RE: [Asterisk-Users] Asterisk on Radio Tonight

[Kerry Garrison]

Thanks! I didn’t like their spin on trying to make it for home users as much
as they did, but oh well, I did what I could.
-Kerry
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Saturday, April 30, 2005 9:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight

You can grab Kerry's Radio Interview from my website

http://www.intruder.com.mx/kerry.mp3

Hope this helps. 

|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of Jim 
|Sturtevant
|Sent: Sábado, 30 de Abril de 2005 10:53 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight
|
|Kerry, can you put an archive of the audio up on your web site or do we 
|need to record the whole 3hr show.  Also, the schedule on their web 
|site shows 5am EST and other repeats.
|I'd love to hear the program.
|
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kerry Garrison
|Sent: Saturday, April 30, 2005 2:28 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] Asterisk on Radio Tonight
|
|Kerry Garrison from The Geek Gazette (http://geekgazette.com)  
|will be interviewed tonight on Mick Mick Williams' Cyber Line 
|radio program at 9:00PM PST. The show is broadcast on the USA 
|Radio network. If you do not have a channel in your area, you 
|can listen listen live online 
|http://www.usaradio.com/listen_live.htm. The show will cover 
|the basic of what the Asterisk PBX is all about and what it 
|takes to implement a system.
|
|-Kerry
|
|
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Re: [Asterisk-Users] Send DTMF *AFTER* channels are bridged

[Peter Svensson]

On Sun, 1 May 2005, Matt Riddell wrote:

  Someone to know how can I send a DTMF after the channels are bridged?
  I need something like the D option of the Dial application, but this
  option sends the DTMF before the channels are bridged. In fact I want the
  caller and the callee to receive the DTMF. Please help :)
 
 If using a codec with inband DTMF, you could always use the option to 
 play an audio file once connected, and just put the DTMF in there.

I think the Dial application was modified recently to allow dtmf to be 
sent to both the caller and callee.

   D([called][:calling])'  -- Send DTMF strings *after* called party 
has answered, but before the call gets bridged. The 'called' 
DTMF string is sent to the called party, and the 'calling' 
DTMF string is sent to the calling party. Both parameters 
can be used alone.\n


Peter


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[Asterisk-Users] Broadvoice limits???

[Tim Connolly]

 Where
are the limitations on the Broadvoice service? I saw a mention on the list
saying two inbound/outbound calls, and Kerry just mentioned it during the radio
interview I dont see that 2 call limitation with my BYOD World
Plus account. Am I lucky, or just missing where the limitation is?






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RE: [Asterisk-Users] Broadvoice limits???

[Kerry Garrison]

There is no physical limitation that I am aware of right 
now. Be sure and check you end user agreement but I think its pretty vauge. They 
told me once "its one call per account" but when I mentioned call waiting they 
said "ok well two calls". We have actually tested it with four to see if there 
was a limit.



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Tim 
ConnollySent: Saturday, April 30, 2005 11:09 PMTo: 
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
[Asterisk-Users] Broadvoice limits???


 
Where are the limitations on the Broadvoice service? I saw a mention on the list 
saying two inbound/outbound calls, and Kerry just mentioned it during the radio 
interview I dont see that 2 call limitation with my BYOD World Plus account. 
Am I lucky, or just missing where the limitation 
is?
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RE: [Asterisk-Users] Asterisk on Radio Tonight

[Anton Krall]

True! I was asking on the irc channel to talk more about Asterisk vs. Cisco
and Avaya solutions... But like you said.. Well... What can We do. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kerry Garrison
|Sent: Domingo, 01 de Mayo de 2005 01:04 a.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight
|
|Thanks! I didn’t like their spin on trying to make it for home 
|users as much as they did, but oh well, I did what I could.
|-Kerry
| 
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Anton Krall
|Sent: Saturday, April 30, 2005 9:58 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight
|
|You can grab Kerry's Radio Interview from my website
|
|http://www.intruder.com.mx/kerry.mp3
|
|Hope this helps. 
|
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Jim 
||Sturtevant
||Sent: Sábado, 30 de Abril de 2005 10:53 p.m.
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: RE: [Asterisk-Users] Asterisk on Radio Tonight
||
||Kerry, can you put an archive of the audio up on your web 
|site or do we 
||need to record the whole 3hr show.  Also, the schedule on their web 
||site shows 5am EST and other repeats.
||I'd love to hear the program.
||
||
||
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Kerry 
||Garrison
||Sent: Saturday, April 30, 2005 2:28 PM
||To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
||Subject: [Asterisk-Users] Asterisk on Radio Tonight
||
||Kerry Garrison from The Geek Gazette (http://geekgazette.com) will be 
||interviewed tonight on Mick Mick Williams' Cyber Line radio 
|program at 
||9:00PM PST. The show is broadcast on the USA Radio network. If you do 
||not have a channel in your area, you can listen listen live online 
||http://www.usaradio.com/listen_live.htm. The show will cover the 
||basic of what the Asterisk PBX is all about and what it takes to 
||implement a system.
||
||-Kerry
||
||
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||
|
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Re: [Asterisk-Users] T1 Technology and VoIP Gateway Primer

[Callum McGillivray]

Hi Adam,

Unfortunatley we are located in Australia and our chosen provider does
not provide this service.

In the future as our client bae grows larger, we may need to look at
implmenting other carriers that provide this kind of service, but in
the meantime we will be using PRI's.

Cheers,

Callum

Adam Robins wrote:

  Why would you use gateways and PRI's when several of the major carriers
(ATT, Global Crossing, etc.) also have products that can interface
directly with SIP for the same per minute cost?

We have a multisite Asterisk call center application and are routing all
calls over private VPN to one central Asterisk location from where we
have multiple point-to-point T1's going straight into Global Crossing.
They are accepting the traffic as SIP g.729a and are handling the
gateway themselves.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Callum
McGillivray
Sent: Friday, April 29, 2005 1:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] T1 Technology and VoIP Gateway Primer

Hi Matt  everyone else,

We have also been steering toward using a gateway for our large
installation.

Ours differs from your significantly in as much as our setup will
involve 8 apartment buildings located throughout the CBD.  Each
apartment building will have as many as 600 extensions (rooms) with an
Asterisk Server in the comms room in the basement.

Incoming and Outgoing calls are going to be trunked from the Asterisk
box along a fiber link back to our core exchange, where the calls will
be handed off to a gateway machine (Cisco?) which will have an
impressively large number of PRI's plugged into the back of it.

My (very vague) examination so far tells me that I can use something
along the lines of a Cisco AS5400 (a couple of which I have kicking
around here in the office).

Has anyone had experience in handing off / receiving calls from a Cisco
AS5400 with Asterisk ? 

How is it done ?

Matt, is this similar to the idea that you have for your project ?  What
Cisco hardware have you looked at so far ?  How many E1/T1 lines are you
going to have terminating on your setup ?

Cheers,

Callum

Matt Roth wrote:

  
  
Michael,



  Have you decided which PSTN-VoIP gateway you'll use?
  



Not yet, but our preference is a Cisco gateway.  Lucent, Quintum, and 
AudioCodes also make TDM-VoIP gateways.

Prior to purchasing any hardware, our entire layout will be posted to 
this list in detail for review.

Matthew Roth
http://voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Deb
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RE: [Asterisk-Users] Broadvoice limits???

[trixter http://www.0xdecafbad.com]

On Sat, 2005-04-30 at 23:19 -0700, Kerry Garrison wrote:
 There is no physical limitation that I am aware of right now. Be sure
 and check you end user agreement but I think its pretty vauge. They
 told me once its one call per account but when I mentioned call
 waiting they said ok well two calls. We have actually tested it with
 four to see if there was a limit.
  


pay particular attention to section 1.3.1 with broadvoice.  I however
accidentally looped my dialplan - had stanaphone (free NY number) used
the whole number including the '1' as its extension.  Set up the
broadvoice dialplan (which what is on broadvoices page seems
imncomplete, never checked to see what else is unlimited but missing but
found one country not there) which defaults to 1NX to go through
broadvoice.  Called my stanaphone number to test, when the call came it
it dialed my stanaphone number. 

Stanaphone didnt like this (and shut my free account off, I got a new
onewith a better number so oh well : but broadovice did not care.  I had
20-30 ports in use at the same time.

One thing broadvoice said in a news article a while back was that they
charge 3.9 cents/min for people who make calls at the same time,
excluding 3-way.  When I did the 30 call loop they didnt charge me
extra, I *think* this only applies if you have multiple IPs register and
place calls at the same time (they do allow multiple IPs to register
according to that same article and all will ring when a call comes in or
something).

I dont have a cite for that article I found it on google, I think it was
a voxilla article though, but it was a while back, and since it was not
an article from today they may have changed their policies onthe 3.9
cents/min.



-- 
Trixter http://www.0xdecafbad.com
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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RE: [Asterisk-Users] Broadvoice limits???

[Tim Connolly]

Broadvoice

Seems to be no limit on inbound, but I
found any channels after 5 outbounds would get an immediate disco. Guess Ill
have to stick to Vonage to blast into the local radio shows. Or maybe 5
on BV, 5 on Vonage, and X on the PRI













From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison
Sent: Sunday, May 01, 2005 1:20 AM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users]
Broadvoice limits???





There is no physical limitation that I am
aware of right now. Be sure and check you end user agreement but I think its
pretty vauge. They told me once its one call per account but when I
mentioned call waiting they said ok well two calls. We have
actually tested it with four to see if there was a limit.











From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Tim Connolly
Sent: Saturday, April 30, 2005
11:09 PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: [Asterisk-Users]
Broadvoice limits???


Where are the limitations on the Broadvoice service? I saw a mention on the
list saying two inbound/outbound calls, and Kerry just mentioned it during the
radio interview I dont see that 2 call limitation with my BYOD
World Plus account. Am I lucky, or just missing where the limitation is?






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RE: [Asterisk-Users] Broadvoice limits???

[Tim Connolly]

I would know if they real-time charged for that... Although I normally only
have 10-12 calls going, I watch pretty close and dispute any
supposed-to-be-free-but-not calls!

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of trixter
http://www.0xdecafbad.com
Sent: Sunday, May 01, 2005 1:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Broadvoice limits???

On Sat, 2005-04-30 at 23:19 -0700, Kerry Garrison wrote:
 There is no physical limitation that I am aware of right now. Be sure
 and check you end user agreement but I think its pretty vauge. They
 told me once its one call per account but when I mentioned call
 waiting they said ok well two calls. We have actually tested it with
 four to see if there was a limit.
  


pay particular attention to section 1.3.1 with broadvoice.  I however
accidentally looped my dialplan - had stanaphone (free NY number) used
the whole number including the '1' as its extension.  Set up the
broadvoice dialplan (which what is on broadvoices page seems
imncomplete, never checked to see what else is unlimited but missing but
found one country not there) which defaults to 1NX to go through
broadvoice.  Called my stanaphone number to test, when the call came it
it dialed my stanaphone number. 

Stanaphone didnt like this (and shut my free account off, I got a new
onewith a better number so oh well : but broadovice did not care.  I had
20-30 ports in use at the same time.

One thing broadvoice said in a news article a while back was that they
charge 3.9 cents/min for people who make calls at the same time,
excluding 3-way.  When I did the 30 call loop they didnt charge me
extra, I *think* this only applies if you have multiple IPs register and
place calls at the same time (they do allow multiple IPs to register
according to that same article and all will ring when a call comes in or
something).

I dont have a cite for that article I found it on google, I think it was
a voxilla article though, but it was a while back, and since it was not
an article from today they may have changed their policies onthe 3.9
cents/min.



-- 
Trixter http://www.0xdecafbad.com
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378

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[Asterisk-Users] Sip calling errors

[iMRAN]

Hi Pros,

I`m new to Asterisk Getting following errors on my * :

   -- Executing Dial(SIP/1000-ee7c, SIP/[EMAIL PROTECTED]) in new stack
   -- Called [EMAIL PROTECTED]
Apr 28 21:06:09 WARNING[2268]: channel.c:2115
ast_channel_make_compatible: No path to translate from
SIP/venus-e8ba(2) to SIP/1000-ee7c(256)
Apr 28 21:06:09 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
Apr 28 21:06:09 NOTICE[2268]: channel.c:1724 ast_set_read_format:
Unable to find a path from g729 to gsm
Apr 28 21:06:09 NOTICE[2268]: channel.c:1691 ast_set_write_format:
Unable to find a path from gsm to g729
   -- SIP/venus-e8ba is making progress passing it to SIP/1000-ee7c
RFC3389: 1 bytes, level 256...
Apr 28 21:06:10 NOTICE[2268]: rtp.c:298 process_rfc3389: RFC3389
support incomplete.  Turn off on client if possible
Apr 28 21:06:13 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
   -- SIP/venus-e8ba answered SIP/1000-ee7c
   -- Attempting native bridge of SIP/1000-ee7c and SIP/venus-e8ba
Apr 28 21:06:21 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)
Apr 28 21:06:22 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)onse)


My SIP.CONF

[general]
port = 5060
bindaddr = 0.0.0.0
canreinvite=no
disallow=all
allow=g723.1
allow=alaw
allow=ulaw
allow=gsm
allow=g729

[venus]
type=friend
context=sip-dial
host=2.2.2.2
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=g729
insecure=very
dtmfmode=info
tos=0x18

[]
type=friend
host=dynamic
username=
secret=imran
dtmf=inband
context=internal
dtmfmode=rfc2833

[1000]
type=friend
username=1000
;secret=password1
host=dynamic
allow=g729
allow=g723.1
context=internal
dtmfmode=rfc2833
=

[general]
static=yes
writeprotect=yes

[globals]
PHONE1=SIP/
PHONE2=SIP/1000
PHONE3=SIP/1001

[internal]
include = local-sip

[local-sip]
exten = ,1,Dial(${PHONE1},40,t)
exten = ,2,Hangup

exten = 1000,1,Dial(${PHONE2},40,t)
exten = 1000,2,Hangup

exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup

exten = _00.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _00.,2,Hangup

Venus is my SIP provider (sorry u might hav guessed already)

1000 and 1001 belongs to my AudioCodec MP108 8 FXS SIP device and 
is my softphone SJphone, i can dial soft to hard and vise versa, i can
call to US number thru my SIP provider using my Sjphone (crapy sound)
but when i try to dial from MP108 i get the above errors i mentioned.

MP108 have preloaded codec i.e. g729 and g723.1, my provider supports
g729 and g723.1

please can anyone help me ?
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[Asterisk-Users] Fwd: Sip calling errors

[iMRAN]

-- Forwarded message --
From: iMRAN [EMAIL PROTECTED]
Date: May 1, 2005 12:16 PM
Subject: Sip calling errors
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com, Alexander Scheerschmidt
[EMAIL PROTECTED]


Hi Pros,

I`m new to Asterisk Getting following errors on my * :

  -- Executing Dial(SIP/1000-ee7c, SIP/[EMAIL PROTECTED]) in new stack
  -- Called [EMAIL PROTECTED]
Apr 28 21:06:09 WARNING[2268]: channel.c:2115
ast_channel_make_compatible: No path to translate from
SIP/venus-e8ba(2) to SIP/1000-ee7c(256)
Apr 28 21:06:09 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
Apr 28 21:06:09 NOTICE[2268]: channel.c:1724 ast_set_read_format:
Unable to find a path from g729 to gsm
Apr 28 21:06:09 NOTICE[2268]: channel.c:1691 ast_set_write_format:
Unable to find a path from gsm to g729
  -- SIP/venus-e8ba is making progress passing it to SIP/1000-ee7c
RFC3389: 1 bytes, level 256...
Apr 28 21:06:10 NOTICE[2268]: rtp.c:298 process_rfc3389: RFC3389
support incomplete.  Turn off on client if possible
Apr 28 21:06:13 NOTICE[2268]: pbx.c:1318 pbx_extension_helper: Cannot
find extension context 'default'
  -- SIP/venus-e8ba answered SIP/1000-ee7c
  -- Attempting native bridge of SIP/1000-ee7c and SIP/venus-e8ba
Apr 28 21:06:21 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)
Apr 28 21:06:22 WARNING[2268]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 25090 (Non-critical Response)onse)


My SIP.CONF

[general]
port = 5060
bindaddr = 0.0.0.0
canreinvite=no
disallow=all
allow=g723.1
allow=alaw
allow=ulaw
allow=gsm
allow=g729

[venus]
type=friend
context=sip-dial
host=2.2.2.2
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=g729
insecure=very
dtmfmode=info
tos=0x18

[]
type=friend
host=dynamic
username=
secret=imran
dtmf=inband
context=internal
dtmfmode=rfc2833

[1000]
type=friend
username=1000
;secret=password1
host=dynamic
allow=g729
allow=g723.1
context=internal
dtmfmode=rfc2833
=

[general]
static=yes
writeprotect=yes

[globals]
PHONE1=SIP/
PHONE2=SIP/1000
PHONE3=SIP/1001

[internal]
include = local-sip

[local-sip]
exten = ,1,Dial(${PHONE1},40,t)
exten = ,2,Hangup

exten = 1000,1,Dial(${PHONE2},40,t)
exten = 1000,2,Hangup

exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup

exten = _00.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _00.,2,Hangup

Venus is my SIP provider (sorry u might hav guessed already)

1000 and 1001 belongs to my AudioCodec MP108 8 FXS SIP device and 
is my softphone SJphone, i can dial soft to hard and vise versa, i can
call to US number thru my SIP provider using my Sjphone (crapy sound)
but when i try to dial from MP108 i get the above errors i mentioned.

MP108 have preloaded codec i.e. g729 and g723.1, my provider supports
g729 and g723.1

please can anyone help me ?
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RE: [Asterisk-Users] Broadvoice limits???

[trixter http://www.0xdecafbad.com]

On Sun, 2005-05-01 at 01:53 -0500, Tim Connolly wrote:
 I would know if they real-time charged for that... Although I normally only
 have 10-12 calls going, I watch pretty close and dispute any
 supposed-to-be-free-but-not calls!

The 20-30 loop was this month, and they havent charged, they all show as
$0.00 on my detail.  As for your 5 outbound limit, I dunno, maybe it was
just that time maybe it was something else because all inbound in that
loop was stanaphone all outbound 1 single broadvoice account.  

I dont intentionally abuse that 'feature' and havent done it since their
upgrades (which it appears every voip provider did this week) so I
dunno.

I believe that its a violation, a tech said (which is not binding
but ...) that you are supposed to have 1 BV account per *person* using
the service, which for a home situation is not relaistic.  The tech did
use the conditional 'technically' when telling me this.  
-- 
Trixter http://www.0xdecafbad.com
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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[Asterisk-Users] Cisco 7960 SIP Reject Call Option

[Asterisk]

Is there anyway of having a Reject Call button appear when there is an 
incoming call. Sometimes I am wating for a call, but one from another 
person comes through - I would like to press a button and send them 
straight to voicemail.

Sort of a Dynamic Do Not Disturb ... :)
Julian
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Re: [Asterisk-Users] Programing a call forward feature to cel phones

[Michiel van Baak]

On 20:28, Sat 30 Apr 05, Anton Krall wrote:
 :)
 No problem dialing another cell phone from asterisk or incoming from cel
 phone, etc.
 
 Console says nothing.
 
 The forwarded call is been directed using zap (x100)
 
 So nothing looks wrong... But still...cant figure out why forwarding the
 call to a cel phone via zap gets those weird sounds after 2 seconds of
 talking and why this happens just when redirecting to a cel phone. Seems
 that if you redirect to a land line is ok.
 
 Also, sometimes, when in a call, any call (cel, land line, etc) sometimes a
 weird sound much like the one I mentioned kicks in the call and I cant get
 the caller because of the sound and he cant listen to me, so I need to hit
 flash and then flash again and the call continues without the sounds...
 Anybody seen that before? Could it be asterisk or the x100? Maybe worth
 mentioning, that I use Monitor to records all calls... Could that be it ? 
 

I saw in another post you have echocancelwhenbridged=off
Did you try to turn this on and see if the problem is gone ?
-- 
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Two of the most famous products of Berkeley are LSD and BSD. I don't think 
that this is a coincidence.

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[Asterisk-Users] mISDN error while compiling

[Sander]

Hi there all!



Does anyone know what this error is???

I am trying to compile the mISDN in kernel 2.6.11.5

I get the same error in kernel 2.6.10.2



Someone?? HELP!!!





WARNING:
/lib/modules/2.6.11.5/kernel/drivers/isdn/hardware/mISDN/hfcmulti.ko needs
unknown symbol pci_find_subsys






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[Asterisk-Users] [Announce] New chan_sccp release adds support for Cisco 7970

[Julien Goodwin]

A new chan_sccp release has just been uploaded which adds support for
the cisco 7970 (min version 6 firmware).

There is one currently known issue with the 7970 support, MWI doesn't
work, and only basic call functions have been tested.

I'd like to publicly thank three people who've helped a lot to get this
release out:
* Jared Mauch for an initial patch
* Adam Megacz for a hardware donation of a 7970 to test on
* Flexion for supplying some ethereal dumps of a 7970 registering to a
callmanager

The next aim for chan_sccp is to support the subscribe/notify features
in asterisk to give BLF type functionality to the 7970 and 7960+7914.

This release has been tested on all the phones I own:
* Cisco 12SP+
* Cisco 30VIP
* Cisco 7905
* Cisco 7940
* Cisco 7970

If I don't have one I can't test it. I'm especially looking for:
* Cisco 7910
* Cisco 7920
* Cisco 7935/36
* And any non-cisco sccp phones (the Kirk IP600 looks very intresting
and if there are any Australian resellers of this here give me a call)

Please see the chan_sccp donations page if you're able to help (Note
that I'm in Australia for those considering hardware donations, but am
willing to pay shipping).
http://chan-sccp.sourceforge.net/donation.html

Thanks,
Julien Goodwin
chan_sccp project lead


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RE: [Asterisk-Users] A good SIP receptionist phone

[Florian Overkamp]

Hi,

Citeren Tim Connolly [EMAIL PROTECTED]:

 Can you show us an example of using the callerID for this purpose?

Simple:

exten = 31531234567,1,SetCIDName(My DIDnr 1)
exten = 31538901234,1,SetCIDName(My DIDnr 2)

exten = _X.,2,Dial(SIP/myphone)


This way, the CallerID number is untouched, but the Name is set to your DID.

Best regards,
Florian
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RE: [Asterisk-Users] IAX2 one way audio

[Trevor G. Hammonds]

Duane Cox wrote on Friday, 29 April 2005 10:17 AM:

 Do you get 2-way audio that sometimes drops off to 1-way audio then
 picks back up as 2-way? (Thats what I see) Not sure if my problem is
 a lost packet issue as I am sending IAX off net.  

My experience has been that there is no two-way audio, except that I am able
to send a SINGLE, brief touch tone digit.  

Let me know if I can provide anything to assist in getting this bug fixed.  

Sincerely,
Trevor Hammonds

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[Asterisk-Users] IPS Version 0.112 released

[Thorben Jensen]

IPSwitchBoard Version 0.112 - 1. may 2005.

Manual for IPSwitchBoard is now included in the download for the new
version, please let me know if you find errors in the manual. 

It's now possible to specify a program to launch along with parameters when
there's an incoming call for IPS. You can also use the Caller ID Number as
parameter.

Download: http://ipswitchboard.thorben.dk


___
IPSwitchBoard is a FREE Windows.Net application that will allow you to: 

Unattended/attended transfers. 
Park calls and retrieve/forward them again. 
Organize all your Zap, SIP and IAX extensions (automatically retrieved from
Asterisk). 
Hotel/Call Shop Billing module
Monitor all extensions. 
Monitor all queues. 
Monitor Agents. 
Monitor Parked Calls. 
Dynamically log extensions in and out of queues. 
Integration with CRM software on the web. 
Record conversations. 
Browse Call Records
Drop any active call. 
Set Do Not Disturb on Extensions and give a reason. 
Speed Dialling. 
User selectable ring tones for IPSwitchBoard. 
User selectable button colors.


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Re: [Asterisk-Users] Problem with PSTN

[Wilson Pickett]

 I want to use this to call on to a Telecom line(PSTN) and vice versa. I read
 somewhere that we need to use some provider for it like FWD or iconnect, do
 we need to use them to make outgoing and incoming calls to PSTN lines or we
 can do it without them.

Try reading these articles:
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html

I don't know what should I put in the .conf files so
 that it enables these calls.

Download and print the available PDFand read it a few times
http://www.asteriskdocs.org

All the answers to your current questions (and more) are there.
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RE: [Asterisk-Users] Problem with Sangoma/Adtran 600 installation

[Chris Mason (Lists)]

 Also, what version of the wanroute driver software are you using?
wanpipe-2.3.2-1

 1. modprobe zaptel
Works.


[EMAIL PROTECTED] asterisk]# service zaptel start
Loading zaptel framework:  [  OK  ]
Waiting for zap to come online...Error: missing /dev/zap!

Chris Mason
www.anguillaguide.com
 

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RE : [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?

[Hakem Taourchi]

Can
this Dell run 90 calls simultaneously? Or need a
higher Dell machine?





-Message
d'origine-
De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Ariel Batista
Envoy: samedi 23
avril 2005 1:27
: 'Ben Hencke';
'Asterisk Users Mailing List - Non-Commercial Discussion'
Objet: RE: [Asterisk-Users]
Dell PowerEdge SC1425 w/ TE405P?



I just setup a SC420 with two TDMO4b cards
in it and it works just fine. No problems what so ever with it so far.











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben Hencke
Sent: Friday, April 22, 2005 6:42
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dell
PowerEdge SC1425 w/ TE405P?





I have head that
the SC prefixed Dells are not good to use with Digium hardware. Considering how
picky my TE405P cards were in other low end Dell servers, I would suggest using
an 1850 instead.

OTOH, if it does work, please let me know :-)
If you go to small biz, you can get the SC1425 trimmed down with dual 2.8hgz
for under $1k
- Ben



On 4/22/05, Greg
Boehnlein [EMAIL PROTECTED]
wrote:

Hello,
I've been asked to build a
couple of Gateway servers for a client
w/ TE405P hardware, and have been looking around at various 1U options.
I've been looking at SuperMicro and Tyan barbones boxes as possible 
platforms, but then was directed to Dell's SC1425 by a friend. Short
story, is that you can purchase a 2x3.0Ghz/1GigDDR400/1xSATA box in a 1U
form factor for $1,498.00. This seems almost too good to be true, so I'm 
asking if anyone has had any experience with this box?

I'm not up on my PCI terminology, but as I understand it, the TE405P can
only be used in a 32 bit 33Mhz slot at 5.0 Volts. This SC1425 lists a
1x 64-bit/1xxMHz PCI-X slot under it's expandability information.
I'd 
venture to guess this is probably NOT going to work with a TE405P.

That being said, if it works, great. If not, what 1U boxes are people
using IN PRODUCTION w/ TE405P cards?

--
Vice President of N2Net, a New Age Consulting Service,
Inc. Company 
 http://www.n2net.net
Where everything clicks into place!

KP-216-121-ST

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Re: [Asterisk-Users] Problem with Sangoma/Adtran 600 installation

[Andrew Kohlsmith]

On May 1, 2005 08:36 am, Chris Mason (Lists) wrote:
 Waiting for zap to come online...Error: missing /dev/zap!

Uh... the error seems obvious.  Have a good look at the various READMEs in the 
zaptel source directory.

-A.
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RE: [Asterisk-Users] Problem with Sangoma/Adtran 600 installation

[Chris Mason (Lists)]

I made some progres on this. The udev issues were causing me problems.
I rectified this and was able to build zaptel successfully.

Wanrouter start works great.

[EMAIL PROTECTED] asterisk]# wanrouter start

Starting WAN Router...
Loading WAN drivers: wanpipe done.
Starting up device: wanpipe1
Configuring interfaces: w1g1 
done.


The next problem comes here

[EMAIL PROTECTED] asterisk]# ztcfg -v

Zaptel Configuration
==

SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)

24 channels configured.

ZT_SPANCONFIG failed on span 1: Invalid argument (22)

Chris Mason
www.anguillaguide.com
 

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RE: [Asterisk-Users] Problem with Sangoma/Adtran 600 installation

[Chris Mason (Lists)]

  x lk x
  x x   Interface Name- w1g1x x
  x x   Operation Mode- TDM_VOICE   x x
  x x   TDM Voice Span- 1  x x
  x x   Override Asterisk Echo Enable - No 

What should the TDM Voice Span be?

Chris Mason
www.anguillaguide.com
 

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[Asterisk-Users] sip based fax client software

[Dean Collins]

I know there is a way to receive faxes via asterisk but is
there any way to send out faxes using a soft client, something that can be
installed on a pc like Winfax that can send out faxes via my asterisk server?



I have a packet 8 ata connected via an x100p card and Id
like to be able to fax out from two pcs on my lan  it seems crazy
to have to buy an fxs card for a fax machine when I already run IAX voice
clients on both pcs





Cheers,

Dean








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[Asterisk-Users] Dutch SIP or IAX numbers

[Asterisk]

How knows where I can get a Dutchphone number for asterisk?

Pilmo is not delivering one for home use.





Thanks



Johannes






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[Asterisk-Users] Playback() stops working.

[Simon Morris]

Hello,

I'm working on configuring asterisk 1.0.7 on Debian Sarge.

The servers been tested a bit and seemed to working fine, but for some
reason now, when I try and run the Playback() or Background()
applications, or even try and goto voicemail asterisk refuses to play
any sounds back to me.

How can I start to debug the cause of this?

Thanks

~sm
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Re: [Asterisk-Users] vmail.cgi: -rwsr-sr-x as root *still* won't read the files

[Tzafrir Cohen]

On Thu, Apr 28, 2005 at 11:43:57PM -0500, Brian Capouch wrote:
 I'm running Apache as nobody.  Wondering why the SUID vmail.cgi script 
 still can't read my files; it comes with the bits set SUID, which I 
 thought would do the trick.
 
 It works just fine if I make the files in the maildir world-readable.
 
 Thanks.  No clues in the archives no Wiki that appear germane.

apache's suexec will not run suid scripts. It will also not run scripts as root.
It has a strict checklist (specified in its docs) that it checks the
target script before exeecuting it. If the script fails one of those
requirements, you'll see a note in suexec's logs.

Linux in general will not run SUID scripts (executables whose magic is 
'#!') as some race conditions will allow you to abuse this to run 
arbitrary command as the target user.

Anyway, asterisk should not be running as root. It should be running 
under its own, separate user. That's what the switch -U is for.
And now you only have to find a way to run that script as that asterisk
user.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] vmail.cgi: -rwsr-sr-x as root *still* won't read the files

[Tzafrir Cohen]

On Fri, Apr 29, 2005 at 10:50:42AM -0400, mike castleman wrote:
 On Fri, Apr 29, 2005 at 12:23:48AM -0500, Brian Capouch wrote:
  
  Drat.  Perl screams bloody murder if you try to just set its SUID bit, 
  which of course is dangerous as hell.
 
 The perl-suid is *not* simply a version of perl with the suid bit set
 but rather a helper binary which allows perl to run suid scripts. Try
 it.

Note that this script does not use any of the standard safety mechanisms
perl provides to achive some safety. It does not use -w or -T or strict.
Nor is it simple to adapt it to use those.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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[Asterisk-Users] TFTP question

[Hermann Wecke]

I'm trying to use TFTP to update the firmware on a Cisco ATA188 but I'm 
receiving this error:

May  1 06:51:50 mail2 in.tftpd[11499]: connect from 192.168.2.2
May  1 06:51:50 mail2 tftpd[11500]: tftpd: trying to get file: 
ata01234567890a
May  1 06:51:50 mail2 tftpd[11500]: tftpd: serving file from /tftpboot
May  1 06:51:50 mail2 in.tftpd[11501]: connect from 192.168.2.2
May  1 06:51:50 mail2 tftpd[11502]: tftpd: trying to get file: 
ata01234567890a
May  1 06:51:50 mail2 tftpd[11502]: tftpd: serving file from /tftpboot
May  1 06:51:50 mail2 in.tftpd[11503]: connect from 192.168.2.2
May  1 06:51:50 mail2 tftpd[11504]: tftpd: trying to get file: 
ata01234567890a
May  1 06:51:50 mail2 tftpd[11504]: tftpd: serving file from /tftpboot
May  1 06:51:55 mail2 tftpd[11500]: tftpd: read: Connection refused
May  1 06:51:55 mail2 tftpd[11502]: tftpd: read: Connection refused
May  1 06:51:55 mail2 tftpd[11504]: tftpd: read: Connection refused

hosts.allow and hosts.deny are empty, directory /tftpboot and files are 
readable by owner/group/others. Running tftpd (0.17-12) on Debian Sarge. 
Similar error message when running atftp.

Ideas?
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Re: [Asterisk-Users] Playback() stops working.

[Eric Wieling aka ManxPower]

I'm working on configuring asterisk 1.0.7 on Debian Sarge.
The servers been tested a bit and seemed to working fine, but for some
reason now, when I try and run the Playback() or Background()
applications, or even try and goto voicemail asterisk refuses to play
any sounds back to me.
I have heard from 5 or so people about this problem.  I run CVS STABLE 
(almost the same as 1.0.7) and had none of these issues.  I wish I 
could help you, but I wanted to let you know that this is not a 
general problem.

--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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Re: [Asterisk-Users] TFTP question

[Derek Whitten]

can you run tftp manually ?

tftp 192.168.2.2

get ata01234567890a


***

from the RTFM in the upgrade package..



sata186us version 3.1

This is a newer manual upgrade server software, also previously known
as upgrade.exe

Upgrade for ATA 186/188

NOTE: This software is to be run from dos command line
 in Windows 9X/ME/NT/2000

Requirements:
	-) network connection between PC and Cisco ATA 186 
	-) PC machine running windows O/S


To upgrade your box, save this executable and the software image
(the software image will have a .zup extension) and language 
image (the language image will have a .kup extension) in a 
directory on your PC. At the DOS prompt of the directory where
the files are saved, you will enter the following:

	sata186us software_file_name -d1 -any2

WARNING -- When upgrading from software version 1.xx to 2.0, make
sure there are entries in the UID0 and UID1 fields, and that you 
hear a dial tone when you pick up the telephone handset. Failure
to do so can result in loss of the MAC address during upgrade.

Your screen will prompt you with instructions on how to upgrade
the box. You will access the voice prompt of the ATA 186 and 
enter the following commands:

	100#ip_address_of_PC*8000#
			(to upgrade the ATA 186's software version)

	101#ip_address_of_PC*8000#
			(to upgrade the ATA 186's language file)

When upgrading many boxes, you can save time by saving the commands
above in your telephone's speed-dial, and using them after accessing 
the ATA 186's voice menu.

Available options: when using this upgrade software:

usage: 

sata186us version 3.1
usage: sata186us {-h[host_ip]} {-p[port]} {-quiet} imageFile
	-h[host_ip] Set host IP to specific IP (in the case where there
	 are more than one IP addresses for the host.
	 Default use 1st IP address obtained by gethostbyname).
	-p[port] Set server port to specific port (default is 8000,
	 use different port only if you are setting up an IP
	 directed upgrade server other than the default).
	-quiet quiet mode, send all output to log file named
	 as [port].log (useful when running the upgrade
	 server as a deamon).
	-any Allow upgrade even if software version is less
	 than or equal to those of client box.
	-any2 Allow upgrade regardless of software type and version.
	-d1,-d2,-d3 Set verbose level for debugging.
	imageFile Image file is file with a '.zup' or '.kup' extension.
e.g.
	sata186us -any -d1 test.zup
	sata186us -h192.168.2.170 -p8002 -quiet test.zup


Below is a sample run of upgrade server in ready mode
PC name: mypc
PC IP: 192.168.1.10

Microsoft(R) Windows Millennium
 (C)Copyright Microsoft Corp 1981-1999.

C:\ata186us ata134-elang.kup
Using Host: mypc with IP : 192.168.1.10 as upgrade server
image found: language 51 -- ata186.itsp2.v1.34

Using dialpad of your telephone (attached to your ATA box),
press ATA button to go to main menu, and enter:

	101#192*168*1*10*8000#	(to upgrade language 51)

NOTE:
Pressing 123# will announce your code's version number.
You can later verify that you have upgraded your ATA box.

---

This program runs continuously; Press ctrl-c to abort.
Upgrade server ready...





from: http://voip-info.org/wiki-Asterisk+phone+cisco+ATA18x

The latest 2.x release SIP/H.323 firmware: 
ftp://ftp.rekom.ru/pub/ata18x/ata18x-v2-16-2-030909a-1.zip
http://kvin.lv/pub/Cisco/ata18x-v2-16-2-030909a-1.zip

3.1(0) firmware: 
http://kvin.lv/pub/Cisco/ata_03_01_00_sip_040211_1.zip



On Sun, 2005-05-01 at 08:49, Hermann Wecke wrote:

I'm trying to use TFTP to update the firmware on a Cisco ATA188 but I'm 
receiving this error:

May  1 06:51:50 mail2 in.tftpd[11499]: connect from 192.168.2.2
May  1 06:51:50 mail2 tftpd[11500]: tftpd: trying to get file: 
ata01234567890a
May  1 06:51:50 mail2 tftpd[11500]: tftpd: serving file from /tftpboot
May  1 06:51:50 mail2 in.tftpd[11501]: connect from 192.168.2.2
May  1 06:51:50 mail2 tftpd[11502]: tftpd: trying to get file: 
ata01234567890a
May  1 06:51:50 mail2 tftpd[11502]: tftpd: serving file from /tftpboot
May  1 06:51:50 mail2 in.tftpd[11503]: connect from 192.168.2.2
May  1 06:51:50 mail2 tftpd[11504]: tftpd: trying to get file: 
ata01234567890a
May  1 06:51:50 mail2 tftpd[11504]: tftpd: serving file from /tftpboot
May  1 06:51:55 mail2 tftpd[11500]: tftpd: read: Connection refused
May  1 06:51:55 mail2 tftpd[11502]: tftpd: read: Connection refused
May  1 06:51:55 mail2 tftpd[11504]: tftpd: read: Connection refused

hosts.allow and hosts.deny are empty, directory /tftpboot and files are 
readable by owner/group/others. Running tftpd (0.17-12) on Debian Sarge. 

Re: [Asterisk-Users] Pattern Matching

[Mojo Jojo]

I do this already with outgoing calls and it works 
fine as long as I am only using the Dial command. 

Where I am running into trouble is when doing 
something like I have created below. I know the syntax is not 100% correct, just 
using it as a quicky example.

What happens here is if the DNIS matches one of the 
first two exact numbers, it plays the background, sets the timeouts then goes on 
and plays thesound in the include and hangs up.

What I want it to do is execute the stuff in the 
include ONLY if none of the exact matches ocurr. I would think this is the way 
it should work but I can't seem to make it happen.


[incoming]
Exten = 
2145550001,1,Answer
Exten = 
2145550001,2,Wait(1)
Exten = 
2145550001,3,Background(MyGreeting)
Exten = 
2145550001,4,Timeout(30) 
Exten = 
2145550001,5,DigitTimeout(3)

Exten = 
2145550002,1,Answer
Exten = 
2145550002,2,Wait(1)
Exten = 
2145550002,3,Background(MyGreeting)
Exten = 
2145550002,4,Timeout(30) 
Exten = 
2145550002,5,DigitTimeout(3)


Include = 
Pattern-Include

[Pattern-Include]

Exten = 
_8XXNXX,1,Answer
Exten = 
_8XXNXX,2,Wait(1)
Exten = 
_8XXNXX,3,Playback(NumNotConfigured)
Exten = 
_8XXNXX,4,Hangup




Private Label Wholesale Internet Access!http://www.YourOwnISP.com

  - Original Message - 
  From: 
  Tim Connolly 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' ; [EMAIL PROTECTED] 
  Sent: Saturday, April 30, 2005 4:21 
  PM
  Subject: RE: [Asterisk-Users] Pattern 
  Matching
  
  
  Like this:
  
  [dids]
  Exten = 
  2145550001,1,dial(SIP/6001)
  Exten = 
  2145550002,1,dial(SIP/6002)
  Exten = 
  2145550003,1,dial(SIP/6003)
  Include = default-did
  
  [default-did]
  Exten = 
  _.,1,dial(SIP/6000)
  
  
  Seems pretty simple. I used this method of 
  least/highest cost routing to choose my LD carrier. Should work the same 
  though.
  
  
  http://www.voip-info.org/tiki-index.php?page=Asterisk%20least%20cost%20routing%20using%20broadvoice
  
  
  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Mojo 
  JojoSent: Saturday, April 30, 2005 3:08 PMTo: [EMAIL PROTECTED]; 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] Pattern Matching
  
  Not sure what you mean exactly... Can you give me a 
  hint?
  
  
  Private Label Wholesale Internet 
  Access!
  http://www.YourOwnISP.com
  
  - Original Message - 
  
  From: "Michael D Schelin" 
  [EMAIL PROTECTED]
  To: "Asterisk Users Mailing List - Non-Commercial 
  Discussion" 
  asterisk-users@lists.digium.com
  Sent: Friday, April 29, 2005 10:10 
  PM
  Subject: Re: [Asterisk-Users] Pattern 
  Matching
  
  
   Hey Mojo, I'm thinking you might try using 
  priorty 's to set some kind 
   routing. just a 
  thought..
  
  
  
   Mojo Jojo wrote:
  
   We recently had our PRI installed, we 
  currently have 100 toll-free's 
   pointing to it.
  
   I have almost everything working great 
  but..
  
   I have setup the first few numbers we want to 
  use coming in from the PRI 
   and they work great, 
  but..
  
   What I want to do is setup an extension with 
  pattern matching to answer 
   for any numbers called that are pointed to 
  our system and PRI but not yet 
   in 
  use/configured.
  
   I have been successful at setting up pattern 
  matching as a catch all for 
   98 or so numbers not in use yet and I have 
  been successful setting up the 
   2 numbers I want to make use of for 
  now.
  
   Problem is, I can't use both at the same 
  time!
  
   If I turn on the pattern matching then my 
  greeting plays for the 
   configured number, then the message plays for 
  the invalid number 
   (basically executing the extension with the 
  pattern matching).
  
   I have read about sorting with pattern 
  matching by using an include, I 
   did this but it's not really 
  helping.
  
   I have set a response timeout after the first 
  extension plays it's 
   greeting, I would think it should wait until 
  it times out but it doesn't, 
   it just immediately moves to the pattern 
  matched extension.
  
   I must be missing something big 
  here..
  
   Any help is 
  appreciated..
  
  
   -- 
   Private Label Wholesale Internet 
  Access!
   
  http://www.YourOwnISP.com
  
   
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Re: [Asterisk-Users] Queues configuration

[Kevin P. Fleming]

Anton Krall wrote:
Weird..
I also have joinwhenempty=no and user can still go into the queue without
any agents logged in.
Are you using queue members (specified in queues.conf or via 
AddQueueMember()), or using agents (specified in agents.conf)? If the 
latter, then the whenempty functions won't work, because as far as 
app_queue is concerned the queue is never empty of members (since it 
only sees the agent definitions, not whether they are logged in or not).
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Re: [Asterisk-Users] Cisco 7960 SIP Reject Call Option

[Kevin P. Fleming]

Asterisk wrote:
Is there anyway of having a Reject Call button appear when there is an 
incoming call. Sometimes I am wating for a call, but one from another 
person comes through - I would like to press a button and send them 
straight to voicemail.
You can press the EndCall button while an unanswered call is ringing 
to achieve the same effect.
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Re: [Asterisk-Users] Cisco 7960 SIP Reject Call Option

[Tom]

At 11:27 AM 5/1/2005, you wrote:
Asterisk wrote:
Is there anyway of having a Reject Call button appear when there is an 
incoming call. Sometimes I am wating for a call, but one from another 
person comes through - I would like to press a button and send them 
straight to voicemail.
You can press the EndCall button while an unanswered call is ringing to 
achieve the same effect.
The only available menu button is Answer when an inbound call is ringing 
on my 7960g.

The menu with EndCall does not come up until I answer the call.
Tom
Sorry for jumping in but I am after the same thing.
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[Asterisk-Users] Asterisk 1.0.6 stable IAX2 Firefly supervised call transfer?

[Paul Redstone]

Hi 

I see various discussions on this but cannot get it to work, and is not clear 
that anyone resolved this. This seems pretty fundamental so I am missing 
something, but I cannot find it anywhere.

# does work for blind transfers - no problem.

But the various * commands given in features.conf do not. OK, I've picked up 
that this may not be in the released one but also I've found that chan_iax2.c 
does talk about attended transfers.

Also iax2 debug shows that the * key is being recognised and passed back.

Can anyone help on this - IAX2 is so much better than SIP which may not have 
this problem.

We're using Firefly phone - neat, simple, seems reliable.

Paul
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Re: [Asterisk-Users] Dutch SIP or IAX numbers

[Michiel van Baak]

On 12:23, Sun 01 May 05, Asterisk wrote:
 How knows where I can get a Dutchphone number for asterisk?
 
 Pilmo is not delivering one for home use.

Three I use:

http://www.speakup.nl (for now only busines accounts)
http://www.talkin2ya.nl (both prepaid and postpaid)
http://www.dutchphone.nl (only postpaid)
-- 
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Two of the most famous products of Berkeley are LSD and BSD. I don't think 
that this is a coincidence.

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[Asterisk-Users] zaptel.conf multiple devices

[Sander]

Hi there my zaptel hardware is giving errors while
loading but they seem to load just fine. the lights wil work and my wctdm card
is also workin and the isdn works to 

But when I stop asterisk I have to reload al cards
again is this normal?



This is my zaptel.conf is there no way to group
these because my te110p is giving an error that it cant find channel 35
but 35 belongs to my wctdm.

Maybe my zaptel.conf is not that good, I cant
find any documentation on multiple cards in one system 

Thanks



ZT_SPANCONFIG failed on span 2: No such device or
address (6)

make: *** [loadlinux26] Error 1

ZT_CHANCONFIG failed on channel 35: No such device or
address (6)

FATAL: Error running install command for wcte11xp

[EMAIL PROTECTED] src]#







loadzone=nl

defaultzone=nl



span=1,1,3,ccs,ami

bchan=1-2

dchan=3





span=1,1,0,ccs,hdb3

bchan=4-18,20-34 # set this to 1-15,17-31 for E1

dchan=19 # set this to 16 for E1





fxoks=35-36






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Re: [Asterisk-Users] Dutch SIP or IAX numbers

[Joris Vandalon]

On Sun, 2005-05-01 at 12:23 -0300, Asterisk wrote:
 How knows where I can get a Dutchphone number for asterisk?

http://www.dis-telecom.nl/v2/dienst.php?id=65



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Re: RE : [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?

[David John Walsh]

what sort of level of PC is required for 300 concurrent calls?

Regards
David

On 5/1/05, Hakem Taourchi [EMAIL PROTECTED] wrote:
  
  
 
 Can this Dell run 90 calls simultaneously ? Or need a higher Dell machine? 
 
   
 
   
 
 -Message d'origine-
  De : [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] De la part
 de Ariel Batista
  Envoyé : samedi 23 avril 2005 1:27
  À : 'Ben Hencke'; 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Objet : RE: [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?
  
 
   
 
 I just setup a SC420 with two TDMO4b cards in it and it works just fine. No
 problems what so ever with it so far. 
 
   
  
  
  
 
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf
 Of Ben Hencke
  Sent: Friday, April 22, 2005 6:42 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P? 
 
   
 
 I have head that the SC prefixed Dells are not good to use with Digium
 hardware. Considering how picky my TE405P cards were in other low end Dell
 servers, I would suggest using an 1850 instead.
  
  OTOH, if it does work, please let me know :-)
  If you go to small biz, you can get the SC1425 trimmed down with dual
 2.8hgz for under $1k
  - Ben 
  
 
 On 4/22/05, Greg Boehnlein [EMAIL PROTECTED] wrote: 
 
 Hello,
  I've been asked to build a couple of Gateway servers for a client
  w/ TE405P hardware, and have been looking around at various 1U options.
  I've been looking at SuperMicro and Tyan barbones boxes as possible 
  platforms, but then was directed to Dell's SC1425 by a friend. Short
  story, is that you can purchase a 2x3.0Ghz/1GigDDR400/1xSATA box in a 1U
  form factor for $1,498.00. This seems almost too good to be true, so I'm 
  asking if anyone has had any experience with this box?
  
  I'm not up on my PCI terminology, but as I understand it, the TE405P can
  only be used in a 32 bit 33Mhz slot at 5.0 Volts. This SC1425 lists a
  1x 64-bit/1xxMHz PCI-X slot under it's expandability information. I'd 
  venture to guess this is probably NOT going to work with a TE405P.
  
  That being said, if it works, great. If not, what 1U boxes are people
  using IN PRODUCTION w/ TE405P cards?
  
  --
  Vice President of N2Net, a New Age Consulting Service, Inc. Company 
   http://www.n2net.net Where everything clicks into place!
   KP-216-121-ST
  
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[Asterisk-Users] TDM400P Power Connector

[Mojo Jojo]

I have a TDM400P I am trying to install but I need a power connector 
extender to be able to get power into the card.

In the meantime can the card run without the power connector if it has only 
one module on it?

Thanks!
Private Label Wholesale Internet Access!
http://www.YourOwnISP.com 

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RE: [Asterisk-Users] Pattern Matching

[Tim Connolly]

Hmm.. The only reason it *should* do that is if it runs out of
priorities on the more significant match, it will then drop back to the next
priority on the next less significant match. Send me your real contexts
offline, maybe were both missing something in the translation to the
list. The incoming extensions are 100% match, right??? Theres
no 9 or 1 prepended on the inbound call? The reason I ask is usually Vonage and
BV send you 1+.. rather than just the 10 digit dnis.











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo Jojo
Sent: Sunday, May 01, 2005 11:21
AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Pattern Matching







I do this already with outgoing calls and it works fine as
long as I am only using the Dial command. 











Where I am running into trouble is when doing something like
I have created below. I know the syntax is not 100% correct, just using it as a
quicky example.











What happens here is if the DNIS matches one of the first
two exact numbers, it plays the background, sets the timeouts then goes on and
plays thesound in the include and hangs up.











What I want it to do is execute the stuff in the include
ONLY if none of the exact matches ocurr. I would think this is the way it
should work but I can't seem to make it happen.











[incoming]

Exten = 2145550001,1,Answer

Exten = 2145550001,2,Wait(1)

Exten = 2145550001,3,Background(MyGreeting)

Exten = 2145550001,4,Timeout(30) 

Exten = 2145550001,5,DigitTimeout(3)



Exten = 2145550002,1,Answer

Exten = 2145550002,2,Wait(1)

Exten = 2145550002,3,Background(MyGreeting)

Exten = 2145550002,4,Timeout(30) 

Exten = 2145550002,5,DigitTimeout(3)





Include = Pattern-Include



[Pattern-Include]



Exten = _8XXNXX,1,Answer

Exten = _8XXNXX,2,Wait(1)

Exten = _8XXNXX,3,Playback(NumNotConfigured)

Exten = _8XXNXX,4,Hangup





















Private Label Wholesale Internet Access!
http://www.YourOwnISP.com







- Original Message - 





From: Tim Connolly 





To: 'Asterisk Users Mailing List -
Non-Commercial Discussion' ; [EMAIL PROTECTED] 





Sent: Saturday, April
30, 2005 4:21 PM





Subject: RE:
[Asterisk-Users] Pattern Matching









Like this:



[dids]

Exten = 2145550001,1,dial(SIP/6001)

Exten = 2145550002,1,dial(SIP/6002)

Exten = 2145550003,1,dial(SIP/6003)

Include = default-did



[default-did]

Exten = _.,1,dial(SIP/6000)





Seems pretty simple. I used this method of least/highest cost routing
to choose my LD carrier. Should work the same though.





http://www.voip-info.org/tiki-index.php?page=Asterisk%20least%20cost%20routing%20using%20broadvoice







-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo Jojo
Sent: Saturday, April 30, 2005 3:08 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Pattern Matching



Not sure what you mean exactly... Can you give me a hint?





Private Label Wholesale Internet Access!

http://www.YourOwnISP.com



- Original Message - 

From: Michael D Schelin [EMAIL PROTECTED]

To: Asterisk Users Mailing List -
 Non-Commercial Discussion 

asterisk-users@lists.digium.com

Sent: Friday, April 29, 2005 10:10 PM

Subject: Re: [Asterisk-Users] Pattern Matching





 Hey Mojo, I'm thinking you might try using priorty 's to set some
kind 

 routing. just a thought..







 Mojo Jojo wrote:



 We recently had our PRI installed, we currently have 100
toll-free's 

 pointing to it.



 I have almost everything working great but..



 I have setup the first few numbers we want to use coming in
from the PRI 

 and they work great, but..



 What I want to do is setup an extension with pattern matching
to answer 

 for any numbers called that are pointed to our system and PRI
but not yet 

 in use/configured.



 I have been successful at setting up pattern matching as a
catch all for 

 98 or so numbers not in use yet and I have been successful
setting up the 

 2 numbers I want to make use of for now.



 Problem is, I can't use both at the same time!



 If I turn on the pattern matching then my greeting plays for
the 

 configured number, then the message plays for the invalid
number 

 (basically executing the extension with the pattern matching).



 I have read about sorting with pattern matching by using an
include, I 

 did this but it's not really helping.



 I have set a response timeout after the first extension plays
it's 

 greeting, I would think it should wait until it times out but
it doesn't, 

 it just immediately moves to the pattern matched extension.



 I must be missing something big here..



 Any help is appreciated..





 -- 

 Private Label Wholesale Internet Access!

 http://www.YourOwnISP.com



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Re: [Asterisk-Users] [Announce] New chan_sccp release adds support forCisco 7970

[Paul A Brown]

Downloaded and did the 'make'
Installed seamlessly...
However my 7920 now keeps coming back saying can't find call manager 0
I get this in the cli
Attempted to check MWI for NULL device
 ==   Got message AlarmMessage
Alarm Message: Severity: 2, 25: Name=SEP000D282E89AA Load=..-(0.0) 
Last=Initialized [2049/1234]
No length in read: Success (errno 0)
 == Sending Packet Type KeepAliveAckMessage (4 bytes)

Any ideas?
Thanks
- Original Message - 
From: Julien Goodwin [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Cc: asterisk-dev@lists.digium.com
Sent: Sunday, May 01, 2005 10:27 AM
Subject: [Asterisk-Users] [Announce] New chan_sccp release adds support 
forCisco 7970


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[Asterisk-Users] 4 - 8 port w/QOS switch for Asterisk

[Joseph]

Can anybody recommend Switch 4 to 8 ports with QOS for Asterisk?

-- 
#Joseph
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[Asterisk-Users] Recording calls

[Jozeph Brasil]

Hi guys,

I need to record all incoming calls. Anyone know how to do this?

Thanks,
Jozeph
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Re: [Asterisk-Users] TDM400P Power Connector

[Kevin P. Fleming]

Mojo Jojo wrote:
In the meantime can the card run without the power connector if it has 
only one module on it?
Power is required to generate ringing voltage for FXS modules; if you 
have only FXO modules, power is not required at all. The number of 
modules is not relevant, as the ringing generator will never draw power 
from the PCI bus.
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[Asterisk-Users] Audio cut off at beginning of call

[Gene Naden]

When we call out from our Asterisk system we consistenly lose the first
roughly 1500 milliseconds of the audio from the destination. This is easiest
to demonstrate with a recorded announcement. In other words, Hello for
example is missing.
We are calling over the PSTN via a voice T1 line.
We are using the stable cvs from about April 1.
I searched lists.digium.com but did not find anyone with this problem
using the PSTN. Does anyone have any ideas?

Gene Naden, MA , MD
Programmer Analyst
GlobalTeldata II, LLC
4700 N. Ravenswood
Chicago, IL 60640
(773) 878-3161 x 223

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Re: RE : [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?

[Greg Boehnlein]

On Sun, 1 May 2005, David John Walsh wrote:

 what sort of level of PC is required for 300 concurrent calls?

Doing what? Ulaw? That could probably be done on a single P4 2.8 Ghz. If 
you want to transcode from Ulaw to something else, you need to scale the 
hardware appropriately. Every case is different.

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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RE: [Asterisk-Users] Queues configuration

[Anton Krall]

I have my agents defined in agents.conf.. Damn.. I normally use
agentcallbacklogin.. So how can I use agentcallbacklogin and addqueuemember?


|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kevin P. Fleming
|Sent: Domingo, 01 de Mayo de 2005 11:24 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Queues configuration
|
|Anton Krall wrote:
| Weird..
| 
| I also have joinwhenempty=no and user can still go into the queue 
| without any agents logged in.
|
|Are you using queue members (specified in queues.conf or via 
|AddQueueMember()), or using agents (specified in agents.conf)? 
|If the latter, then the whenempty functions won't work, 
|because as far as app_queue is concerned the queue is never 
|empty of members (since it only sees the agent definitions, 
|not whether they are logged in or not).
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|

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Re: RE : [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?

[Daniel Salama]

Along the same lines, is there some sort of capacity chart that maps 
capacity based on translations/transcoding?

- Daniel
On May 1, 2005, at 2:45 PM, Greg Boehnlein wrote:
On Sun, 1 May 2005, David John Walsh wrote:
what sort of level of PC is required for 300 concurrent calls?
Doing what? Ulaw? That could probably be done on a single P4 2.8 Ghz. 
If
you want to transcode from Ulaw to something else, you need to scale 
the
hardware appropriately. Every case is different.

--
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST

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RE: [Asterisk-Users] Programing a call forward feature to cel phones

[Anton Krall]

I tried eventhough the call are not been bridge like when Asterisk bridges
2 sip calls and steps out of the way.

I tried using this:

echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=800

No luck either :(

Any ideas? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Michiel van Baak
|Sent: Domingo, 01 de Mayo de 2005 02:52 a.m.
|To: asterisk-users@lists.digium.com
|Subject: Re: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|On 20:28, Sat 30 Apr 05, Anton Krall wrote:
| :)
| No problem dialing another cell phone from asterisk or incoming from 
| cel phone, etc.
| 
| Console says nothing.
| 
| The forwarded call is been directed using zap (x100)
| 
| So nothing looks wrong... But still...cant figure out why forwarding 
| the call to a cel phone via zap gets those weird sounds after 2 
| seconds of talking and why this happens just when 
|redirecting to a cel 
| phone. Seems that if you redirect to a land line is ok.
| 
| Also, sometimes, when in a call, any call (cel, land line, etc) 
| sometimes a weird sound much like the one I mentioned kicks in the 
| call and I cant get the caller because of the sound and he 
|cant listen 
| to me, so I need to hit flash and then flash again and the 
|call continues without the sounds...
| Anybody seen that before? Could it be asterisk or the x100? Maybe 
| worth mentioning, that I use Monitor to records all calls... 
|Could that be it ?
| 
|
|I saw in another post you have echocancelwhenbridged=off Did 
|you try to turn this on and see if the problem is gone ?
|--
|Michiel van Baak
|http://lunteren.vanbaak.info
|[EMAIL PROTECTED]
|GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
|
|Two of the most famous products of Berkeley are LSD and BSD. 
|I don't think that this is a coincidence.
|
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Re: [Asterisk-Users] Intel 536EP

[Gavin Hamill]

On Saturday 30 April 2005 18:09, Jeff wrote:

 Will the Intel 536EP function as a FXO? 

No.

Cheers,
Gavin.
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Re: [Asterisk-Users] Queues configuration

[Kevin P. Fleming]

Anton Krall wrote:
I have my agents defined in agents.conf.. Damn.. I normally use
agentcallbacklogin.. So how can I use agentcallbacklogin and addqueuemember?
AddQueueMember does pretty much the same thing as AgentCallbackLogin, it 
causes the queue to dial the agent when a call is being delivered to 
them. It just doesn't use any of the chan_agent infrastructure, so you'd 
have to build your own agent code/password entry and validation before 
calling AddQueueMember. There is an example on the wiki of doing this.
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[Asterisk-Users] Make Webvmail Error

[Manjit Riat]

I did a make webvmail and I get
the following error on redhat 9.0



No HTTP directory

make: *** [webvmail] Error 1



I have the perl-suidperl rpm
installed and apache installed



Thanx.








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RE: [Asterisk-Users] Programing a call forward feature to cel phones

[Tim Connolly]

Maybe turn echotraining off altogether.. I wonder if the cell company is
also doing some line conditioning that is killing the call quality after the
training (at both ends) stops.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall
Sent: Sunday, May 01, 2005 2:05 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Programing a call forward feature to cel
phones

I tried eventhough the call are not been bridge like when Asterisk bridges
2 sip calls and steps out of the way.

I tried using this:

echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echotraining=800

No luck either :(

Any ideas? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Michiel van Baak
|Sent: Domingo, 01 de Mayo de 2005 02:52 a.m.
|To: asterisk-users@lists.digium.com
|Subject: Re: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|On 20:28, Sat 30 Apr 05, Anton Krall wrote:
| :)
| No problem dialing another cell phone from asterisk or incoming from 
| cel phone, etc.
| 
| Console says nothing.
| 
| The forwarded call is been directed using zap (x100)
| 
| So nothing looks wrong... But still...cant figure out why forwarding 
| the call to a cel phone via zap gets those weird sounds after 2 
| seconds of talking and why this happens just when 
|redirecting to a cel 
| phone. Seems that if you redirect to a land line is ok.
| 
| Also, sometimes, when in a call, any call (cel, land line, etc) 
| sometimes a weird sound much like the one I mentioned kicks in the 
| call and I cant get the caller because of the sound and he 
|cant listen 
| to me, so I need to hit flash and then flash again and the 
|call continues without the sounds...
| Anybody seen that before? Could it be asterisk or the x100? Maybe 
| worth mentioning, that I use Monitor to records all calls... 
|Could that be it ?
| 
|
|I saw in another post you have echocancelwhenbridged=off Did 
|you try to turn this on and see if the problem is gone ?
|--
|Michiel van Baak
|http://lunteren.vanbaak.info
|[EMAIL PROTECTED]
|GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
|
|Two of the most famous products of Berkeley are LSD and BSD. 
|I don't think that this is a coincidence.
|
|___
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|http://lists.digium.com/mailman/listinfo/asterisk-users
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|   http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] Recording calls

[Tim Connolly]

You might check the wiki next time before you ask:
http://www.voip-info.org/wiki-Asterisk+cmd+Monitor
http://www.voip-info.org/wiki-Asterisk+cmd+record

Tim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jozeph Brasil
Sent: Sunday, May 01, 2005 1:19 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Recording calls

Hi guys,

I need to record all incoming calls. Anyone know how to do this?

Thanks,
Jozeph

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[Asterisk-Users] Pre-Parse Extensions.conf?

[Matthew Boehm]

Is there a way to pre-parse your extensions.conf so you can check for errors
before making it live?

ARA Extensions is a really cool tool and will allow us to let our customers
create/manage their own dialplans. It would be nice if when a customer
changes their dialplan that it gets parsed and checked for errors before
being inserted into the database.

-Matthew 


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Re: [Asterisk-Users] Recording calls

[Joseph]

On Sun, 2005-05-01 at 15:18 -0300, Jozeph Brasil wrote:
 Hi guys,
 
 I need to record all incoming calls. Anyone know how to do this?
 
 Thanks,
 Jozeph

Very easy, take a look:

exten = 718,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = 718,2,Monitor(wav,${CALLFILENAME},m)
exten = 718,3,AbsoluteTimeout(15); retun control to T
exten = 718,4,Dial(${phone1},20,rw)

-- 
#Joseph
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[Asterisk-Users] TDM400P does not detect hangup on UK BT analogue line

[Stuart Ford]

Dear Collective ...

I know that this problem crops up again and again, but I've yet to find
something that works for me. I've completely exhausted Google.

I have a TDM400P card with a single FXO module connected to a standard
analogue BT telephone line. The card works fine, there are no IRQ issues or
crackling or echoing or any of that rubbish.

However, the card and/or Asterisk fail to detect when the remote party have
hungup. This isn't a problem if it's a real telephone conversation with a
user of the system as in such cases Asterisk detects the hangup from the
local end and terminates the channel, but if someone calls in and gets to an
un-manned service such as an IVR menu or voicemail and then hangs up, the
channel remains open and you get the 2 minutes of continous tone voicemail
problem.

I have correctly modprobed wcfxs with UK as the opermode (dmesg confirms):

Zapata Telephony Interface Registered on major 196
PCI: setting IRQ 11 as level-triggered
PCI: Assigned IRQ 11 for device :00:08.0
Freshmaker version: 71
Freshmaker passed register test
Module 0: Not installed
Module 1: Not installed
Module 2: Not installed
Module 3: Installed -- AUTO FXO (UK mode)
Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)
Registered tone zone 4 (United Kingdom)

This is my /etc/zaptel.conf:

loadzone=uk
defaultzone=uk
fxsks=4

This is my /etc/asterisk/zapata.conf:

[channels]

language=en
context=default
signalling=fxs_ks

usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
restrictcid=no

callwaiting=no
echocancel=yes
busydetect=yes
callprogress=yes
hanguponpolarityswitch=yes

echocancel=yes
rxgain=4.5
txgain=4.5

immediate=no
context=incoming
channel = 4

All the polarity-based caller-ID stuff works, so I know that the card is
capable of detecting polarity switches, but the (ideal looking)
hanguponpolarityswitch parameter has no effect.

What am I missing? Surely this can't be a bug!

Thanks

Stuart


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RE: [Asterisk-Users] Programing a call forward feature to cel phones

[Anton Krall]

My other question is.. Why does that sound also happen sometimes while in a
call with a pstn number? 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Tim Connolly
|Sent: Domingo, 01 de Mayo de 2005 02:14 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|Maybe turn echotraining off altogether.. I wonder if the cell 
|company is also doing some line conditioning that is killing 
|the call quality after the training (at both ends) stops.
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Anton Krall
|Sent: Sunday, May 01, 2005 2:05 PM
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: RE: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|I tried eventhough the call are not been bridge like when 
|Asterisk bridges
|2 sip calls and steps out of the way.
|
|I tried using this:
|
|echocancel=yes
|echocancelwhenbridged=yes
|echotraining=yes
|echotraining=800
|
|No luck either :(
|
|Any ideas? 
|
||-Original Message-
||From: [EMAIL PROTECTED]
||[mailto:[EMAIL PROTECTED] On Behalf Of Michiel 
||van Baak
||Sent: Domingo, 01 de Mayo de 2005 02:52 a.m.
||To: asterisk-users@lists.digium.com
||Subject: Re: [Asterisk-Users] Programing a call forward 
|feature to cel 
||phones
||
||On 20:28, Sat 30 Apr 05, Anton Krall wrote:
|| :)
|| No problem dialing another cell phone from asterisk or 
|incoming from 
|| cel phone, etc.
|| 
|| Console says nothing.
|| 
|| The forwarded call is been directed using zap (x100)
|| 
|| So nothing looks wrong... But still...cant figure out why 
|forwarding 
|| the call to a cel phone via zap gets those weird sounds after 2 
|| seconds of talking and why this happens just when
||redirecting to a cel
|| phone. Seems that if you redirect to a land line is ok.
|| 
|| Also, sometimes, when in a call, any call (cel, land line, etc) 
|| sometimes a weird sound much like the one I mentioned kicks in the 
|| call and I cant get the caller because of the sound and he
||cant listen
|| to me, so I need to hit flash and then flash again and the
||call continues without the sounds...
|| Anybody seen that before? Could it be asterisk or the x100? Maybe 
|| worth mentioning, that I use Monitor to records all calls...
||Could that be it ?
|| 
||
||I saw in another post you have echocancelwhenbridged=off Did 
|you try to 
||turn this on and see if the problem is gone ?
||--
||Michiel van Baak
||http://lunteren.vanbaak.info
||[EMAIL PROTECTED]
||GnuPG key: 
|http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
||
||Two of the most famous products of Berkeley are LSD and BSD. 
||I don't think that this is a coincidence.
||
||___
||Asterisk-Users mailing list
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||http://lists.digium.com/mailman/listinfo/asterisk-users
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||   http://lists.digium.com/mailman/listinfo/asterisk-users
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|
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|

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RE: [Asterisk-Users] Queues configuration

[Anton Krall]

Worth taking a look..thx! 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Kevin P. Fleming
|Sent: Domingo, 01 de Mayo de 2005 02:08 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Queues configuration
|
|Anton Krall wrote:
| I have my agents defined in agents.conf.. Damn.. I normally use 
| agentcallbacklogin.. So how can I use agentcallbacklogin and 
|addqueuemember?
|
|AddQueueMember does pretty much the same thing as 
|AgentCallbackLogin, it causes the queue to dial the agent when 
|a call is being delivered to them. It just doesn't use any of 
|the chan_agent infrastructure, so you'd have to build your own 
|agent code/password entry and validation before calling 
|AddQueueMember. There is an example on the wiki of doing this.
|___
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|   http://lists.digium.com/mailman/listinfo/asterisk-users
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|

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RE: [Asterisk-Users] 4 - 8 port w/QOS switch for Asterisk

[Max W Blackmer Jr]

Linksys has a low end router with an 8 port switch that does QoS model
BEFSR81. It can be gotten for under $100 USD.  For more information
http://www.linksys.com/products/product.asp?prid=604scid=29

Max W. Blackmer, Jr.

  Original Message 
 Subject: [Asterisk-Users] 4 - 8 port w/QOS switch for Asterisk
 From: Joseph [EMAIL PROTECTED]
 Date: Sun, May 01, 2005 1:05 pm
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com

 Can anybody recommend Switch 4 to 8 ports with QOS for Asterisk?

 --
 #Joseph
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[Asterisk-Users] Problems in new implemenation....

[Stephen Malenshek]

I have recently implemented a SIP VoIP implementation using Asterisk.  I can
go through and place a call to a particular number from the PSTN, the phone
rings, but I am not getting the ring response back to the calling party.  I
am not sure as to where this problem is coming from, but I know it stopped
working once I added the configurations

dial-peer voice 82010151 pots
 incoming called-number 2010151
 direct-inward-dial
 forward-digits all
!
dial-peer voice 2010151 voip
 destination-pattern 2010151
 session protocol sipv2
 session target ipv4:XXX.XXX.XXX.XXX
 session transport udp
 incoming called-number 2010151
 dtmf-relay sip-notify rtp-nte
 codec g711ulaw
!
dial-peer voice 82010152 pots
 incoming called-number 2010152
 direct-inward-dial
 forward-digits all
!
dial-peer voice 2010152 voip
 destination-pattern 2010152
 session protocol sipv2
 session target ipv4: XXX.XXX.XXX.XXX
 session transport udp
 incoming called-number 2010152
 dtmf-relay sip-notify rtp-nte
 codec g711ulaw
!
!
sip-ua
 max-forwards 15
 retry invite 10
 timers trying 1000
 timers expires 30
 sip-server ipv4: XXX.XXX.XXX.XXX
 no transport tcp
!

I also have a Cisco Call Manager Express sending and receiving calls to and
from this same equipment without the problem existing.  I am sure that this
problem is something with the way that I have the SIP commands configured on
this AS5400, but I just do not know enough to fix it.

Thanks for your thoughts.

Stephen

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Re: [Asterisk-Users] Pre-Parse Extensions.conf?

[Kevin P. Fleming]

Matthew Boehm wrote:
Is there a way to pre-parse your extensions.conf so you can check for errors
before making it live?
No, there is not. And any pre-parser that existed today would be 
incomplete anyway, because most of the dialplan is not actually parsed 
until the applications are called (i.e. it's not parsed a load time, but 
at run time).

There have been discussions about improving the dialplan/application 
interfaces in a way that would allow for load-time parsing, but there 
hasn't been any movement in that direction yet. If that does happen, 
you'd at least be able to load your dialplan in a separate testing 
instance of Asterisk to see if it parses properly.
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RE: [Asterisk-Users] Recording calls

[Tim Connolly]

Wouldn't that example kill the call after 15 seconds? I use option 'b' in
Monitor() also. Seems to cut down on recordings where you hear lots of
ringing.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joseph
Sent: Sunday, May 01, 2005 2:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Recording calls

On Sun, 2005-05-01 at 15:18 -0300, Jozeph Brasil wrote:
 Hi guys,
 
 I need to record all incoming calls. Anyone know how to do this?
 
 Thanks,
 Jozeph

Very easy, take a look:

exten = 718,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = 718,2,Monitor(wav,${CALLFILENAME},m)
exten = 718,3,AbsoluteTimeout(15); retun control to T
exten = 718,4,Dial(${phone1},20,rw)

-- 
#Joseph
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RE: [Asterisk-Users] 4 - 8 port w/QOS switch for Asterisk

[Joseph]

Yes, I've seen one and comparing it to: 
The MIL-S8001TG a Layer 2 Gigabit Ethernet Smart switch that features
eight 10/100/1000 Copper ports plus a Mini-GBIC (SFP) interface slot.

I don't need a router, as I'm using Freesco with print-server and port
knocking configuration.  Though Freesco doesn't do QOS.

#Joseph

On Sun, 2005-05-01 at 12:43 -0700, Max W Blackmer Jr wrote:
 Linksys has a low end router with an 8 port switch that does QoS model
 BEFSR81. It can be gotten for under $100 USD.  For more information
 http://www.linksys.com/products/product.asp?prid=604scid=29
 
 Max W. Blackmer, Jr.
 
   Original Message 
  Subject: [Asterisk-Users] 4 - 8 port w/QOS switch for Asterisk
  From: Joseph [EMAIL PROTECTED]
  Date: Sun, May 01, 2005 1:05 pm
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
 
  Can anybody recommend Switch 4 to 8 ports with QOS for Asterisk?
 
  --
  #Joseph
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#Joseph
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[Asterisk-Users] Execute commands after phone hangs up

[Chuck Smith]

Here is a good one.


If I wanted to record a call when it comes into a particular extension, then
e-mail the wav file to a particular email address what would I do? I know
how to initiate the recording but I don't know how to initiate the e-mail
after the call is complete. I can initiate a call before the call is
complete but not after.

Thanks in advance.

Chuck






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Re: [Asterisk-Users] Execute commands after phone hangs up

[Shady]

The easiest way is to use the g parameter of the Dial application. Then in 
the next priority you may use the System application to execute an external 
mail command.

- Original Message - 
From: Chuck Smith [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com
Sent: Monday, May 02, 2005 12:11 AM
Subject: [Asterisk-Users] Execute commands after phone hangs up


Here is a good one.
If I wanted to record a call when it comes into a particular extension, 
then
e-mail the wav file to a particular email address what would I do? I know
how to initiate the recording but I don't know how to initiate the e-mail
after the call is complete. I can initiate a call before the call is
complete but not after.

Thanks in advance.
Chuck


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Re: [Asterisk-Users] Execute commands after phone hangs up

[Michiel van Baak]

On 00:17, Mon 02 May 05, Shady wrote:
 The easiest way is to use the g parameter of the Dial application. Then 
 in the next priority you may use the System application to execute an 
 external mail command.
 
 
 - Original Message - 
 From: Chuck Smith [EMAIL PROTECTED]
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
 asterisk-users@lists.digium.com
 Sent: Monday, May 02, 2005 12:11 AM
 Subject: [Asterisk-Users] Execute commands after phone hangs up
 
 
 Here is a good one.
 
 
 If I wanted to record a call when it comes into a particular extension, 
 then
 e-mail the wav file to a particular email address what would I do? I know
 how to initiate the recording but I don't know how to initiate the e-mail
 after the call is complete. I can initiate a call before the call is
 complete but not after.
 
 Thanks in advance.
 
 Chuck
 

Or you can use exten = h,1,System in that particular
extension
-- 
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Two of the most famous products of Berkeley are LSD and BSD. I don't think 
that this is a coincidence.

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[Asterisk-Users] Sipura 401 Unauthorized.

[Forrest W. Christian]

I'm having ongoing registration fits with some SPA-2000's.

Right now I have one which, based on the debugging output repeatedly fails
with 401 unauthorized:

-
-- SIP read from 206.127.114.240:5060:
REGISTER sip:voip-proxy.mt.net SIP/2.0
Via: SIP/2.0/UDP 206.127.114.240:50291;branch=z9hG4bK-f4e3f6bc
From: MIC Sipura User
sip:[EMAIL PROTECTED];tag=abf0b0fbea0a3e68o0
To: MIC Sipura User sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 52 REGISTER
Max-Forwards: 70
Contact: MIC Sipura User sip:[EMAIL PROTECTED]:50291;expires=60
User-Agent: Sipura/Sipura/SPA2000-2.0.13(g)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


--- (12 headers 0 lines)---
Using latest request as basis request
Sending to 206.127.114.240 : 50291 (NAT)
Transmitting (NAT) to 206.127.114.240:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
206.127.114.240:50291;branch=z9hG4bK-f4e3f6bc;received=206.127.114.240;rport=5060
From: MIC Sipura User
sip:[EMAIL PROTECTED];tag=abf0b0fbea0a3e68o0
To: MIC Sipura User sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 52 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0


---
Transmitting (NAT) to 206.127.114.240:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
206.127.114.240:50291;branch=z9hG4bK-f4e3f6bc;received=206.127.114.240;rport=5060
From: MIC Sipura User
sip:[EMAIL PROTECTED];tag=abf0b0fbea0a3e68o0
To: MIC Sipura User sip:[EMAIL PROTECTED];tag=as3bb6fc64
Call-ID: [EMAIL PROTECTED]
CSeq: 52 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
WWW-Authenticate: Digest realm=asterisk, nonce=36609d12
Content-Length: 0
-



This box will register and then be ok for several days and then will go
into this mode.  I've seen other sipuras doing something similar but with
407.  In fact, calls were made with this box yesterday with absolutely no
configuration changes on either end, until I started to try to figure out
why it isn't registering.

With only 5-6 SPA-2000's in test and several of them acting flaky
registration-wise I'm feeling that I'm missing something which causes this
flakiness.

Both the SPA and asterisk have been rebooted.  Asterisk has actually been
updated to the latest CVS version today in case there was an
already-in-cvs fix.

The config I have in asterisk for this sipura box is:

[A0974L1]
type=friend
host=dynamic
context=cosinternational
secret=REMOVED
callerid=MIC Sipura User 406###
dtmfmode=rfc2833
reinvite=no
canreinvite=no
nat=yes
qualify=no

Ideas?  Other places I should look?

-forrest


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RE: [Asterisk-Users] Web interface Suggestions

[G.Marshall]

 I think you will find AMP is about to implement a multi tenant solution.
But does AMP deal with realtime? or just flat files the data for which is
held in a db?


 Open Source project I assume. I am interested in this project do you
Only open source.

 have a webpage about it?

You can find the current version at
https://www.dalmany.co.uk/asterisk/index.html I am open to suggestions and
requests.  Pages waiting incorporation include voicemail, sip users and
sip peers.

This only deals with Realtime, it does not replicate AMP with a db and
flatfiles.  It does not modify any flatfiles, only the realtime database
so one has to know about realtime and how it works to get the full
benefit.

I am in the throws of moving house which is preventing me from developing
it as quickly as I would like.


 Thanks,
  _
 /-\ ndrew

 On 4/28/05, G.Marshall [EMAIL PROTECTED] wrote:
   Has anyone come across any software that can control
 adding/editing
   SIP extension properties and perhaps dial plan properties on a
 context
   basis. What I mean is I would like it so an admin user from
 Company A
   can manipulate
   properties for extensions in his context but not in another
 Companies.
 I
   know AMP does something similar
   to this but from what I understand it does not allow for different
 users
   at different companies to control
   only things that pertain to them.
  In my spare time, I am developing a php webfrontend to realtime
 asterisk
  database which modifies dialplan, users etc.  Should not be too
 difficult
  to  add a login facility which means the user can see their own
 context
  only.
 
  Regards,
 
  Spencer
  ---
  https://www.dalmany.co.uk/dundi/dundi.php
  https://www.dalmany.co.uk/asterisk/index.php

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[Asterisk-Users] Re: Dutch SIP or IAX numbers

[Wessel de Roode]

 Message: 1
 Date: Sun, 1 May 2005 19:01:24 +0200
 From: Michiel van Baak [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Dutch SIP or IAX numbers
 To: asterisk-users@lists.digium.com
 Message-ID: [EMAIL PROTECTED]
 Content-Type: text/plain; charset=us-ascii
 
 On 12:23, Sun 01 May 05, Asterisk wrote:
  How knows where I can get a Dutchphone number for asterisk?
  
  Pilmo is not delivering one for home use.

www.voipgate.nl

Not using it, but offers IAX2.

Wessel

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.11.0 - Release Date: 29-04-05
 

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[Asterisk-Users] Centos - Hylafax Install

[mr. barker]

Has anyone tried to install Hylafax on Centos ?



If so is there an rpm .. or what was your compiling
procedure ?



Thanks in return






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Re: [Asterisk-Users] Centos - Hylafax Install

[Mike]

Does this sound like the centos mailing list.  Check irc, #centos
On Sun, 1 May 2005, mr. barker wrote:
Has anyone tried to install Hylafax on Centos ?

If so is there an rpm .. or what was your compiling procedure ?

Thanks in return

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[Asterisk-Users] Latest CVS Head Nukes Server

[Rod Bacon]

Has anyone experienced problems with recent CVS HEAD (as of 30th April)
version of * completely crashing the PC on shutdown? (I can't see the
console, because the server is located in remote data centre).

The problem doesn't appear to happen all the time. Only when * has been
running for a while.


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RE: [Asterisk-Users] Polycom IP500 - Phone TIme

[Paul Hales]

It now works - but only in the latest (1.5+) firmware releases.

Later,

PaulH 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Richard Scobie
Sent: Friday, 29 April 2005 6:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom IP500 - Phone TIme



Paul Hales wrote:
 And my dreamthat one day Polycom phones will support Australian Daylight 
 savings...
 
 But it's only a dream.

Unless I am missing something, you don't need to dream about it - set it in 
ipmid.cfg.

Look at the Sip Admim PDF for an explanation of:

tcpIpApp.sntp.daylightSavings.enable=1 
tcpIpApp.sntp.daylightSavings.fixedDayEnable=0 
tcpIpApp.sntp.daylightSavings.start.month=4 
tcpIpApp.sntp.daylightSavings.start.date=1 
tcpIpApp.sntp.daylightSavings.start.time=2 
tcpIpApp.sntp.daylightSavings.start.dayOfWeek=1
tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth=0 
tcpIpApp.sntp.daylightSavings.stop.month=10 
tcpIpApp.sntp.daylightSavings.stop.date=1
tcpIpApp.sntp.daylightSavings.stop.time=2
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek=1
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth=1

Regards,

Richard
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[Asterisk-Users] IAX channels do not disconnect

[Rizwan Chaudhry]

Hey I'm experiencing this strange thing.

My setup is : PSTN = Asterisk =IAX= Asterisk = PSTN

The thing is when I dial from the PSTN to the asterisk server and the
latter has called the PSTN number on the other asterisk server, even
if I hangup both the phnes on the PSTN, the ASterisk servers still are
talking to each other over IAX. The IAX channel is not hanging up when
a PSTN phone hangs up!

Please help

Regards

Riz
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Re: [Asterisk-Users] Pre-Parse Extensions.conf?

[Matthew Boehm]

 Matthew Boehm wrote:
 Is there a way to pre-parse your extensions.conf so you can check for errors
 before making it live?
 
 No, there is not. And any pre-parser that existed today would be
 incomplete anyway, because most of the dialplan is not actually parsed
 until the applications are called (i.e. it's not parsed a load time, but
 at run time).
 
 There have been discussions about improving the dialplan/application
 interfaces in a way that would allow for load-time parsing, but there
 hasn't been any movement in that direction yet. If that does happen,
 you'd at least be able to load your dialplan in a separate testing
 instance of Asterisk to see if it parses properly.

I think I said it wrong. I'm not looking to pre-compile the dialplan;

Looking for more of a syntax checker than anything. Something like that
isn't possible?

It happens when you call extensions reload so I figured the function(s)
that read and parse the extensions.conf could accept an extra variable to
where when the final function that actually loads it into memory is skipped.

Sorta like extensions reload [check] or extensions check|reload

-Matthew


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RE: [Asterisk-Users] Paging and intercom

[Paul Hales]

I was under the impression that the intercom function on the Snom phones only 
worked if you did some hacking to asterisk...is that still the case?


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Saturday, 30 April 2005 7:33 AM
To: Jacob Cazzell; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Paging and intercom

Polycom phones and Snom phones supoprt paging. 

As far as your Overhead paging all you need is an FXO port on your system. The 
* system will work perfectly with this. Even allowing the zones to be set from 
the dialplan so your users won't need to learn any new 'paging codes'

Email me off -list of you need some help. 



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jacob Cazzell
Sent: Friday, April 29, 2005 5:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Paging and intercom

Hello all,

We are considering implementing a new system based on Asterisk on the back end. 
 I am very intrigued by the IP phones, but I have two questions regarding 
paging and intercom functions.

I know that * supports these functions, but I'm not sure I fully understand 
how.  On our existing phone system, if you dial an extention the other end will 
beep and then setup an intercom channel that's hands free for the called 
station.  I'm not sure how this would be duplicated in *, or is it more of a 
function of the phone we use?

We also have an overhead paging system, our current system is tied into a 
Valcom 3-zone paging system.  Would * support this paging system?  How do you 
get a connection to it, an analog port?

These are probably my two biggest hurdles to overcome and I need some pointers 
on how to implement or where to research my options.

Thanks!
Jacob
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RE: [Asterisk-Users] A good SIP receptionist phone

[Paul Hales]

Rumour has is that Polycom will be releasing a reception console... 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Brown
Sent: Sunday, 1 May 2005 5:08 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] A good SIP receptionist phone

I have a problem. The average person is too freaking stupid to use a VOIP 
phone. My experience has so far been that if it doesn't have 20 buttons with 
little red LED's on it, the user cannot comprehend call parking, attended 
transfer, blind transfer, DND, and navigating through a voicemail menu.

I need a good receptionist phone that works with Asterisk. It basically needs 
to act like an avaya partner phone, I don't need 20 buttons with little red 
LED's...what I do need is for the phone to register multiple extensions to my 
asterisk server and act like each SIP extension is a line, so if the idiot 
receptionist has a call ringing in on line 1, she can pick it up, look at the 
buttons, see a call ringing in on line 2 (and the phone ringer rings), put call 
1 on hold without hanging the caller up, and hit the little I am an idiot and 
need a line 2 button to pick up line 2, so on and so forth.

I love VOIP systems and all the functionality they bring and features I get. 
Unfortunately, the average person in this country anymore is apparently 
completely stupid and cannot understand how to juggle calls without hanging up 
on people.

/rant

So seriously does anyone have a recommendation for a good receptionist phone? I 
tried the Snom today and I can't get the programmable buttons to do this, even 
by following the manual. So please, any suggestions would be great, before I 
get fired at my dayjob for everyone else's idiocy.

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Re: [Asterisk-Users] Pre-Parse Extensions.conf?

[Kevin P. Fleming]

Matthew Boehm wrote:
I think I said it wrong. I'm not looking to pre-compile the dialplan;
No, I understood what you meant :-)
Looking for more of a syntax checker than anything. Something like that
isn't possible?
Yes, but it won't look into the data provided in each extension 
priority, because that is parsed by the apps themselves.

In addition, there are various bits of the dialplan load (merging 
contexts together, etc.) that can't be done without looking at the one 
that is currently in memory, or it would miss potential errors.

Sorta like extensions reload [check] or extensions check|reload
I agree it would be handy to have, but I think the implementation is 
non-trivial at best.
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[Asterisk-Users] asterisk and USRobotics Courier V.Everything

[Guilherme Baião]

hi for all, would like to know if asterisk supports the modem USRobotics
Courier V.Everything.  Looking for very I found one link that it says on
a called module chan_modem_usr2976.so that the principle would function,
but lowering the sources of asterisk I did not find this module,
somebody can help me?


thanks for all

Guilherme


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Re: [Asterisk-Users] Programing a call forward feature to cel phones

[Matt Riddell]

What have you got for rxgain and txgain for the channels that are going 
out over the PSTN?

It sounds to me like you have echo above 0dB which would mean that it 
would get louder with each iteration.

This could be achieved by having bad echo as well as having the gains 
set too high.

Cool sound file!  Mind if I use it in an industrial techno song?
:)
--
Cheers,
Matt Riddell
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[Asterisk-Users] post-dial variable for whoever answered?

[Tim Connolly]

 Im
looking for suggestions on how to use changemonitor() to modify the filename
monitor() is writing to, once an extension has picked up the call. For
instance, the filename is currently accounting-${TIMESTAMP}-${CALLERIDNUM}.
Id like to do a changemonitor() to make the filename accounting-${TIMESTAMP}-${CALLERIDNUM}-ext
where ext is the extension number of the SIP client who actually answered.
Keep in mind, its dialing multiple destinations, so I dont know what
number picks up beforehand.



Thanks

Tim










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[Asterisk-Users] Set up SIP, now I'm getting a busy tone and weird (to me) messages on the console

[beonice]

Folks,

I'm hoping someone has already run into this ... the
only other complaint I've seen is here:

http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000640.html

and that basically was a problem with the /etc/hosts
... my server is definitely described in my hosts
file.

I've been using asterisk with IAX and a voicepulse
connect number. No problems at all receiving calls.

Now, I've just purchased a DID in Canada from another
provider, and their proxy only supports SIP. So,
following the generic instructions I've found off the
web, I set up my SIP.conf to point to voicepulse's
server, and set up the other DID to point into this
newly defined sip context, i.e., to
uid:[EMAIL PROTECTED]/888

The problem? The remote DID, when called, simply gives
me a busy signal. Also, on the asterisk console, I'm
seeing these messages that don't tell me anything:

May  1 18:37:09 WARNING[12065]: chan_sip.c:695
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno
103 (Critical Request)
May  1 18:37:23 NOTICE[12065]: chan_sip.c:4036
sip_reg_timeout:-- Registration for
'[EMAIL PROTECTED]' timed out, trying again
May  1 18:37:23 DEBUG[12065]: chan_sip.c:4150
transmit_register: Scheduled a registration timeout #
5
May  1 18:37:29 WARNING[12065]: chan_sip.c:695
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno
104 (Critical Request)
May  1 18:37:43 NOTICE[12065]: chan_sip.c:4036
sip_reg_timeout:-- Registration for
'[EMAIL PROTECTED]' timed out, trying again
May  1 18:37:43 DEBUG[12065]: chan_sip.c:4150
transmit_register: Scheduled a registration timeout #
7




It looks like the remote DID is failing to register
with the voicepulse server. Any hints on what could be
the problem?

If it helps, here is the relevant portion of my
sip.conf file.

[general]
;context=default; Default
context for incoming calls
context=unwelcome-calls ; Default context for
incoming calls
; After all, we don't
want any random
; incoming calls to
have access to outbound
; calling - Maya
Kurup, May 1, 2005
;recordhistory=yes  ; Record SIP history
by default
; (see sip history /
sip no history)
;realm=mydomain.tld ; Realm for digest
authentication
; defaults to
asterisk
; Realms MUST be
globally unique according to RFC 3261
; Set this to your
host name or domain name
port=5060   ; UDP Port to bind to
(SIP standard port is 5060)
bindaddr=0.0.0.0; IP address to bind
to (0.0.0.0 binds to all)

...

register = uid:[EMAIL PROTECTED]

; We need to allow at least incoming calls to
; accept calls via libretel, etc.
; So, let's add a context for that:

[888]; For incoming calls ONLY
type=user ; This device takes incoming calls
username=uid ; Username on device
secret=secret ; Password for device
host=srvr.voicepulse.com  ; This host will not 
  ; change frequently
context=allowed_context  ; Inbound calls from 
 ; this host go
 ; to the normal context

 -

and I have allowed_context described in my
extensions.conf, it's the same one I'm using for
regular IAX incoming calls, and works fine.
The context for unwelcome-calls is as follows:

[unwelcome-calls]
;
; Take unknown callers that may have found
; our system, and send them to a re-order tone.
; The string _. matches any dialed sequence, so all
; calls will result in the Congestion tone application
; being called. They'll get bored and hang up
eventually.
;

exten = _.,1,Congestion
---


Any help would be appreciated.

Thanks,
Maya


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RE: [Asterisk-Users] Dutch SIP or IAX numbers

[Asterisk]

Thanks,

But a small correction Dutchphone works only with one addpack modem and can
only work with asterisk in combination with a X100P interface.

It is the same solution as pilmo.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michiel van
Baak
Sent: Sunday, May 01, 2005 2:01 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Dutch SIP or IAX numbers

On 12:23, Sun 01 May 05, Asterisk wrote:
 How knows where I can get a Dutchphone number for asterisk?
 
 Pilmo is not delivering one for home use.

Three I use:

http://www.speakup.nl (for now only busines accounts)
http://www.talkin2ya.nl (both prepaid and postpaid)
http://www.dutchphone.nl (only postpaid)
-- 
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Two of the most famous products of Berkeley are LSD and BSD. I don't think
that this is a coincidence.

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RE: [Asterisk-Users] Set up SIP, now I'm getting a busy tone and weird (to me)messages on the console

[Tim Connolly]

Is NAT=yes on, are you behind a firewall? Give us some connectivity details.
Usually when you see maximum retries, its because you have one-way
communications with the far end for some reason. Are you setting externip
statically?



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of beonice
Sent: Sunday, May 01, 2005 8:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Set up SIP,now I'm getting a busy tone and weird
(to me)messages on the console

Folks,

I'm hoping someone has already run into this ... the
only other complaint I've seen is here:

http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000640.html

and that basically was a problem with the /etc/hosts
... my server is definitely described in my hosts
file.

I've been using asterisk with IAX and a voicepulse
connect number. No problems at all receiving calls.

Now, I've just purchased a DID in Canada from another
provider, and their proxy only supports SIP. So,
following the generic instructions I've found off the
web, I set up my SIP.conf to point to voicepulse's
server, and set up the other DID to point into this
newly defined sip context, i.e., to
uid:[EMAIL PROTECTED]/888

The problem? The remote DID, when called, simply gives
me a busy signal. Also, on the asterisk console, I'm
seeing these messages that don't tell me anything:

May  1 18:37:09 WARNING[12065]: chan_sip.c:695
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno
103 (Critical Request)
May  1 18:37:23 NOTICE[12065]: chan_sip.c:4036
sip_reg_timeout:-- Registration for
'[EMAIL PROTECTED]' timed out, trying again
May  1 18:37:23 DEBUG[12065]: chan_sip.c:4150
transmit_register: Scheduled a registration timeout #
5
May  1 18:37:29 WARNING[12065]: chan_sip.c:695
retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno
104 (Critical Request)
May  1 18:37:43 NOTICE[12065]: chan_sip.c:4036
sip_reg_timeout:-- Registration for
'[EMAIL PROTECTED]' timed out, trying again
May  1 18:37:43 DEBUG[12065]: chan_sip.c:4150
transmit_register: Scheduled a registration timeout #
7




It looks like the remote DID is failing to register
with the voicepulse server. Any hints on what could be
the problem?

If it helps, here is the relevant portion of my
sip.conf file.

[general]
;context=default; Default
context for incoming calls
context=unwelcome-calls ; Default context for
incoming calls
; After all, we don't
want any random
; incoming calls to
have access to outbound
; calling - Maya
Kurup, May 1, 2005
;recordhistory=yes  ; Record SIP history
by default
; (see sip history /
sip no history)
;realm=mydomain.tld ; Realm for digest
authentication
; defaults to
asterisk
; Realms MUST be
globally unique according to RFC 3261
; Set this to your
host name or domain name
port=5060   ; UDP Port to bind to
(SIP standard port is 5060)
bindaddr=0.0.0.0; IP address to bind
to (0.0.0.0 binds to all)

...

register = uid:[EMAIL PROTECTED]

; We need to allow at least incoming calls to
; accept calls via libretel, etc.
; So, let's add a context for that:

[888]; For incoming calls ONLY
type=user ; This device takes incoming calls
username=uid ; Username on device
secret=secret ; Password for device
host=srvr.voicepulse.com  ; This host will not 
  ; change frequently
context=allowed_context  ; Inbound calls from 
 ; this host go
 ; to the normal context

 -

and I have allowed_context described in my
extensions.conf, it's the same one I'm using for
regular IAX incoming calls, and works fine.
The context for unwelcome-calls is as follows:

[unwelcome-calls]
;
; Take unknown callers that may have found
; our system, and send them to a re-order tone.
; The string _. matches any dialed sequence, so all
; calls will result in the Congestion tone application
; being called. They'll get bored and hang up
eventually.
;

exten = _.,1,Congestion
---


Any help would be appreciated.

Thanks,
Maya


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Re: [Asterisk-Users] post-dial variable for whoever answered?

[Kevin P. Fleming]

Tim Connolly wrote:
accounting-${TIMESTAMP}-${CALLERIDNUM}. I'd like to do a changemonitor()
to make the filename accounting-${TIMESTAMP}-${CALLERIDNUM}-ext where
ext is the extension number of the SIP client who actually answered. Keep
in mind, its dialing multiple destinations, so I don't know what number
picks up beforehand.
From doc/README.variables:
${BRIDGEPEER}bridged peer
This will contain the channel name of the channel that answered the 
call; it will not contain an 'extension number', as there is no such 
thing at point (since you could have dialed a peer that doesn't even 
_have_ an extension number in your dialplan).
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RE: [Asterisk-Users] Asterisk@Home bug

[Alejandro Kauffmann]

We installed AAH .05, tweaked it and learned more about dialplans, Queues
(not included in that version of AMP), upgrading to CVS Head (needed
atxfer/automon) and anything else we needed to scale AAH to our needs (75
agents 15k+ calls/day)than I believe we would have learned by simply trying
to start with asterisk.  AAH does a great job of showing people new to *
what it can do once it's all put together.  

While I agree that AAH questions tend to be simple in nature and in some
cases, as this one, not related to * mainstream installations, most of the
questions deal with AMP, zaptel, libpri, spandsp, .conf files, etc.  Please
try not to immediately dismiss AAH questions to the forums simply because
they are prefixed by need help AAH.

Now as for this question in particular, come on guys do a bit of leg
work

From install_addon.sh (added in .9)

echo  ---
echo |Installing RS-232 console on COM1  |
echo  ---
echo 
echo 
echo 
echo 
if ! grep ttyS0 /etc/securetty /dev/null 21; then
echo s0:12345:respawn:/sbin/agetty -i -h -L 9600 ttyS0 vt100 
/etc/inittab
echo ttyS0  /etc/securetty
fi

Try downloading the tarball and looking at what it is that you are blindly
installing on your system everytime you download the iso and burn it.

/rant off

Alex

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Saturday, April 30, 2005 6:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] bug


This is not that AAH mailing list, check out the fourms.



On Sat, 30 Apr 2005, Manny A. Wise wrote:

 After installation of [EMAIL PROTECTED] v1, I have an annoying message in 
 the screen, anyone know how to fix it



 INIT: Id s0 respawning too fast: disable for 5 minutes



 Thanks


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RE: [Asterisk-Users] Programing a call forward feature to cel phones

[Anton Krall]

:) go ahead and send me a copy when you are done :)

My gains are around 6 so you might be right... Maybe lowering them to 3 or
so...  

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Matt Riddell
|Sent: Domingo, 01 de Mayo de 2005 08:13 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Programing a call forward 
|feature to cel phones
|
|What have you got for rxgain and txgain for the channels that 
|are going out over the PSTN?
|
|It sounds to me like you have echo above 0dB which would mean 
|that it would get louder with each iteration.
|
|This could be achieved by having bad echo as well as having 
|the gains set too high.
|
|Cool sound file!  Mind if I use it in an industrial techno song?
|
|:)
|
|--
|Cheers,
|
|Matt Riddell
|___
|
|http://www.sineapps.com/news.php (Daily Asterisk News - html) 
|http://www.sineapps.com/rssfeed.php (Daily Asterisk News - 
|rss) ___
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|

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RE: [Asterisk-Users] Set up SIP, now I'm getting a busy tone and weird (to me)messages on the console

[beonice]

--- Tim Connolly [EMAIL PROTECTED] wrote:

 Is NAT=yes on, are you behind a firewall? Give us
 some connectivity details.
 Usually when you see maximum retries, its because
 you have one-way
 communications with the far end for some reason. Are
 you setting externip
 statically?

To answer your questions, yes, I am behind a firewall.
The asterisk server is the only device connected to a
cheapo Netgear 4-port router/firewall. I'm not setting
externip myself, so whatever the default is, it's
getting used. I'm also NOT making outgoing calls, and
there are no actual SIP devices attached ... I'm just
trying to receive incoming calls forwarded from a
different provider via SIP.

Here is a complete sip.conf file ... do I need to
provide anything else?

sip.conf:
;
; SIP Configuration for Asterisk
;
; Syntax for specifying a SIP device in
extensions.conf is
; SIP/devicename where devicename is defined in a
section below.
;
; You may also use
; SIP/[EMAIL PROTECTED] to call any SIP user on the
Internet
; (Don't forget to enable DNS SRV records if you want
to use this)
;
; If you define a SIP proxy as a peer below, you may
call
; SIP/proxyhostname/user or SIP/[EMAIL PROTECTED]
; where the proxyhostname is defined in a section
below
;
; Useful CLI commands to check peers/users:
;   sip show peers  Show all SIP peers
(including friends)
;   sip show users  Show all SIP users
(including friends)
;   sip show registry   Show status of hosts
we register with
;
;   sip debug   Show all SIP messages
;

[general]
;context=default; Default
context for incoming calls
context=unwelcome-calls ; Default context for
incoming calls
; After all, we don't
want any random
; incoming calls to
have access to outbound
; calling 
;recordhistory=yes  ; Record SIP history
by default
; (see sip history /
sip no history)
;realm=mydomain.tld ; Realm for digest
authentication
; defaults to
asterisk
; Realms MUST be
globally unique according to RFC 3261
; Set this to your
host name or domain name
port=5060   ; UDP Port to bind to
(SIP standard port is 5060)
bindaddr=0.0.0.0; IP address to bind
to (0.0.0.0 binds to all)
srvlookup=yes   ; Enable DNS SRV
lookups on outbound calls
; Note: Asterisk only
uses the first host
; in SRV records
; Disabling DNS SRV
lookups disables the
; ability to place SIP
calls based on domain
; names to some other
SIP users on the Internet

;pedantic=yes   ; Enable slow,
pedantic checking for Pingtel
; and multiline
formatted headers for strict
; SIP compatibility
(defaults to no)
;tos=184; Set IP QoS to either
a keyword or numeric val
;tos=lowdelay   ;
lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600; Max length of
incoming registration we allow
;defaultexpirey=120 ; Default length of
incoming/outoing registration
;notifymimetype=text/plain  ; Allow overriding of
mime type in MWI NOTIFY
;videosupport=yes   ; Turn on support for
SIP video

;disallow=all   ; First disallow all
codecs
;allow=ulaw ; Allow codecs in
order of preference
;allow=ilbc ; Note: codec order is
respected only in [general]
;musicclass=default ; Sets the default
music on hold class for all SIP calls
; This may also be set
for individual users/peers
;language=en; Default language
setting for all users/peers
; This may also be set
for individual users/peers
;relaxdtmf=yes  ; Relax dtmf handling
;rtptimeout=60  ; Terminate call if 60
seconds of no RTP activity
; when we're not on
hold
;rtpholdtimeout=300 ; Terminate call if
300 seconds of no RTP activity
; when we're on hold
(must be  rtptimeout)
;trustrpid = no ; If Remote-Party-ID
should be trusted
;progressinband=no  ; If we should
generate in-band ringing always
;useragent=Asterisk PBX ; Allows you to change
the user agent string
;nat=no ; NAT settings
; yes = Always ignore
info and assume NAT
; no = Use NAT mode
only according to RFC3581
; never = Never
attempt NAT mode or RFC3581 

Re: [Asterisk-Users] Problem with Sangoma/Adtran 600 installation

[Michael Bielicki]

the same you want it to be linked to in zaptel.conf
so if you config it as span1 in zaptel.conf it has to say 1


On 5/1/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
   x lk x
   x x   Interface Name- w1g1x x
   x x   Operation Mode- TDM_VOICE   x x
   x x   TDM Voice Span- 1  x x
   x x   Override Asterisk Echo Enable - No
 
 What should the TDM Voice Span be?
 
 Chris Mason
 www.anguillaguide.com
 
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-- 
Michal Bielicki
http://www.aefirion.org/
http://www.asterisk.com.pl/
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[Asterisk-Users] how to disconnect a call manually

[Asterisk guy]

1 after giving command oh323 show channels,  

i want to disconnect a call,  is there any command  to disconnect a call?  


2 how asterisk kill a hung/dead call ?  for most commercial
softswitch, there are a setting for maximum duration for a call. they
will hang up it l if its duration reachs the limit.
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[Asterisk-Users] which port is used when asterisk -r

[Asterisk guy]

which TCP port  is used when asterisk -r   ?

is there a command  to connect to a remote machine ?

( asterisk -r  remote-machine-ip  ?)
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RE: [Asterisk-Users] which port is used when asterisk -r

[McMorrine, Mark]

Asterisk -r is used to reconnect to CLI of a running Asterisk system from
the console, so there is no TCP port in use to do that.  You sould SSH to
your Asterisk server and asterisk -r to interact with the running Asterisk
application.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk guy
Sent: Sunday, May 01, 2005 7:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] which port is used when asterisk -r 


which TCP port  is used when asterisk -r   ?

is there a command  to connect to a remote machine ?

( asterisk -r  remote-machine-ip  ?)
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[Asterisk-Users] Caller Hears Ring During Attended Transfer?

[Daryll Strauss]

I'm running Asterisk 1.0.7 and SPA-3000 Firmware 2.0.13g. 

A call comes in to my asterisk box via SIP (the Sipura isn't involved)
and I answer it using an analog phone on the Sipura. I then decide to
forward it to another phone. I flash the line, dial the new extension,
and flash again. At this point I and the caller hear the destination
phone ring. If I then hang up, the caller stops hearing the ring.
Eventually the voicemail picks up and the caller hears the
announcement and everything works as normal. It's just rather
disconcerting for the caller to not hear anything.

Is this an Asterisk bug, Sipura bug, or operator (me) error?

- |Daryll
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Re: [Asterisk-Users] Caller Hears Ring During Attended Transfer?

[Eric Wieling aka ManxPower]

Daryll Strauss wrote:
I'm running Asterisk 1.0.7 and SPA-3000 Firmware 2.0.13g. 

A call comes in to my asterisk box via SIP (the Sipura isn't involved)
and I answer it using an analog phone on the Sipura. I then decide to
forward it to another phone. I flash the line, dial the new extension,
and flash again. At this point I and the caller hear the destination
phone ring. If I then hang up, the caller stops hearing the ring.
Eventually the voicemail picks up and the caller hears the
announcement and everything works as normal. It's just rather
disconcerting for the caller to not hear anything.
Is this an Asterisk bug, Sipura bug, or operator (me) error?
This is a known (by at least some people) problem.  I don't know of a 
fix or a bug number.
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Re: [Asterisk-Users] Caller Hears Ring During Attended Transfer?

[Eric Wieling aka ManxPower]

Eric Wieling aka ManxPower wrote:
Daryll Strauss wrote:
I'm running Asterisk 1.0.7 and SPA-3000 Firmware 2.0.13g.
A call comes in to my asterisk box via SIP (the Sipura isn't involved)
and I answer it using an analog phone on the Sipura. I then decide to
forward it to another phone. I flash the line, dial the new extension,
and flash again. At this point I and the caller hear the destination
phone ring. If I then hang up, the caller stops hearing the ring.
Eventually the voicemail picks up and the caller hears the
announcement and everything works as normal. It's just rather
disconcerting for the caller to not hear anything.
Is this an Asterisk bug, Sipura bug, or operator (me) error?
Happens with Polycom phones as well.  It may be related to this message 
when a transfer happens:

Unable to handle indication 3 for 'SIP/0004f201e4b3-a-682d'

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