
> cvs checkout -r v1-0_stable asterisk-addons zaptel libpri
> 
> but now it's not their. is stable-version removed from the CVS ?
> or is their some different procedure ?
> 
> thank you
> --
> Atif
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 



-- 
Michael Bielicki
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Cisco 7905

[M. Willigs]

Hi everybody.
I have a Cisco 7905 IP Phone and as I see, the device isn't send the
registration message to the server, so to receive calls need to configure
static ip address.
Is there some way to make the Cisco send any sip registration? or Is there
some way to make the Cisco phone receive calls whitout asociate it whith a
static IP address?

Thanks in advance

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] RC1 still broken with Cisco 7960?

[Michael Bielicki]

hmmm I have no problems with 7960's and lates CVS since weeks

On Tue, 21 Sep 2004 10:41:54 -0400, Brian Cuthie <[EMAIL PROTECTED]> wrote:
> 
> After downloading the latest CVS head and testing it with the Cisco 7960
> (SIP v7.2), it appears that it's still necessary to patch rtp.c to avoid
> audio dropouts.
> 
> I'm quite sure my gateway provider is running an older version of
> Asterisk, and I suppose that this may be the root cause. But I mention
> the issue here because it seems like it would be a mistake to ship
> Asterisk 1.0 if it doesn't work properly with Cisco phones (as there are
> undoubtedly a lot of them out there).
> 
> -brian
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 



-- 
Michael Bielicki
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Need Help !!

[Thomas Niesel]

Hallo Daniel Eboa
On Tue, 21 Sep 2004 16:16:44 +0100 you wrote:

> Hello to all,
> 
>  
> 
> I'm new user of Asterisk. I'm running Asterisk on a RedHat 9 platform.
> Everything seems to be ok but I got lot of error messages and I don't
> know their meaning. Can somebody help me ??
> 
>  
> 
> These are the messages:
> 
>  
> 
> WARNING[163850]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on
> call [EMAIL PROTECTED] for seqno 102
> (Non-critical Request)
> 
>  
> 
> The server displayed this message when the party I called picks up the
> phone. Immediately after picking up the phone, the call drop.

Ah, I remember... 
> 
>  
> 
> -- Executing Dial("SIP/Daniel-7e41", "SIP/[EMAIL PROTECTED]")
> in new stack
> 
> -- Called [EMAIL PROTECTED]
> 
> -- SIP/195.149.11.98-86d9 is ringing
> 
> -- SIP/195.149.11.98-86d9 answered SIP/Daniel-7e41
> 
> -- Attempting native bridge of SIP/Daniel-7e41 and
> SIP/195.149.11.98-86d9

Here (*) try to connect the 2 partys directly!

You should check the docu/wiki for sip and (can)reinvite.

This chapter is a bit tricky cause it will affect also your firewall.


...cut

-- 
Tho/\/\as
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Voicemail forward to a remote server?

[Dominique Kull]

I was thinking that there could be a way to do this through IAX, without 
re-encoding of course. You could for example specify a special extension 
on the remote server which would then pickup the stream like a regular VM.

Dominique
Kris Boutilier wrote:
Agreed, however these rely on foreknowledge of the remote end configuration
and are non-transactional. I was thinking more along the lines of VPIM
(http://www.google.ca/search?q=%22Voice+Profile+for+Internet+Mail).
Consider a large-scale private networking scenario - it would be very nice
to have a fall back voicemail system running on the PSTN/Private Network
bridges that, in the event of a transport network or remote end outage,
could catch targeted incoming calls with a generic message, record a
voicemail and then pass it off to be delivered by some other existing
infrastructure. Ie. Large volumes of SMTP mail can be delivered in
circumstances where RTP protocols would be rendered useless such as over a
congested, temporary consumer-grade satellite connection.

-Original Message-
From: John Baker [mailto:[EMAIL PROTECTED]
Sent: September 21, 2004 8:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Voicemail forward to a remote server?
Why not just use rsync or netcat?  There are about a dozen different 
ways to do this.

John
Kris Boutilier wrote:

I was having this thought also and I couldn't find any 
implementations.
Likely it could be done using the sendmail 'pipe to shell' facility,
combined with some kind of delivery receipt system and a 
few minor hacks on
app_voicemail.c
{clip}
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
dominique kull
taridium.communications ltd
t: +44 207 731 1562
f: +44 207 900 6564
v: fwd 515151
w: http://taridium.com
e: [EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] RDSI vs Analogic

[Rodolfo Grave]

OK, that's it. I wont use the RDSI/ISDN connection and will get the 
ANALOG :) (sorry about my english) lines.

Thanks a lot for your help.
RODOLFO
Michael Loftis wrote:
I'm not sure if he means RDSI/ISDN and *ANALOG* (POTS)if 
'analogic' means POTS then yes, he needs that ...  TDM400P is an 
POTS/Analog NOT ISDN device

--On Tuesday, September 21, 2004 11:28 -0300 Marconi Rivello 
<[EMAIL PROTECTED]> wrote:

On Tue, 21 Sep 2004 15:52:57 +0200, Rodolfo Grave 
<[EMAIL PROTECTED]>
wrote:

Hi. I'm getting new lines for using with Asterisk. In my Telco they 
said
I could choose between Analogic lines and RDSI lines... I've already
bought a TDM400P with FXO modules. Can you give some hints on the
differences between RDSI and normal Analogic lines? Would I have
problems for using a RDSI line with the TDM? Any other issue in 
general?

Thanks in advance,
RODOLFO

I don't know the (practical) difference, but this will help others
answer your question:
RDSI (portuguese, and as you are from es i believe in spanish too :))
= ISDN (english)
Marconi.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
Undocumented Features quote of the moment...
"It's not the one bullet with your name on it that you
have to worry about; it's the twenty thousand-odd rounds
labeled `occupant.'"
  --Murphy's Laws of Combat
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
---
avast! Antivirus: Inbound message clean.
Virus Database (VPS): 0439-0, 20/09/2004
Tested on: 21/09/2004 18:09:28
avast! - copyright (c) 2000-2004 ALWIL Software.
http://www.avast.com



---
avast! Antivirus: Outbound message clean.
Virus Database (VPS): 0439-0, 20/09/2004
Tested on: 21/09/2004 18:21:23
avast! - copyright (c) 2000-2004 ALWIL Software.
http://www.avast.com

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] chan_sccp/SEP.cnf.xml

[Jesse Tyler]

Thanks Matt:
(damn cisco) :) == > is right!!
I have already compiled the chan_sccp module. It is working just fine. 
My main issue is actually configuring/loading the software the 7914 and 
then using it like a main switchboard.

Thanks Again,
Jesse Tyler
On 21-Sep-04, at 9:53 AM, Matthew Boehm wrote:
If you are going to use the 7914 (which yes, unfortunatly isn't 
supported on
SIP, dammit Cisco) you might want to check out

http://chan-sccp.sourceforge.net
an alternative sccp module for *. Before we switched all our 7960's to 
SIP
we used this and it seemed alot better than the built in one.

Matthew
- Original Message -
From: "Jesse Tyler" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 21, 2004 10:23 AM
Subject: [Asterisk-Users] chan_sccp/SEP.cnf.xml

HI all:
I have spent a large amount of time configuring/installing phones
connected to Asterisk. Halfway through the process I discovered that 
my
Cisco7960 with 2 7914 expansions was not supported in the SIP 
protocol.
After reverting to SCCP 6.0(4.0) I am now perplexed with the hassle of
configuring SCCP to properly work with Asterisk.

So far I have gotten the phone to dial and receive calls from the 
other
participating SIP 7.2 phones on the LAN. Obviously this verifies that
chan_sccp is correctly installed and running. I have scoured the
internet looking for documentation regarding the 7914 call modules 
with
asterisk and turned up nothing.

So this is my question
Does anyone have any kind of documentation regarding 7960 phones with
7914 expansion modules running on AsteriskPBX???
I also have the proper load file for the expansion module...
Any Help would be greatly appreciated,
Thanks,
Jesse Tyler
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] chan_sccp/SEP.cnf.xml

[Craig Guy]

Hi Jesse,

I would strongly recommend changing over to the SIP image and uisng
something like the Flash Operators Panel (www.asternic.org) instead of the
7914's.  I experimented with chan_sccp2 a few weeks ago and decided that it
wasn't for me right now due to both the very limited support for the 7914
and the lack of robustness of the protocol module.  For example configure a
speed dial on the 7960 and place a call, then whilst in the call, press the
speed dial on the 7960 - Asterisk will die.  For a second example take the
handset offhook and leave it offhook for a minute or so without dialling and
you will find that Asterisk will die.

I am hoping to experiment again with the sccp module once it gets a bit more
development time under its belt.

I found the hardest part of configuring sccp was creating the
SEP.cnf.xml file and after much searching I found one which has been
added to the wiki at
http://www.voip-info.org/tiki-index.php?page=SCCP-HOWTO2

Craig

- Original Message - 
From: "Jesse Tyler" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, September 21, 2004 11:23 PM
Subject: [Asterisk-Users] chan_sccp/SEP.cnf.xml


> HI all:
>
> I have spent a large amount of time configuring/installing phones
> connected to Asterisk. Halfway through the process I discovered that my
> Cisco7960 with 2 7914 expansions was not supported in the SIP protocol.
> After reverting to SCCP 6.0(4.0) I am now perplexed with the hassle of
> configuring SCCP to properly work with Asterisk.
>
> So far I have gotten the phone to dial and receive calls from the other
> participating SIP 7.2 phones on the LAN. Obviously this verifies that
> chan_sccp is correctly installed and running. I have scoured the
> internet looking for documentation regarding the 7914 call modules with
> asterisk and turned up nothing.
>
> So this is my question
>
> Does anyone have any kind of documentation regarding 7960 phones with
> 7914 expansion modules running on AsteriskPBX???
>
> I also have the proper load file for the expansion module...
>
>
> Any Help would be greatly appreciated,
>
> Thanks,
>
>
> Jesse Tyler
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] cvs stable

[Andrew Thompson]

Michael Bielicki wrote:
Stable seized to exist quite some time ago.
To expand on  Michael's answer, stable wasn't being kept up to date like 
it should have been, so the statement "get the latest stable version" 
became "get the latest cvs version" as the standard answer for resolving 
people's issues/bugs in asterisk.

Without enough effort being directed at stable, it was not really worth 
calling stable, so it was dropped.

Someone correct me if I'm wrong, but as of this writing, there are no 
specific revisions that you can point to with known this works/this 
doesn't work lists for asterisk. You can pull cvs as of right now, or 
from any point in time, but that's about it.

As I understand it, there is a push in progress for a true 1.0 release.
Hopefully, it will be properly maintained for bugfix'es by digium, or 
the best answer might be copying the release, whenever it happens, to 
someone else's cvs and having the developer community that's interested 
in doing it fix the bugs there.

--
Andrew Thompson
http://aktzero.com/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Anti Ex-Girlfriend feature for entire area codes?

[Nate Carlson]

Hey all,
Someone's posted one of my 800#'s on a poster in California for free 
concert tickets, so I'm getting calls from California AC's at all times of 
the day asking for tickets. I'm just using the 800# for friends and 
family, and don't know anyone in these area codes, so I'd like to just 
give these callers either congestion or a prerecorded message.

Works fine if I do:
exten => 8005551212/4085551212,1,Congestion
exten => 8005551212/4085551212,2,Hangup()
But if I try:
exten => 8005551212/408XXX,1,Congestion
exten => 8005551212/408XXX,2,Hangup()
It doesn't catch it. Is there any way to do something similar and allow 
wildcards? Thanks!


| nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com |
|   depriving some poor village of its idiot since 1981|

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] chan_sccp/SEP.cnf.xml

[Jesse Tyler]

Craig:
Thanks very much for the pointer. I suppose a guy could use dual 
monitors on a Reception PC running Flash Operators Panel. This would 
work well for my application.

Thanks Very much for the info.
(SCCP sucks!)
Jesse Tyler
On 21-Sep-04, at 11:00 AM, Craig Guy wrote:

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Anti Ex-Girlfriend feature for entire area codes?

[Andrew Thompson]

Nate Carlson wrote:
But if I try:
exten => 8005551212/408XXX,1,Congestion
exten => 8005551212/408XXX,2,Hangup()
It doesn't catch it. Is there any way to do something similar and allow 
wildcards? Thanks!

See: 
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf

Look a few blocks into the examples section, you need an underscore in 
there.

When I saw this message, I realized that I goofed in my example for the 
.call file earlier today.

--
Andrew Thompson
http://aktzero.com/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Anti Ex-Girlfriend feature for entire area codes?

[Rob Fugina]

On Tue, 21 Sep 2004 12:05:29 -0500 (CDT), Nate Carlson
<[EMAIL PROTECTED]> wrote:
> Hey all,
> 
> Someone's posted one of my 800#'s on a poster in California for free
> concert tickets, so I'm getting calls from California AC's at all times of
> the day asking for tickets. I'm just using the 800# for friends and
> family, and don't know anyone in these area codes, so I'd like to just
> give these callers either congestion or a prerecorded message.
> 
> Works fine if I do:
> 
> exten => 8005551212/4085551212,1,Congestion
> exten => 8005551212/4085551212,2,Hangup()
> 
> But if I try:
> 
> exten => 8005551212/408XXX,1,Congestion
> exten => 8005551212/408XXX,2,Hangup()
> 
> It doesn't catch it. Is there any way to do something similar and allow
> wildcards? Thanks!

A pattern should start with an underscore, right?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Sanity Check --Zapras With T-1

[John Millican]

Hello All,
I am planning on setting up an * server for a customer and was hoping to get
a sanity check on my Plan.  What I am trying to accomplish is a * voice and
16 data channel T-1 connection (ESF/B8ZS). I am planning on using a 2.8 ghz
P4, 1gig ram, on an Abit AS* Mobo, probably 3Com 10/100nic, T100p, 2 X
TDm4xxp(4 ports FXS).  Does this sound like a reasonable configuration.
Call levels will be light with generally not more that two or three calls at
any given time.  I could talk customer into using only 1 TDM4xxp since he
does not need all 8 FXS. Knowing him he will first want all 8 but...   I am
then going to pull 16 (1024kbps) channel off of T-1 for data through ZapRas.
Am I crazy or will this work.  From what i have read it should work very
well at least if i go with the X100p and one TDM card, not sure about 2 TDM
and X100p in the same machine.

Thank you in advance for any help.
John Millican
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.764 / Virus Database: 511 - Release Date: 9/15/2004

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Polycom IP500 problem updating bootrom

[Matthew Marlowe]

I've had an IP300 for a while now and it's been working fine.  I just
got an IP500 and when it connects to the FTP server it downloads the new
bootrom and says error loading.
 
The bootrom is fine and works on the 300... In addition, I downloaded a
new copy to be sure and it still doesn't work.
 
Can anyone give me some advice?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Zyxel P2000W or WiSIP with asterisk?

[Philip Jander]

Hi,
I'm trying to get a Zyxel P2000W (reportedly also sold as WiSIP by Pulver) 
to work with an asterisk box.
The phone connects nicely to an external VoIP company (sipgate.de 
reportedly using asterisk themselves) but there is a strange problem with 
my asterisk:

- Incoming calls via ISDN (chan_capi) to the Zyxel work perfectly.
- Outgoing calls (capi/sip/iax) via asterisk are dialling out but the Zyxel 
never notes that the other end picked up the call. It just keeps playing a 
ringtone. Of course, no sound is going either way in that case. Quitting 
the call on the Zyxel does hang up the outgoing line, hanging up on the 
other end does not impress the zyxel at all, it just keeps ringing...
- Outgoing calls to the voicemail system do not forward sound (but there is 
no ringtone audible)
- Incoming calls to the zyxel via xlite and iaxphone do not make it due to 
"400 Bad Request". This might be a secondary problem not related to the 
original but it is still strange.
- While trying to make an outgoing call, the zyxel repeatedly sends CANCEL 
messages to asterisk which is strange imho.

And of course:
- there is no NAT involved. Just a standard LAN.
- I have tried about any possible combination of settings for the Zyxel and 
sip.conf, including those listed an voip-info.org
- Everything works nicely with a sip softphone (xlite)
- Judging from google, there seem to be other people with the same problem, 
no solution found.
- I have tried firmwares 00.0e and 00.0f (latest) and asterisk RC1, RC2 and 
current CVS. Same problems all the time.


Since there seem to be a number of people actually using this phone with 
asterisk, I would be really happy about any working config examples 
(including asterisk + firmware versions).

If anybody has actually encountered and solved this problem, please respond :)
If anyone would like to have a look at the sip debug, pls pm me, I will be 
happy to provide those ;)

Thanks,
Phil
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users