Re: [asterisk-users] extension not found

[Ben Schorr]

Is there some reason why I keep getting this same message from "cool
dude" over and over and over?  And under different subject lines?

 

Ben M. Schorr
Chief Executive Officer
__
Roland Schorr & Tower
www.rolandschorr.com  
b...@rolandschorr.com

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cool dude
Sent: Friday, February 12, 2010 21:24
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] extension not found

 

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005
can make call outside and exten 2006 to 2010 can not make call outside.
heres my dial plan.
 
sip.conf
 
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic

[2001]
type=friend
context=outside
secret=1234
host=dynamic

[2002]
type=friend
context=outside
secret=1234
host=dynamic

[2003]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2004]
type=friend
contex=outside
secret=1234
host=dynamic

[2005]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2006]
type=friend
contex=internal
secret=1234
host=dynamic

[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic

[2009]
type=friend
contex=internal
secret=1234
host=dynamic

[2010]
type=friend
contex=internal
secret=1234
host=dynamic


## ##
vi /etc/asterisk/extensions.conf
[from-zaptel]
exten => s,1,wait(2)
exten => s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten => s,n,Playback(tt-weasels)
[others]
include => internal
include => outside
[internal]
exten => _20XX,1,Dial(SIP/${EXTEN})
exten => _20XX,n,VoiceMail(${ext...@others,u)
exten => _20XX,n,Hangup()
[outside]
exten => 2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup
exten => 2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup
exten => 2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup
exten => 2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup
exten => 2005,1,Dial(Zap/1-1/${EXTEN})
exten => 2005,n,Hangup

this is the log when i am calling from exten 2000 to outside

Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite:
Call from '2002' to extension '9193696136' rejected because extension
not found.

 



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[asterisk-users] extension not found

[cool dude]

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can 
make call outside and exten 2006 to 2010 can not make call outside. heres my 
dial plan.
 
sip.conf
 
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic

[2001]
type=friend
context=outside
secret=1234
host=dynamic

[2002]
type=friend
context=outside
secret=1234
host=dynamic

[2003]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2004]
type=friend
contex=outside
secret=1234
host=dynamic

[2005]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2006]
type=friend
contex=internal
secret=1234
host=dynamic

[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic

[2009]
type=friend
contex=internal
secret=1234
host=dynamic

[2010]
type=friend
contex=internal
secret=1234
host=dynamic


vi /etc/asterisk/extensions.conf
[from-zaptel]
exten => s,1,wait(2)
exten => s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten => s,n,Playback(tt-weasels)
[others]
include => internal
include => outside
[internal]
exten => _20XX,1,Dial(SIP/${EXTEN})
exten => _20XX,n,VoiceMail(${ext...@others,u)
exten => _20XX,n,Hangup()
[outside]
exten => 2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup
exten => 2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup
exten => 2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup
exten => 2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup
exten => 2005,1,Dial(Zap/1-1/${EXTEN})
exten => 2005,n,Hangup

this is the log when i am calling from exten 2000 to outside

Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite: Call 
from '2002' to extension '9193696136' rejected because extension not found.


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[asterisk-users] Robotic sound sometimes

[Rudi Oosthuizen]

>> We experienced this a couple of months ago. It went away when we
upgraded the phones to the latest firmware.
>> Another symptom: temporarily putting the caller on hold cures the
problem sometimes.

We have Snom 320 phones and had similar problems happening in the call
centre intermittently. We resolved by running a weekly script clearing
the 
phones memory. UPGRADES did not work for us. Hope this helps some one.

Rudi Oosthuizen  

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Re: [asterisk-users] PAP2

[Brian]

On Fri, 2010-02-12 at 17:23 -0400, Tim Johnson wrote:
> I know this is slightly off topic, but I was wondering if anyone can help
> with a problem getting my PAP2's to connect to Asterisk. I use a
> provisioning file, and I recently re-wrote the files for each PAP2. I had
> a small typo and the PAPs logged it as a corrupt file. I corrected the
> file, however, Line 1 on both of the PAP2's now wont register. Line 2
> works fine though. I've done the "  73738", but it wont come back.
> Anyone know of a way to really wipe it's memory?
> 
> Tim
> 
> 
This is a long shot - but check the Codec being selected for L1. Whilst
I don't recall the specifics, does it not default to a licenced codec by
default - this may perhaps be an issue? Like I say - it's a long shot
and it may be totally irrelevant.


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Re: [asterisk-users] Robotic sound sometimes

[cool dude]

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can 
make call outside and exten 2006 to 2010 can not make call outside. heres my 
dial plan.
 
sip.conf
 
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic

[2001]
type=friend
context=outside
secret=1234
host=dynamic

[2002]
type=friend
context=outside
secret=1234
host=dynamic

[2003]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2004]
type=friend
contex=outside
secret=1234
host=dynamic

[2005]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2006]
type=friend
contex=internal
secret=1234
host=dynamic

[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic

[2009]
type=friend
contex=internal
secret=1234
host=dynamic

[2010]
type=friend
contex=internal
secret=1234
host=dynamic


vi /etc/asterisk/extensions.conf
[from-zaptel]
exten => s,1,wait(2)
exten => s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten => s,n,Playback(tt-weasels)
[others]
include => internal
include => outside
[inside]
exten => _20XX,1,Dial(SIP/${EXTEN})
exten => _20XX,n,VoiceMail(${ext...@others,u)
exten => _20XX,n,Hangup()
[outside]
exten => 2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup
exten => 2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup
exten => 2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup
exten => 2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup
exten => 2005,1,Dial(Zap/1-1/${EXTEN})
exten => 2005,n,Hangup

this is the log when i am calling from exten 2000 to outside

Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite: Call 
from '2002' to extension '919369613616' rejected because extension not found.
 
 
any help n support will be highly appreciated

--- On Sat, 13/2/10, Michelle Dupuis  wrote:


From: Michelle Dupuis 
Subject: Re: [asterisk-users] Robotic sound sometimes
To: "'Asterisk Users List'" 
Date: Saturday, 13 February, 2010, 3:35 AM



#yiv1877693794 P {
MARGIN:0px;}


No, phone on LAN, through Asterisk box on LAN, through firewall, out to fiber 
connection (lots of capacity there), to ITSP.
 
Codec is uLaw.
 
This only happens sometimes, so I'm wondering if it's an asterisk bug? Aastra 
bug?  Network latency?  LAN capacity, etc.  Never seen this before...



From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Friday, February 12, 2010 4:34 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Robotic sound sometimes



What codecs are you using? Are the calls internal(local network) only?

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105

- "Michelle Dupuis"  wrote: 
> 
We have inherited an installation with Ast 1.4 and Aastra phones.  The client 
complains that sometimes the call audio turns tinny and robotic...I heard it 
and it sounds wierd.
 
Has anyone else experienced this?  Cause?  Solutions?
 
Thanks,
MD
> -- _ -- 
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Re: [asterisk-users] Robotic sound sometimes

[cool dude]

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can 
make call outside and exten 2006 to 2010 can not make call outside. heres my 
dial plan.
 
sip.conf
 
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic

[2001]
type=friend
context=outside
secret=1234
host=dynamic

[2002]
type=friend
context=outside
secret=1234
host=dynamic

[2003]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2004]
type=friend
contex=outside
secret=1234
host=dynamic

[2005]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2006]
type=friend
contex=internal
secret=1234
host=dynamic

[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic

[2009]
type=friend
contex=internal
secret=1234
host=dynamic

[2010]
type=friend
contex=internal
secret=1234
host=dynamic


vi /etc/asterisk/extensions.conf
[from-zaptel]
exten => s,1,wait(2)
exten => s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten => s,n,Playback(tt-weasels)
[others]
include => internal
include => outside
[inside]
exten => _20XX,1,Dial(SIP/${EXTEN})
exten => _20XX,n,VoiceMail(${ext...@others,u)
exten => _20XX,n,Hangup()
[outside]
exten => 2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup
exten => 2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup
exten => 2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup
exten => 2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup
exten => 2005,1,Dial(Zap/1-1/${EXTEN})
exten => 2005,n,Hangup

this is the log when i am calling from exten 2000 to outside

Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite: Call 
from '2002' to extension '919369613616' rejected because extension not found.
 
 
any help n support will be highly appreciated

--- On Sat, 13/2/10, Peder  wrote:


From: Peder 
Subject: Re: [asterisk-users] Robotic sound sometimes
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" 

Date: Saturday, 13 February, 2010, 4:25 AM


There is a statistics area and you can select sip or voip calls to see
calls.  It shows packet loss, jitter, latency, out of sequence packets, etc.
It can even play them back, so you can check where the loss is and play back
the call to see if the noise is in the same spot.  Here is some info from
the wireshark website:

http://wiki.wireshark.org/VoIP_calls



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, February 12, 2010 4:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Robotic sound sometimes

On Fri, 12 Feb 2010, Peder wrote:

> Since it is sporadic, my guess would be network latency / packet loss 
> /jitter to ITSP.  You may have lots of capacity and they may claim to 
> have lots of capacity, but what about the links between you and them. 
> Who knows when/if there is loss and latency and jitter there.  Setup 
> wireshark to grab some calls and when someone complains about it, look 
> at the stats for that call.  It will tell you if there is loss or 
> latency or jitter.

Do you know of a link that explains how to detect latency or jitter? My 
wireshark skills are pretty non-existent. Does it have a wizzy filter that 
will tell you or do you need to check the timestamps of the RTP packets? 
Any chance you could write up your technique?

-- 
Thanks in advance,
-
Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
Newline                                              Fax: +1-760-731-3000

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Re: [asterisk-users] Robotic sound sometimes

[cool dude]

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can 
make call outside and exten 2006 to 2010 can not make call outside. heres my 
dial plan.
 
sip.conf
 
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic

[2001]
type=friend
context=outside
secret=1234
host=dynamic

[2002]
type=friend
context=outside
secret=1234
host=dynamic

[2003]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2004]
type=friend
contex=outside
secret=1234
host=dynamic

[2005]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2006]
type=friend
contex=internal
secret=1234
host=dynamic

[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic

[2009]
type=friend
contex=internal
secret=1234
host=dynamic

[2010]
type=friend
contex=internal
secret=1234
host=dynamic


vi /etc/asterisk/extensions.conf
[from-zaptel]
exten => s,1,wait(2)
exten => s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten => s,n,Playback(tt-weasels)
[others]
include => internal
include => outside
[inside]
exten => _20XX,1,Dial(SIP/${EXTEN})
exten => _20XX,n,VoiceMail(${ext...@others,u)
exten => _20XX,n,Hangup()
[outside]
exten => 2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup
exten => 2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup
exten => 2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup
exten => 2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup
exten => 2005,1,Dial(Zap/1-1/${EXTEN})
exten => 2005,n,Hangup

this is the log when i am calling from exten 2000 to outside

Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite: Call 
from '2002' to extension '919369613616' rejected because extension not found.
 
 
any help n support will be highly appreciated

--- On Sat, 13/2/10, Ron Arts  wrote:


From: Ron Arts 
Subject: Re: [asterisk-users] Robotic sound sometimes
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Saturday, 13 February, 2010, 3:38 AM


We experienced this a couple of months ago. It went away when
we upgraded the phones to the latest firmware.
Another symptom: temporarily putting the caller on hold cures
the problem sometimes.

Ron


Op 12-02-10 22:34, Tim Nelson schreef:
> What codecs are you using? Are the calls internal(local network) only?
>
> Tim Nelson
> Systems/Network Support
> Rockbochs Inc.
> (218)727-4332 x105
>
> - "Michelle Dupuis"  wrote:
>  >
> We have inherited an installation with Ast 1.4 and Aastra phones. The
> client complains that sometimes the call audio turns tinny and
> robotic...I heard it and it sounds wierd.
> Has anyone else experienced this? Cause? Solutions?
> Thanks,
> MD
>
>  > --
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>


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Op dit bericht is de volgende disclaimer van toepassing:
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Re: [asterisk-users] Robotic sound sometimes

[cool dude]

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can 
make call outside and exten 2006 to 2010 can not make call outside. heres my 
dial plan.
 
sip.conf
 
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic

[2001]
type=friend
context=outside
secret=1234
host=dynamic

[2002]
type=friend
context=outside
secret=1234
host=dynamic

[2003]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2004]
type=friend
contex=outside
secret=1234
host=dynamic

[2005]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2006]
type=friend
contex=internal
secret=1234
host=dynamic

[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic

[2009]
type=friend
contex=internal
secret=1234
host=dynamic

[2010]
type=friend
contex=internal
secret=1234
host=dynamic


vi /etc/asterisk/extensions.conf
[from-zaptel]
exten => s,1,wait(2)
exten => s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten => s,n,Playback(tt-weasels)
[others]
include => internal
include => outside
[inside]
exten => _20XX,1,Dial(SIP/${EXTEN})
exten => _20XX,n,VoiceMail(${ext...@others,u)
exten => _20XX,n,Hangup()
[outside]
exten => 2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup
exten => 2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup
exten => 2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup
exten => 2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup
exten => 2005,1,Dial(Zap/1-1/${EXTEN})
exten => 2005,n,Hangup

this is the log when i am calling from exten 2000 to outside

Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite: Call 
from '2002' to extension '919369613616' rejected because extension not found.
 
 
any help n support will be highly appreciated

--- On Sat, 13/2/10, Steve Edwards  wrote:


From: Steve Edwards 
Subject: Re: [asterisk-users] Robotic sound sometimes
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Saturday, 13 February, 2010, 4:09 AM


On Fri, 12 Feb 2010, Peder wrote:

> Since it is sporadic, my guess would be network latency / packet loss 
> /jitter to ITSP.  You may have lots of capacity and they may claim to 
> have lots of capacity, but what about the links between you and them. 
> Who knows when/if there is loss and latency and jitter there.  Setup 
> wireshark to grab some calls and when someone complains about it, look 
> at the stats for that call.  It will tell you if there is loss or 
> latency or jitter.

Do you know of a link that explains how to detect latency or jitter? My 
wireshark skills are pretty non-existent. Does it have a wizzy filter that 
will tell you or do you need to check the timestamps of the RTP packets? 
Any chance you could write up your technique?

-- 
Thanks in advance,
-
Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
Newline                                              Fax: +1-760-731-3000

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Re: [asterisk-users] problems with 1.6

[cool dude]

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can 
make call outside and exten 2006 to 2010 can not make call outside. heres my 
dial plan.
 
sip.conf
 
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic

[2001]
type=friend
context=outside
secret=1234
host=dynamic

[2002]
type=friend
context=outside
secret=1234
host=dynamic

[2003]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2004]
type=friend
contex=outside
secret=1234
host=dynamic

[2005]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2006]
type=friend
contex=internal
secret=1234
host=dynamic

[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic

[2009]
type=friend
contex=internal
secret=1234
host=dynamic

[2010]
type=friend
contex=internal
secret=1234
host=dynamic


vi /etc/asterisk/extensions.conf
[from-zaptel]
exten => s,1,wait(2)
exten => s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten => s,n,Playback(tt-weasels)
[others]
include => internal
include => outside
[inside]
exten => _20XX,1,Dial(SIP/${EXTEN})
exten => _20XX,n,VoiceMail(${ext...@others,u)
exten => _20XX,n,Hangup()
[outside]
exten => 2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup
exten => 2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup
exten => 2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup
exten => 2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup
exten => 2005,1,Dial(Zap/1-1/${EXTEN})
exten => 2005,n,Hangup

this is the log when i am calling from exten 2000 to outside

Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite: Call 
from '2002' to extension '919369613616' rejected because extension not found.
 
 
any help n support will be highly appreciated

--- On Sat, 13/2/10, Jonathan Addleman  wrote:


From: Jonathan Addleman 
Subject: Re: [asterisk-users] problems with 1.6
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Saturday, 13 February, 2010, 4:54 AM


I'm still unable to do much with my new 1.6 installation. I just tried
reinstalling, and using the standard debian configuration files, with
just the necessary modifications, in case I had some legacy stuff in
there from earlier versions that was interfering. I'm testing in a xen
domU with debian's asterisk package, version 1:1.6.2.0-1

When I try to connect a channel into a conference, I still get all sorts
of "Unable to write to alert pipe" and then "Exceptionally long voice
queue length queuing" errors. I've tried this with meetme and
appkonference, with no difference in the error messages.

This happens with various sorts of channels - audio playing, eagi
scripts, sip connections. For example, I just tried this through the
manager interface:

Action: originate
Channel: Local/confere...@veco/n
Context: veco
Exten: playaudiofile
Priority: 1
Variable: tour=test
Variable: dir=
Variable: conference=ConferenceA
Variable: provider=teliax
Variable:
extravalue=/var/www/vecotourism/media//transcoded/audio/long-asterisk
Variable: title=Long test sound (long-asterisk.wav)


with this in extensions.conf:
exten => playaudiofile,1,Answer
exten => playaudiofile,n,Wait(1)
exten => playaudiofile,n,Playback(${extravalue})

exten => meetme,1,Answer()
exten => meetme,n,Wait(1)
exten => meetme,n,MeetMe(${conference}_${tour},1qd)

exten => conference,1,Answer()
exten => conference,n,Noop(Trying to start conference ${conference}_${tour})
exten => conference,n,konference(${conference}_${tour})



Seems this should be very straightforward, but it isn't working. What
might be wrong?
-- 
Jon-o Addleman - http://www.redowl.ca

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Re: [asterisk-users] Call Pickup with 1.6.2.1 and Snom

[cool dude]

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can 
make call outside and exten 2006 to 2010 can not make call outside. heres my 
dial plan.
 
sip.conf
 
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic

[2001]
type=friend
context=outside
secret=1234
host=dynamic

[2002]
type=friend
context=outside
secret=1234
host=dynamic

[2003]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2004]
type=friend
contex=outside
secret=1234
host=dynamic

[2005]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2006]
type=friend
contex=internal
secret=1234
host=dynamic

[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic

[2009]
type=friend
contex=internal
secret=1234
host=dynamic

[2010]
type=friend
contex=internal
secret=1234
host=dynamic


vi /etc/asterisk/extensions.conf
[from-zaptel]
exten => s,1,wait(2)
exten => s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten => s,n,Playback(tt-weasels)
[others]
include => internal
include => outside
[inside]
exten => _20XX,1,Dial(SIP/${EXTEN})
exten => _20XX,n,VoiceMail(${ext...@others,u)
exten => _20XX,n,Hangup()
[outside]
exten => 2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup
exten => 2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup
exten => 2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup
exten => 2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup
exten => 2005,1,Dial(Zap/1-1/${EXTEN})
exten => 2005,n,Hangup

this is the log when i am calling from exten 2000 to outside

Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite: Call 
from '2002' to extension '919369613616' rejected because extension not found.
 
 
any help n support will be highly appreciated
--- On Sat, 13/2/10, Loris Santamaria  wrote:


From: Loris Santamaria 
Subject: [asterisk-users] Call Pickup with 1.6.2.1 and Snom
To: asterisk-users@lists.digium.com
Date: Saturday, 13 February, 2010, 8:39 AM


Hi,

I've used various patches with asterisk 1.4 to have support for call
pickup and notification with good results.

Now I'm trying vanilla 1.6.2 with its official support for "dialog-info
+xml" notifications with no success. This is what i'm doing:

- Phone A has a key configured as type "extension" pointing to Phone B.
- In sip.conf I added notifycid=ignore-context
- Phone A and B and C are in the same callgroup and pickupgroup
- Phone A and B and C are in the same context

Phone C calls Phone B and asterisk generates a notification for phone A:





sip:35...@10.40.23.179



sip:35...@10.40.23.179


early



With this notification, Phone A shows on the screen that Phone C is
calling Phone B, and the function key blinks. If one presses the
blinking function key, the phone generates an Invite with replaces, to
try to pickup the call:

INVITE sip:35...@10.40.23.179 SIP/2.0
Via: SIP/2.0/UDP 10.40.24.175:5060;branch=z9hG4bK-qoz3zjhmyfcw;rport
From: "Lab 4" ;tag=o28fq65rfu
To: "Lab 1" 
Call-ID: 3c2672b3f35a-dpd0zv11yegl
CSeq: 1 INVITE
Max-Forwards: 70
Contact: ;flow-id=1
Replaces: pickup-3c26701519b8-5xxapzoav2u4
P-Key-Flags: keys="3"
User-Agent: snom320/7.1.39
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, 
MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 368

Then asterisk receives the pickup request:

[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use 
Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag:  Totag: 
[Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35...@rededelca
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no 
NAT)
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method 
INVITE - callid 3c2672b3f35a-dpd0zv11yegl
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis 
request - 3c2672b3f35a-dpd0zv11yegl
[...]
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: INVITE part of call transfer. 
Replaces [pickup-3c26701519b8-5xxapzoav2u4]
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use 
Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag:  Totag: 
[Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35...@rededelca
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no 
NAT)
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method 
INVITE - callid 3c2672b3f35a-dpd0zv11yegl
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis 
request - 3c2672b3f35a-dpd0zv11yegl
[...]
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: About to call Pickup(35..

Re: [asterisk-users] parked calls

[cool dude]

hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can 
make call outside and exten 2006 to 2010 can not make call outside. heres my 
dial plan.
 
sip.conf
 
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic

[2001]
type=friend
context=outside
secret=1234
host=dynamic

[2002]
type=friend
context=outside
secret=1234
host=dynamic

[2003]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2004]
type=friend
contex=outside
secret=1234
host=dynamic

[2005]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2006]
type=friend
contex=internal
secret=1234
host=dynamic

[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic

[2009]
type=friend
contex=internal
secret=1234
host=dynamic

[2010]
type=friend
contex=internal
secret=1234
host=dynamic


vi /etc/asterisk/extensions.conf
[from-zaptel]
exten => s,1,wait(2)
exten => s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten => s,n,Playback(tt-weasels)
[others]
include => internal
include => outside
[inside]
exten => _20XX,1,Dial(SIP/${EXTEN})
exten => _20XX,n,VoiceMail(${ext...@others,u)
exten => _20XX,n,Hangup()
[outside]
exten => 2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup
exten => 2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup
exten => 2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup
exten => 2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup
exten => 2005,1,Dial(Zap/1-1/${EXTEN})
exten => 2005,n,Hangup

this is the log when i am calling from exten 2000 to outside

Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite: Call 
from '2002' to extension '919369613616' rejected because extension not found.
 
 
any help n support will be highly appreciated

--- On Sat, 13/2/10, hin lee  wrote:


From: hin lee 
Subject: Re: [asterisk-users] parked calls
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Saturday, 13 February, 2010, 8:42 AM






Thank you Doug!  I added "courtesytone = beep" and that worked! 






From: Doug Lytle 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Fri, February 12, 2010 6:40:17 AM
Subject: Re: [asterisk-users] parked calls

hin lee wrote:
> Using FreePBX, is there a way to play a beep sound when you are 
> connected to a parked call? Right now, it's dead silence and we can't 
> tell if the call has been connected.
>
I don't know about FreePBX, but under the features.conf, there is:

courtesytone = local/stutter    ; Sound file to play to the parked caller
                                ; when someone dials a parked call

I recorded our Definity's stutter tone and put it in the local folder.

Doug


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Re: [asterisk-users] how to allow some extenstions to call outside and some extensions cant call outside

[cool dude]

hi friend,
thx for the reply i am trying this way to achieve what i want i.e exten 2000 to 
2005 can call outside and 2006 to 2010 cant call outside.
 
 
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic

[2001]
type=friend
context=outside
secret=1234
host=dynamic

[2002]
type=friend
context=outside
secret=1234
host=dynamic

[2003]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2004]
type=friend
contex=outside
secret=1234
host=dynamic

[2005]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2006]
type=friend
contex=internal
secret=1234
host=dynamic

[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic

[2009]
type=friend
contex=internal
secret=1234
host=dynamic

[2010]
type=friend
contex=internal
secret=1234
host=dynamic


vi /etc/asterisk/extensions.conf
[from-zaptel]
exten => s,1,wait(2)
exten => s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten => s,n,Playback(tt-weasels)
[others]
include => internal
include => outside
[inside]
exten => _20XX,1,Dial(SIP/${EXTEN})
exten => _20XX,n,VoiceMail(${ext...@others,u)
exten => _20XX,n,Hangup()
[outside]
exten => 2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup
exten => 2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup
exten => 2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup
exten => 2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup
exten => 2005,1,Dial(Zap/1-1/${EXTEN})
exten => 2005,n,Hangup

this is the log when i am calling from exten 2000 to outside
 
 
Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite: Call 
from '2002' to extension '919369613616' rejected because extension not found.

 
any help n support will be highly appreciated.
 
thx
--- On Sat, 13/2/10, Gergo Csibra  wrote:


From: Gergo Csibra 
Subject: Re: [asterisk-users] how to allow some extenstions to call outside and 
some extensions cant call outside
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Saturday, 13 February, 2010, 2:57 AM


Friday, February 12, 2010, 9:57:42 PM, cool wrote:

> how to allow some extenstions to call outside and some extensions
> cant call outside. i am attaching sipand extensions.conf
> thx

Put the extensions into different contexts, and create outside call
extensions only in the allowed context. Remember to create outside
calls for emergency numbers in the other context too.

-- 
Best regards,
Gergo                            mailto:csi...@gmail.com


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Re: [asterisk-users] parked calls

[hin lee]

Thank you Doug!  I added "courtesytone = beep" and that worked! 






From: Doug Lytle 
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Fri, February 12, 2010 6:40:17 AM
Subject: Re: [asterisk-users] parked calls

hin lee wrote:
> Using FreePBX, is there a way to play a beep sound when you are 
> connected to a parked call? Right now, it's dead silence and we can't 
> tell if the call has been connected.
>
I don't know about FreePBX, but under the features.conf, there is:

courtesytone = local/stutter; Sound file to play to the parked caller
 ; when someone dials a parked call

I recorded our Definity's stutter tone and put it in the local folder.

Doug


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[asterisk-users] Call Pickup with 1.6.2.1 and Snom

[Loris Santamaria]

Hi,

I've used various patches with asterisk 1.4 to have support for call
pickup and notification with good results.

Now I'm trying vanilla 1.6.2 with its official support for "dialog-info
+xml" notifications with no success. This is what i'm doing:

- Phone A has a key configured as type "extension" pointing to Phone B.
- In sip.conf I added notifycid=ignore-context
- Phone A and B and C are in the same callgroup and pickupgroup
- Phone A and B and C are in the same context

Phone C calls Phone B and asterisk generates a notification for phone A:





sip:35...@10.40.23.179



sip:35...@10.40.23.179


early



With this notification, Phone A shows on the screen that Phone C is
calling Phone B, and the function key blinks. If one presses the
blinking function key, the phone generates an Invite with replaces, to
try to pickup the call:

INVITE sip:35...@10.40.23.179 SIP/2.0
Via: SIP/2.0/UDP 10.40.24.175:5060;branch=z9hG4bK-qoz3zjhmyfcw;rport
From: "Lab 4" ;tag=o28fq65rfu
To: "Lab 1" 
Call-ID: 3c2672b3f35a-dpd0zv11yegl
CSeq: 1 INVITE
Max-Forwards: 70
Contact: ;flow-id=1
Replaces: pickup-3c26701519b8-5xxapzoav2u4
P-Key-Flags: keys="3"
User-Agent: snom320/7.1.39
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, 
MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 368

Then asterisk receives the pickup request:

[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use 
Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag:  Totag: 
[Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35...@rededelca
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no 
NAT)
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method 
INVITE - callid 3c2672b3f35a-dpd0zv11yegl
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis 
request - 3c2672b3f35a-dpd0zv11yegl
[...]
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: INVITE part of call transfer. 
Replaces [pickup-3c26701519b8-5xxapzoav2u4]
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use 
Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag:  Totag: 
[Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35...@rededelca
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no 
NAT)
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method 
INVITE - callid 3c2672b3f35a-dpd0zv11yegl
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis 
request - 3c2672b3f35a-dpd0zv11yegl
[...]
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: About to call Pickup(35...@pickupmark)
[Feb 11 10:44:13] DEBUG[4649] devicestate.c: Changing state for SIP/35504 - 
state 2 (In use)
[Feb 11 10:44:13] DEBUG[4649] devicestate.c: device 'SIP/35504' state '2'
[Feb 11 10:44:13] NOTICE[4659] app_directed_pickup.c: No target channel found 
for 35505.
[Feb 11 10:44:13] DEBUG[4659] channel.c: Hanging up channel 'SIP/35504-000f'
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Hangup call SIP/35504-000f, SIP 
callid 3c2672b3f35a-dpd0zv11yegl

After this obviously phone A hasn't picked up the call, and Phone B
keeps ringing.

Did I miss something in the dialplan or is it a bug?

-- 
Loris Santamaria   linux user #70506   xmpp:lo...@lgs.com.ve
Links Global Services, C.A.http://www.lgs.com.ve
Tel: 0286 952.06.87  Cel: 0414 095.00.10  sip:1...@lgs.com.ve

-O9 -omg-optimize -fomit-instructions



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Re: [asterisk-users] problems with 1.6

[Jonathan Addleman]

I'm still unable to do much with my new 1.6 installation. I just tried
reinstalling, and using the standard debian configuration files, with
just the necessary modifications, in case I had some legacy stuff in
there from earlier versions that was interfering. I'm testing in a xen
domU with debian's asterisk package, version 1:1.6.2.0-1

When I try to connect a channel into a conference, I still get all sorts
of "Unable to write to alert pipe" and then "Exceptionally long voice
queue length queuing" errors. I've tried this with meetme and
appkonference, with no difference in the error messages.

This happens with various sorts of channels - audio playing, eagi
scripts, sip connections. For example, I just tried this through the
manager interface:

Action: originate
Channel: Local/confere...@veco/n
Context: veco
Exten: playaudiofile
Priority: 1
Variable: tour=test
Variable: dir=
Variable: conference=ConferenceA
Variable: provider=teliax
Variable:
extravalue=/var/www/vecotourism/media//transcoded/audio/long-asterisk
Variable: title=Long test sound (long-asterisk.wav)


with this in extensions.conf:
exten => playaudiofile,1,Answer
exten => playaudiofile,n,Wait(1)
exten => playaudiofile,n,Playback(${extravalue})

exten => meetme,1,Answer()
exten => meetme,n,Wait(1)
exten => meetme,n,MeetMe(${conference}_${tour},1qd)

exten => conference,1,Answer()
exten => conference,n,Noop(Trying to start conference ${conference}_${tour})
exten => conference,n,konference(${conference}_${tour})



Seems this should be very straightforward, but it isn't working. What
might be wrong?
-- 
Jon-o Addleman - http://www.redowl.ca

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Re: [asterisk-users] Robotic sound sometimes

[Peder]

There is a statistics area and you can select sip or voip calls to see
calls.  It shows packet loss, jitter, latency, out of sequence packets, etc.
It can even play them back, so you can check where the loss is and play back
the call to see if the noise is in the same spot.  Here is some info from
the wireshark website:

http://wiki.wireshark.org/VoIP_calls



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Friday, February 12, 2010 4:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Robotic sound sometimes

On Fri, 12 Feb 2010, Peder wrote:

> Since it is sporadic, my guess would be network latency / packet loss 
> /jitter to ITSP.  You may have lots of capacity and they may claim to 
> have lots of capacity, but what about the links between you and them. 
> Who knows when/if there is loss and latency and jitter there.  Setup 
> wireshark to grab some calls and when someone complains about it, look 
> at the stats for that call.  It will tell you if there is loss or 
> latency or jitter.

Do you know of a link that explains how to detect latency or jitter? My 
wireshark skills are pretty non-existent. Does it have a wizzy filter that 
will tell you or do you need to check the timestamps of the RTP packets? 
Any chance you could write up your technique?

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Robotic sound sometimes

[Steve Edwards]

On Fri, 12 Feb 2010, Peder wrote:

> Since it is sporadic, my guess would be network latency / packet loss 
> /jitter to ITSP.  You may have lots of capacity and they may claim to 
> have lots of capacity, but what about the links between you and them. 
> Who knows when/if there is loss and latency and jitter there.  Setup 
> wireshark to grab some calls and when someone complains about it, look 
> at the stats for that call.  It will tell you if there is loss or 
> latency or jitter.

Do you know of a link that explains how to detect latency or jitter? My 
wireshark skills are pretty non-existent. Does it have a wizzy filter that 
will tell you or do you need to check the timestamps of the RTP packets? 
Any chance you could write up your technique?

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Robotic sound sometimes

[Peder]

Since it is sporadic, my guess would be network latency / packet loss
/jitter to ITSP.  You may have lots of capacity and they may claim to have
lots of capacity, but what about the links between you and them.  Who knows
when/if there is loss and latency and jitter there.  Setup wireshark to grab
some calls and when someone complains about it, look at the stats for that
call.  It will tell you if there is loss or latency or jitter.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Friday, February 12, 2010 4:06 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] Robotic sound sometimes

 

No, phone on LAN, through Asterisk box on LAN, through firewall, out to
fiber connection (lots of capacity there), to ITSP.

 

Codec is uLaw.

 

This only happens sometimes, so I'm wondering if it's an asterisk bug?
Aastra bug?  Network latency?  LAN capacity, etc.  Never seen this before...

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Friday, February 12, 2010 4:34 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Robotic sound sometimes

What codecs are you using? Are the calls internal(local network) only?

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105

- "Michelle Dupuis"  wrote: 
> 

We have inherited an installation with Ast 1.4 and Aastra phones.  The
client complains that sometimes the call audio turns tinny and robotic...I
heard it and it sounds wierd.

 

Has anyone else experienced this?  Cause?  Solutions?

 

Thanks,

MD


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Re: [asterisk-users] Robotic sound sometimes

[Ron Arts]

We experienced this a couple of months ago. It went away when
we upgraded the phones to the latest firmware.
Another symptom: temporarily putting the caller on hold cures
the problem sometimes.

Ron


Op 12-02-10 22:34, Tim Nelson schreef:
> What codecs are you using? Are the calls internal(local network) only?
>
> Tim Nelson
> Systems/Network Support
> Rockbochs Inc.
> (218)727-4332 x105
>
> - "Michelle Dupuis"  wrote:
>  >
> We have inherited an installation with Ast 1.4 and Aastra phones. The
> client complains that sometimes the call audio turns tinny and
> robotic...I heard it and it sounds wierd.
> Has anyone else experienced this? Cause? Solutions?
> Thanks,
> MD
>
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>


-- 
NeoNova BV
innovatieve internetoplossingen

http://www.neonova.nl  Science Park 140   1098 XG Amsterdam
info: 020-5611300  servicedesk: 020-5611302   fax: 020-5611301
KvK Amsterdam 34151241

Op dit bericht is de volgende disclaimer van toepassing:
http://www.neonova.nl/maildisclaimer

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Re: [asterisk-users] Robotic sound sometimes

[Michelle Dupuis]

No, phone on LAN, through Asterisk box on LAN, through firewall, out to
fiber connection (lots of capacity there), to ITSP.
 
Codec is uLaw.
 
This only happens sometimes, so I'm wondering if it's an asterisk bug?
Aastra bug?  Network latency?  LAN capacity, etc.  Never seen this before...

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Friday, February 12, 2010 4:34 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Robotic sound sometimes


What codecs are you using? Are the calls internal(local network) only?

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105

- "Michelle Dupuis"  wrote: 
> 
We have inherited an installation with Ast 1.4 and Aastra phones.  The
client complains that sometimes the call audio turns tinny and robotic...I
heard it and it sounds wierd.
 
Has anyone else experienced this?  Cause?  Solutions?
 
Thanks,
MD

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Re: [asterisk-users] Robotic sound sometimes

[Tim Nelson]

What codecs are you using? Are the calls internal(local network) only? 

Tim Nelson 
Systems/Network Support 
Rockbochs Inc. 
(218)727-4332 x105 

- "Michelle Dupuis"  wrote: 
> 
We have inherited an installation with Ast 1.4 and Aastra phones. The client 
complains that sometimes the call audio turns tinny and robotic...I heard it 
and it sounds wierd. 

Has anyone else experienced this? Cause? Solutions? 

Thanks, 
MD 
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Re: [asterisk-users] how to allow some extensions to make call outside and some extensions cant call outside

[Kyle Kienapfel]

internal numbers are in the internal context right?

[others]
include => internal
;extension rules for dialing out

also change your default context in sip.conf, that default context is hit by
anything incoming that doesn't match as one of your regular phones. set it
to notothers? ;)

Please keep this on list, thanks

On Fri, Feb 12, 2010 at 12:48 PM, Kyle Kienapfel wrote:

> use two contexts, one for internal numbers, and one for outside, and
> include the inside phones in the outside context.
>
>
>
> On Fri, Feb 12, 2010 at 12:39 PM, cool dude wrote:
>
>> i  had configured asterisk with a minimum dial plan, made 10 extentions.
>> below is extensions and sip.conf
>>
>> i want configure dial plan so that
>>
>> Extention 2000-2005 can receive calls from outside and make calls outside
>> and can dial all ten extentions.
>> Extention 2006-2010 can only receive calls from outside but cant call
>> outside and can dial all ten extentions
>> thx
>>
>>
>> vi /etc/asterisk/sip.conf
>>
>> [r...@localhost ~]# vi /etc/asterisk/sip.conf
>>
>> [general]
>> port = 5060
>> bindaddr = 0.0.0.0
>> context = others
>>
>> [2000]
>> type=friend
>> context=my-phones
>> secret=1234
>> host=dynamic
>>
>> [2001]
>> type=friend
>> context=my-phones
>> secret=1234
>> host=dynamic
>>
>> [2002]
>> type=friend
>> context=my-phones
>> secret=1234
>> host=dynamic
>>
>> [2003]
>> type=friend
>> contex=my-phones
>> secret=1234
>> host=dynamic
>>
>>
>> [2004]
>> type=friend
>> contex=my-phones
>> secret=1234
>> host=dynamic
>>
>> [2005]
>> type=friend
>> contex=myphones
>>
>> secret=1234
>> host=dynamic
>>
>> [2006]
>> type=friend
>> contex=my-phones
>> secret=1234
>> host=dynamic
>>
>>
>> [2007]
>> type=friend
>> contex=my-phones
>> secret=1234
>>
>> [2008]
>> type=friend
>> contex=my-phones
>> secret=1234
>> host=dynamic
>>
>>
>> [2009]
>> type=friend
>> contex=my-phones
>> secret=1234
>> host=dynamic
>>
>>
>> [2010]
>> type=friend
>> contex=my-phones
>> secret=1234
>> host=dynamic
>>
>> ##
>>  #
>>
>> vi /etc/asterisk/extentions.conf
>> [from-zaptel]
>> exten => s,1,wait(2)
>> exten => s,n,dial(sip/2000)
>> exten => s,n,dial(sip/2001)
>> exten => s,n,Playback(tt-weasels)
>>
>> [others]
>> include => my-phones
>>
>> [my-phones]
>> exten => _20XX,1,Dial(SIP/${EXTEN})
>> exten => _20XX,n,VoiceMail(${ext...@others,u)
>> exten => _20XX,n,Hangup()
>>
>> exten => 2001,1,Dial(Zap/1-1/${EXTEN})
>> exten => 2001,n,Hangup
>>
>> exten => 2002,1,Dial(Zap/1-1/${EXTEN})
>> exten => 2002,n,Hangup
>>
>> exten => 2003,1,Dial(Zap/1-1/${EXTEN})
>> exten => 2003,n,Hangup
>>
>> exten => 2004,1,Dial(Zap/1-1/${EXTEN})
>> exten => 2004,n,Hangup
>>
>> exten => 2005,1,Dial(Zap/1-1/${EXTEN})
>> exten => 2005,n,Hangup
>>
>>
>> --
>> Your Mail works best with the New Yahoo Optimized IE8. Get it 
>> NOW!
>> .
>>
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>>
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>>
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[asterisk-users] Robotic sound sometimes

[Michelle Dupuis]

We have inherited an installation with Ast 1.4 and Aastra phones.  The
client complains that sometimes the call audio turns tinny and robotic...I
heard it and it sounds wierd.
 
Has anyone else experienced this?  Cause?  Solutions?
 
Thanks,
MD
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Re: [asterisk-users] how to allow some extenstions to call outside and some extensions cant call outside

[Gergo Csibra]

Friday, February 12, 2010, 9:57:42 PM, cool wrote:

> how to allow some extenstions to call outside and some extensions
> cant call outside. i am attaching sipand extensions.conf
> thx

Put the extensions into different contexts, and create outside call
extensions only in the allowed context. Remember to create outside
calls for emergency numbers in the other context too.

-- 
Best regards,
 Gergomailto:csi...@gmail.com


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[asterisk-users] PAP2

[Tim Johnson]

I know this is slightly off topic, but I was wondering if anyone can help
with a problem getting my PAP2's to connect to Asterisk. I use a
provisioning file, and I recently re-wrote the files for each PAP2. I had
a small typo and the PAPs logged it as a corrupt file. I corrected the
file, however, Line 1 on both of the PAP2's now wont register. Line 2
works fine though. I've done the "  73738", but it wont come back.
Anyone know of a way to really wipe it's memory?

Tim


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Re: [asterisk-users] dropping line (s) for 911

[mir shahnawaz]

I got it working now. I was not including context ninioneone in default context.


On Fri, Feb 12, 2010 at 1:49 PM, mir shahnawaz  wrote:
> Hi all,
>
> I am trying to implement call dropping funtionality in asterisk for
> 911. I mean if all lines are busy and someone wants to dial 911 at
> least one line should be dropped. Here is my extensions.conf which i
> copied from internet. Could somebody help me figure out what is wrong.
> Thanks in advance.
>
>
> [globals]
> CONSOLE=Console/dsp                             ; Console interface for demo
> ;CONSOLE=DAHDI/1
> ;CONSOLE=Phone/phone0
> TRUNK=DAHDI/g0                                  ; Trunk interface
> EMERGENCY=0
> EMERGENCY_TRUNK=DAHDI/g0
> EMERGENCY_NUM=12345678 (for testing)
>
>
> [default]
>
> exten => 911,1,Goto(nineoneone,s,1)
>
>
> [nineoneone]
> exten => s,1,Set(SET_EMERG_FLAG=0)
> exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
> exten => s,n,Set(EMERGENCY=1,g)
> exten => s,n,Set(SET_EMERG_FLAG=1)
> exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
> exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY}=1]?inprogress)
> exten => s,n,SoftHangup(${EMERGENCY_TRUNK}-1)
> exten => s,n,Wait(12)
> exten => s,n,Goto(checkavail)
> exten => s,s+2(inprogress),Congestion
> exten => s,checkavail+101(notavail),Goto(trunkbusy)
> exten => h,1,GotoIf($[${SET_EMERG_FLAG}=1]?3)
> exten => h,3,Set(EMERGENCY=0,g)
>
>
> Regards
>
> Shahnawaz Mir
>

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[asterisk-users] how to allow some extenstions to call outside and some extensions cant call outside

[cool dude]

how to allow some extenstions to call outside and some extensions cant call 
outside. i am attaching sipand extensions.conf
thx



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[asterisk-users] PRI Problems with 1.6.0.10

[James Lamanna]

Hi, I have a PRI problem where it appears that my system is not
responding to SETUP messages on a channel.
It seems to be retransmitting a significant number of RELEASE messages
to clear a call that is most likely
to be long gone.
This causes a huge issue because I get a bunch of hangup cause 102s (timeout).

I'm using a TE410P (1st Gen) as my PRI card.

Has anyone seen this at all?

Thanks.

--James

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Re: [asterisk-users] how to allow some extensions to make call outside and some extensions cant call outside

[Kyle Kienapfel]

use two contexts, one for internal numbers, and one for outside, and include
the inside phones in the outside context.



On Fri, Feb 12, 2010 at 12:39 PM, cool dude wrote:

> i  had configured asterisk with a minimum dial plan, made 10 extentions.
> below is extensions and sip.conf
>
> i want configure dial plan so that
>
> Extention 2000-2005 can receive calls from outside and make calls outside
> and can dial all ten extentions.
> Extention 2006-2010 can only receive calls from outside but cant call
> outside and can dial all ten extentions
> thx
>
>
> vi /etc/asterisk/sip.conf
>
> [r...@localhost ~]# vi /etc/asterisk/sip.conf
>
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = others
>
> [2000]
> type=friend
> context=my-phones
> secret=1234
> host=dynamic
>
> [2001]
> type=friend
> context=my-phones
> secret=1234
> host=dynamic
>
> [2002]
> type=friend
> context=my-phones
> secret=1234
> host=dynamic
>
> [2003]
> type=friend
> contex=my-phones
> secret=1234
> host=dynamic
>
>
> [2004]
> type=friend
> contex=my-phones
> secret=1234
> host=dynamic
>
> [2005]
> type=friend
> contex=myphones
>
> secret=1234
> host=dynamic
>
> [2006]
> type=friend
> contex=my-phones
> secret=1234
> host=dynamic
>
>
> [2007]
> type=friend
> contex=my-phones
> secret=1234
>
> [2008]
> type=friend
> contex=my-phones
> secret=1234
> host=dynamic
>
>
> [2009]
> type=friend
> contex=my-phones
> secret=1234
> host=dynamic
>
>
> [2010]
> type=friend
> contex=my-phones
> secret=1234
> host=dynamic
>
> ##
>  #
>
> vi /etc/asterisk/extentions.conf
> [from-zaptel]
> exten => s,1,wait(2)
> exten => s,n,dial(sip/2000)
> exten => s,n,dial(sip/2001)
> exten => s,n,Playback(tt-weasels)
>
> [others]
> include => my-phones
>
> [my-phones]
> exten => _20XX,1,Dial(SIP/${EXTEN})
> exten => _20XX,n,VoiceMail(${ext...@others,u)
> exten => _20XX,n,Hangup()
>
> exten => 2001,1,Dial(Zap/1-1/${EXTEN})
> exten => 2001,n,Hangup
>
> exten => 2002,1,Dial(Zap/1-1/${EXTEN})
> exten => 2002,n,Hangup
>
> exten => 2003,1,Dial(Zap/1-1/${EXTEN})
> exten => 2003,n,Hangup
>
> exten => 2004,1,Dial(Zap/1-1/${EXTEN})
> exten => 2004,n,Hangup
>
> exten => 2005,1,Dial(Zap/1-1/${EXTEN})
> exten => 2005,n,Hangup
>
>
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[asterisk-users] dropping line (s) for 911

[mir shahnawaz]

Hi all,

I am trying to implement call dropping funtionality in asterisk for
911. I mean if all lines are busy and someone wants to dial 911 at
least one line should be dropped. Here is my extensions.conf which i
copied from internet. Could somebody help me figure out what is wrong.
Thanks in advance.


[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
TRUNK=DAHDI/g0  ; Trunk interface
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/g0
EMERGENCY_NUM=12345678 (for testing)


[default]

exten => 911,1,Goto(nineoneone,s,1)


[nineoneone]
exten => s,1,Set(SET_EMERG_FLAG=0)
exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten => s,n,Set(EMERGENCY=1,g)
exten => s,n,Set(SET_EMERG_FLAG=1)
exten => s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten => s,s+2(trunkbusy),GotoIf($[${EMERGENCY}=1]?inprogress)
exten => s,n,SoftHangup(${EMERGENCY_TRUNK}-1)
exten => s,n,Wait(12)
exten => s,n,Goto(checkavail)
exten => s,s+2(inprogress),Congestion
exten => s,checkavail+101(notavail),Goto(trunkbusy)
exten => h,1,GotoIf($[${SET_EMERG_FLAG}=1]?3)
exten => h,3,Set(EMERGENCY=0,g)


Regards

Shahnawaz Mir

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[asterisk-users] how to allow some extensions to make call outside and some extensions cant call outside

[cool dude]

i  had configured asterisk with a minimum dial plan, made 10 extentions. below 
is extensions and sip.conf

i want configure dial plan so that

Extention 2000-2005 can receive calls from outside and make calls outside and 
can dial all ten extentions.
Extention 2006-2010 can only receive calls from outside but cant call outside 
and can dial all ten extentions
thx


vi /etc/asterisk/sip.conf

[r...@localhost ~]# vi /etc/asterisk/sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = others

[2000]
type=friend
context=my-phones
secret=1234
host=dynamic

[2001]
type=friend
context=my-phones
secret=1234
host=dynamic

[2002]
type=friend
context=my-phones
secret=1234
host=dynamic

[2003]
type=friend
contex=my-phones
secret=1234
host=dynamic


[2004]
type=friend
contex=my-phones
secret=1234
host=dynamic

[2005]
type=friend
contex=myphones
secret=1234
host=dynamic

[2006]
type=friend
contex=my-phones
secret=1234
host=dynamic


[2007]
type=friend
contex=my-phones
secret=1234

[2008]
type=friend
contex=my-phones
secret=1234
host=dynamic


[2009]
type=friend
contex=my-phones
secret=1234
host=dynamic


[2010]
type=friend
contex=my-phones
secret=1234
host=dynamic

##
#

vi /etc/asterisk/extentions.conf
[from-zaptel]
exten => s,1,wait(2)
exten => s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten => s,n,Playback(tt-weasels)

[others]
include => my-phones

[my-phones]
exten => _20XX,1,Dial(SIP/${EXTEN})
exten => _20XX,n,VoiceMail(${ext...@others,u)
exten => _20XX,n,Hangup()

exten => 2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup

exten => 2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup

exten => 2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup

exten => 2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup

exten => 2005,1,Dial(Zap/1-1/${EXTEN})
exten => 2005,n,Hangup




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[asterisk-users] how to allow some extensions to make call outside and some extensions cant call outside

[asterisk]

i had configured asterisk with a minimum dial plan, made 10 extentions.
below is extensions and sip.conf

i want configure dial plan so
that

Extention 2000-2005 can receive calls from outside and make calls
outside and can dial all ten extentions.
Extention 2006-2010 can only
receive calls from outside but cant call outside and can dial all ten
extentions
thx

vi /etc/asterisk/sip.conf

[r...@localhost ~]# vi
/etc/asterisk/sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context =
others

[2000]
type=friend
context=my-phones
secret=1234
host=dynamic

[2001]
type=friend
context=my-phones
secret=1234
host=dynamic

[2002]
type=friend
context=my-phones
secret=1234
host=dynamic

[2003]
type=friend
contex=my-phones
secret=1234
host=dynamic

[2004]
type=friend
contex=my-phones
secret=1234
host=dynamic

[2005]
type=friend
contex=myphones
secret=1234
host=dynamic

[2006]
type=friend
contex=my-phones
secret=1234
host=dynamic

[2007]
type=friend
contex=my-phones
secret=1234

[2008]
type=friend
contex=my-phones
secret=1234
host=dynamic

[2009]
type=friend
contex=my-phones
secret=1234
host=dynamic

[2010]
type=friend
contex=my-phones
secret=1234
host=dynamic

##
#

vi
/etc/asterisk/extentions.conf
[from-zaptel]
exten => s,1,wait(2)
exten =>
s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten =>
s,n,Playback(tt-weasels)

[others]
include => my-phones

[my-phones]
exten
=> _20XX,1,Dial(SIP/${EXTEN})
exten =>
_20XX,n,VoiceMail(${ext...@others,u)
exten => _20XX,n,Hangup()

exten =>
2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup

exten =>
2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup

exten =>
2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup

exten =>
2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup

exten =>
2005,1,Dial(Zap/1-1/${EXTEN})
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[asterisk-users] how to allow some extensions to make call outside and some extensions cant call outside

[cool dude]

i  had configured asterisk with a minimum dial plan, made 10 extentions. below 
is extensions and sip.conf

i want configure dial plan so that

Extention 2000-2005 can receive calls from outside and make calls outside and 
can dial all ten extentions.
Extention 2006-2010 can only receive calls from outside but cant call outside 
and can dial all ten extentions
thx





#
vi /etc/asterisk/sip.conf 

[general]
port = 5060
bindaddr = 0.0.0.0
context = others 

[2000]
type=friend
context=my-phones
secret=1234
host=dynamic 

[2001]
type=friend
context=my-phones
secret=1234
host=dynamic 

[2002]
type=friend
context=my-phones
secret=1234
host=dynamic 

[2003]
type=friend
contex=my-phones
secret=1234
host=dynamic 

[2004]
type=friend
contex=my-phones
secret=1234
host=dynamic 

[2005]
type=friend
contex=my-phones
secret=1234
host=dynamic 

[2006]
type=friend
contex=my-phones
secret=1234
host=dynamic 

[2007]
type=friend
contex=my-phones
secret=1234
host=dynamic 

[2008]
type=friend
contex=my-phones
secret=1234
host=dynamic 

[2009]
type=friend
contex=my-phones
secret=1234
host=dynamic 

[2010]
type=friend
contex=my-phones
secret=1234
host=dynamic 

#

vi /etc/asterisk/extensions.conf 
[others] 
exten => 2000,1,Dial(SIP/2000)
exten => 2001,1,Dial(SIP/2001)
exten => 2002,1,Dial(SIP/2002)
exten => 2003,1,Dial(SIP/2003)
exten => 2004,1,Dial(SIP/2004)
exten => 2005,1,Dial(SIP/2005)
exten => 2006,1,Dial(SIP/2006)
exten => 2007,1,Dial(SIP/2007)
exten => 2008,1,Dial(SIP/2008)
exten => 2009,1,Dial(SIP/2009)
exten => 2010,1,Dial(SIP/2010) 

plz guide me how to achieve this


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Re: [asterisk-users] Asterisk Cepstral TTS

[mj]

Using the latest trixbox.



On Fri, Feb 12, 2010 at 1:44 PM, Steve Edwards
 wrote:
> On Fri, 12 Feb 2010, mj wrote:
>
>> I have created a call file as follows:
>>
>> Channel: IAX2/trunk/cell number
>> Application: swift
>> Data: test call
>>
>> it calls my cell then nothing...just hangs up.
>
> Sounds like you don't have app_swift.so loaded.
>
> What does your console look like?
>
> --
> Thanks in advance,
> -
> Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
> Newline                                              Fax: +1-760-731-3000
>
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Re: [asterisk-users] Asterisk Cepstral TTS

[Steve Edwards]

On Fri, 12 Feb 2010, mj wrote:

> I have created a call file as follows:
>
> Channel: IAX2/trunk/cell number
> Application: swift
> Data: test call
>
> it calls my cell then nothing...just hangs up.

Sounds like you don't have app_swift.so loaded.

What does your console look like?

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

[Tzafrir Cohen]

On Fri, Feb 12, 2010 at 06:24:02PM +, Brian wrote:
> On Fri, 2010-02-12 at 19:20 +0200, Tzafrir Cohen wrote:
> 
> > Adding documentation is what you do when things fail to work. I suspect
> > you use an older init.d script.
> > 
> You are just being silly now. Good documentation is essential to
> everything.

I prefer fixing a bug than documenting a workaround.

> 
> With regards to your comments about the init.d script, it was provided
> with 1.6.1 as an extra and worked out of the box. The change is with
> 1.6.2 where it no longer works out of the box. To me that is not
> intended operation - or a change which should be documented.
> 
> I don't agree with you as what you are saying has no legs - the fix is
> simple insofar as removing '(!)' from asterisk.conf, but it's a bit
> sloppy to expect a directory to stay persistent where it may well not be
> the case. Even by your own metric this would make 'things fail to work'
> and by your own suggestion the fix for that is to document the issue?

The pathes in the generated asterisk.conf are the built-in ones. If
removing the '(!)' solved anything, something went wrong. My guess is
that your asterisk.conf is older than your asterisk binaries.

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http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Asterisk Cepstral TTS

[mj]

I have created a call file as follows:

Channel: IAX2/trunk/cell number
Application: swift
Data: test call

it calls my cell then nothing...just hangs up.



On Fri, Feb 12, 2010 at 12:23 PM, Steve Edwards
 wrote:
> On Fri, 12 Feb 2010, mj wrote:
>
>> I already have Cepstral installed I guess I just need to figure out
>> where in the .call file and format to call cepstral and then the txt for
>> the message. Thanks again for all of your help!
>
> Set the text as a channel variable in the call file. Specify a context,
> extension, priority so that after the channel is answered you can use
> system() or swift().
>
> --
> Thanks in advance,
> -
> Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
> Newline                                              Fax: +1-760-731-3000
>
> --
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> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Asterisk Cepstral TTS

[Richard Kenner]

> where in the .call file and format to call cepstral and then the txt
> for the message. 

Application and Data, respectively.

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Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

[Brian]

On Fri, 2010-02-12 at 19:20 +0200, Tzafrir Cohen wrote:

> Adding documentation is what you do when things fail to work. I suspect
> you use an older init.d script.
> 
You are just being silly now. Good documentation is essential to
everything.

With regards to your comments about the init.d script, it was provided
with 1.6.1 as an extra and worked out of the box. The change is with
1.6.2 where it no longer works out of the box. To me that is not
intended operation - or a change which should be documented.

I don't agree with you as what you are saying has no legs - the fix is
simple insofar as removing '(!)' from asterisk.conf, but it's a bit
sloppy to expect a directory to stay persistent where it may well not be
the case. Even by your own metric this would make 'things fail to work'
and by your own suggestion the fix for that is to document the issue?

Have a pleasant weekend.


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Re: [asterisk-users] Asterisk Cepstral TTS

[Steve Edwards]

On Fri, 12 Feb 2010, mj wrote:

> I already have Cepstral installed I guess I just need to figure out 
> where in the .call file and format to call cepstral and then the txt for 
> the message. Thanks again for all of your help!

Set the text as a channel variable in the call file. Specify a context, 
extension, priority so that after the channel is answered you can use 
system() or swift().

-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Asterisk Cepstral TTS

[ABBAS SHAKEEL]

As you have done Cepstral for in bound call you have developed contexts for
it, in the similar way in .call file you can specify the context and it will
start execution of commands in the context after dialing the number.
Hope this helps

On Fri, Feb 12, 2010 at 11:01 PM, mj  wrote:

> Thanks guys...
> I already have Cepstral installed I guess I just need to figure out
> where in the .call file and format to call cepstral and then the txt
> for the message. Thanks again for all of your help!
>
>
>
> On Fri, Feb 12, 2010 at 11:50 AM, ABBAS SHAKEEL
>  wrote:
> > hello ,
> > First you check
> > out http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
> > Once you are done with auto dial out then look for Cepstral
> > TTS.
> http://www.google.com.pk/search?hl=en&safe=active&q=asterisk+with+cepstral&btnG=Search&meta=&aq=f&oq=
> > One more thing that there are other ways as well for auto dial out ..
> > AMI can also be used
> > I hope this helps
> > Kind Regards
> >
> >
> > On Fri, Feb 12, 2010 at 10:24 PM, mj  wrote:
> >>
> >> Can someone point me to a page about writing a text file to call an
> >> external number and play a TTS with cepstral? I know it includes the
> >> creation of a .call file but beyond that im a bit lost.
> >>
> >> --
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> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> > --
> > Best Regards
> > Shakeel Abbas
> >
> >
> > --
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> >
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> >
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-- 
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Shakeel Abbas
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Re: [asterisk-users] Asterisk Cepstral TTS

[mj]

Thanks guys...
I already have Cepstral installed I guess I just need to figure out
where in the .call file and format to call cepstral and then the txt
for the message. Thanks again for all of your help!



On Fri, Feb 12, 2010 at 11:50 AM, ABBAS SHAKEEL
 wrote:
> hello ,
> First you check
> out http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
> Once you are done with auto dial out then look for Cepstral
> TTS. http://www.google.com.pk/search?hl=en&safe=active&q=asterisk+with+cepstral&btnG=Search&meta=&aq=f&oq=
> One more thing that there are other ways as well for auto dial out ..
> AMI can also be used
> I hope this helps
> Kind Regards
>
>
> On Fri, Feb 12, 2010 at 10:24 PM, mj  wrote:
>>
>> Can someone point me to a page about writing a text file to call an
>> external number and play a TTS with cepstral? I know it includes the
>> creation of a .call file but beyond that im a bit lost.
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> Best Regards
> Shakeel Abbas
>
>
> --
> _
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>
> asterisk-users mailing list
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Re: [asterisk-users] Asterisk Cepstral TTS

[Jeff Grollo]

On
&],-
By
On Feb 12, 2010, at 12:45 PM, Steve Edwards  
 wrote:

> On Fri, 12 Feb 2010, mj wrote:
>
>> Can someone point me to a page about writing a text file to call an
>> external number
>
>http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
>
>> and play a TTS with cepstral? I know it includes the creation of  
>> a .call
>> file but beyond that im a bit lost.
>
> Using Cepstral's command line utility:
>
>swift\
>-n Allison\
>-o example.wav\
>-p audio/sampling-rate=8000\
>- "Pay up deadbeat."
>
> or app_swift?
>
>http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Swift
>
> -- 
> Thanks in advance,
> --- 
> --
> Steve Edwards   sedwa...@sedwards.com  Voice:  
> +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
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Re: [asterisk-users] Asterisk Cepstral TTS

[ABBAS SHAKEEL]

hello ,

First you check out
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out

Once you are done with auto dial out then look for Cepstral TTS.
http://www.google.com.pk/search?hl=en&safe=active&q=asterisk+with+cepstral&btnG=Search&meta=&aq=f&oq=

One more thing that there are other ways as well for auto dial out ..
AMI can also be used

I hope this helps

Kind Regards



On Fri, Feb 12, 2010 at 10:24 PM, mj  wrote:

> Can someone point me to a page about writing a text file to call an
> external number and play a TTS with cepstral? I know it includes the
> creation of a .call file but beyond that im a bit lost.
>
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Re: [asterisk-users] Asterisk Cepstral TTS

[Steve Edwards]

On Fri, 12 Feb 2010, mj wrote:

> Can someone point me to a page about writing a text file to call an 
> external number

http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out

> and play a TTS with cepstral? I know it includes the creation of a .call 
> file but beyond that im a bit lost.

Using Cepstral's command line utility:

swift\
-n Allison\
-o example.wav\
-p audio/sampling-rate=8000\
- "Pay up deadbeat."

or app_swift?

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Swift

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[asterisk-users] Asterisk Cepstral TTS

[mj]

Can someone point me to a page about writing a text file to call an
external number and play a TTS with cepstral? I know it includes the
creation of a .call file but beyond that im a bit lost.

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Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

[Tzafrir Cohen]

On Fri, Feb 12, 2010 at 04:51:29PM +, Brian wrote:
> On Fri, 2010-02-12 at 10:20 -0600, Kevin P. Fleming wrote:
> > Brian wrote:
> > 
> > > If you leave the defaults as they are (dictated by (!) in asterisk.conf)
> > > then the behaviour you get is Asterisk looking to /var/run/asterisk
> > > which the OS deletes on reboot. The question is why is this the default
> > > behaviour when it breaks systems that clear /var/run
> > 
> > Because it's impossible to have defaults that work on every possible
> > system that might do anything the system installer can come up with. 
> Quite - but the core as it is expect /var/run to be persistent and on
> several flavours of Linux it is not. 

Asterisk is not the only service with this problem in Ubuntu.

Check others under /etc/init.d .

> It's fair to say that /var/run will
> probably be available in most cases so could the default behaviour not
> be to look there rather than /var/run/asterisk which may not?

So you say it should be fixed in the init.d script?

As in
http://svnview.digium.com/svn/asterisk?view=revision&revision=177852

> 
> I don't want to fight about it and I appreciate your defence of the
> current status quo - it's no big deal. I'm more than happy to admit I
> did not relaize that /var/run was emptied on Debian based distros on
> boot. Perhaps you could pass that information up to whoever builds the
> makefile -or add a little into the README as it confuses new users like
> me - who just want it to work.

Adding documentation is what you do when things fail to work. I suspect
you use an older init.d script.

Alternatively, you can use the simpler upstart conf file:
http://svnview.digium.com/svn/asterisk/trunk/contrib/upstart/asterisk.user.conf?view=markup

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Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

[Brian]

On Fri, 2010-02-12 at 10:20 -0600, Kevin P. Fleming wrote:
> Brian wrote:
> 
> > If you leave the defaults as they are (dictated by (!) in asterisk.conf)
> > then the behaviour you get is Asterisk looking to /var/run/asterisk
> > which the OS deletes on reboot. The question is why is this the default
> > behaviour when it breaks systems that clear /var/run
> 
> Because it's impossible to have defaults that work on every possible
> system that might do anything the system installer can come up with. 
Quite - but the core as it is expect /var/run to be persistent and on
several flavours of Linux it is not. It's fair to say that /var/run will
probably be available in most cases so could the default behaviour not
be to look there rather than /var/run/asterisk which may not?

I don't want to fight about it and I appreciate your defence of the
current status quo - it's no big deal. I'm more than happy to admit I
did not relaize that /var/run was emptied on Debian based distros on
boot. Perhaps you could pass that information up to whoever builds the
makefile -or add a little into the README as it confuses new users like
me - who just want it to work.

> 
> -- 
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kpflem...@digium.com
> Check us out at www.digium.com & www.asterisk.org
> 


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Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

[Kevin P. Fleming]

Brian wrote:

> If you leave the defaults as they are (dictated by (!) in asterisk.conf)
> then the behaviour you get is Asterisk looking to /var/run/asterisk
> which the OS deletes on reboot. The question is why is this the default
> behaviour when it breaks systems that clear /var/run

Because it's impossible to have defaults that work on every possible
system that might do anything the system installer can come up with. It
has only been very recently that any distros at all decided to empty
/var/run on boot, and until now I haven't seen anyone comment about it
or do anything to get Asterisk to be able to accommodate that.

With that said, though, in general Asterisk does not *ever* create any
of the directories it is told to use in /etc/asterisk.conf (or the
defaults in the code, if that file is not present or does not override
them), and this is generally the behavior of most service applications
like Asterisk.

-- 
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Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
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[asterisk-users] T.38 with reinvite

[Deepesh D]

Hello,

Is it possible to use asterisk in T.38 pass through mode with reinvite?

My fax calls are getting disconnected if canreinvite=yes. It works
only if I make canreinvite=no. Normal calls work in both cases.


Thanks

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Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

[Brian]

On Fri, 2010-02-12 at 15:18 +0200, Tzafrir Cohen wrote:
> On Fri, Feb 12, 2010 at 12:28:53PM +, Brian wrote:
> > On Fri, 2010-02-12 at 14:15 +0200, Tzafrir Cohen wrote:
> > {snip}
> > > So the questions to ask are, I believe:
> > > 
> > > * Should asterisk here be run as root? If so: why?
> > It suits me to run it that way on the device concerned.
> 
> Could you please be more specific?
Is it relevant at all to the problem or bug?
> 
> > > * Where should the astvarrundir be?
> > If you leave the defaults as they are (dictated by (!) in asterisk.conf)
> > then the behaviour you get is Asterisk looking to /var/run/asterisk
> > which the OS deletes on reboot. The question is why is this the default
> > behaviour when it breaks systems that clear /var/run
> 
> Because we should make it simple to run Asterisk as non-root.
Is this relevant at all to the issue?
> 
> > 
> > This was - btw - a compile from source, not packaged offering.
> 
> Right. But packagers have a way of running into platform-specific bugs.
The bug is simple and clear. Default build expects /var/run/asterisk to
be persistent. This has no relevance to the user that Asterisk is being
run as. 


Forgive me for being blunt, but if you want to argue politics or fight
about root -v- non privileged that is your prerogative, but please pick
on somebody else - I'm not interested. The simple test is this - would
running it as some other user avoid this problem - answer: no.

Kind regards
Brian.


> 
> -- 
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> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
> 


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Re: [asterisk-users] SIP RTP ports not released when channel is hung up

[Klaus Darilion]

Am 11.02.2010 21:09, schrieb Olle E. Johansson:
>
> 11 feb 2010 kl. 13.30 skrev Klaus Darilion:
>
>> Am 11.02.2010 11:21, schrieb Armin Schindler:
>>> Hello,
>>>
>>> using Asterisk 1.4.28, I encountered a problem with SIP
>>> RTP port allocation.
>>>
>>> I found some entries in mailinglist and bugtracker regarding
>>> this issue, but only old ones.
>>>
>>> My rtp.conf has
>>>[general]
>>>rtpstart=3
>>>rtpend=30100
>>>
>>> so 100 ports available. I know that up to 4 ports per channel can be used
>>> and so up to 25 channels are possible.
> 4 ports only if you use audio and video. We use two ports per RTP stream - 
> and send on two ports, but this is for incoming media.
> So 100 ports is enough for 50 audio calls.
>
>>> But even earlier I often get the error about "No RTP ports remaining".
>>>
>>> I had a look at
>>>netstat -nuap
>>> and it shows that a lot of ports are still assigned, even if there is no
>>> channel in use.
>>> But "sip show channels" show a lot of (unused) entries with no
>>> codec/Format and "Last Message" like INVITE, REGISTER, OPTIONS.
> REGISTER and OPTIONS allocate no RTP ports, so those are not a problem. If 
> you have a SIP channel that has a last message being INVITE and still say you 
> have no calls, you have a problem right there.
>>
>> If the channels exists even after 32 seconds after BYE, and BYE was
>> signaled correctly, I would file a bug report.
>
> Yes, the RTP ports should be closed at least at that point, when we destroy 
> the SIP channel. Anything else is a bug. I am not really sure about when 
> they're closed, but I'm trying to understand that in my RTCP adventures since 
> I want to change it.
>
> While we are discussing this, I would like some feedback.
>
> If we receive RTCP bye from the other end, we can close the port at that 
> point.
> When we hang up the call, we send RTCP BYE and a final RTCP report.
>
> If we don't receive the RTCP BYE or a final report - I would like to keep the 
> RTCP port open a bit longer - but at maximum up to the destruction of the SIP 
> channel - so I can have a chance of receiving a final RTCP report from the 
> other end or/and RTCP BYE.
>
> What do you think?

Will the channel only be kept alive in chan_sip or also in the core? 
Somehow we need a method to export the data received in the final reply, 
otherwise it makes no sense to wait.

klaus

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Re: [asterisk-users] parked calls

[Doug Lytle]

hin lee wrote:
> Using FreePBX, is there a way to play a beep sound when you are 
> connected to a parked call? Right now, it's dead silence and we can't 
> tell if the call has been connected.
>
I don't know about FreePBX, but under the features.conf, there is:

courtesytone = local/stutter; Sound file to play to the parked caller
 ; when someone dials a parked call

I recorded our Definity's stutter tone and put it in the local folder.

Doug


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Re: [asterisk-users] app_dial.c: Unable to create channel oftype 'Zap' (cause 34 - Circuit/channel congestion)

[Mariano Lecuona]

As far as my experience, this problem occurs when the asterisk tries to take
a new channel and teco does not count with any available channels.
Contact your E1/T1 provider and work with them to search on the teco side.


2010/2/12 Tzafrir Cohen 

> On Fri, Feb 12, 2010 at 11:15:17AM +1100, Lee, John (Sydney) wrote:
> >
> > > What is the output of 'cat /proc/dahdi/1' ?
> > I did not record it but it just shows every channel as 'red alarm'.
>
> How many channels?
>
> E1 or T1?
>
> >
> > > What do you have in /etc/zaptel.conf ?
> > loadzone=au
> > defaultzone=au
> > #
> > # For OnRamp 10
> > #
> > span=1,1,0,ccs,hdb3,crc4
> > bchan=1-10
> > unused=11-15,17-31
> > dchan=16
> > #
> > # Rhino 24-port Channel Bank
> > #
> > span=2,0,0,esf,b8zs
> > fxols=32-55
>
> So span 1 is E1 and span 2 is T1. Are you sure things weren't confused
> somehow?
>
> --
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[asterisk-users] parked calls

[hin lee]

Using FreePBX, is there a way to play a beep sound when you are connected to a 
parked
call? Right now, it's dead silence and we can't tell if the call has
been connected.


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Re: [asterisk-users] billsec is set to duration if call is not answered

[Frank Church]

I have still not been able to resolve this problem. It only happens
when the call is routed through an agi script.

Whether the dialling itself is done within the AGI, or the AGI drops
into the dialplan to allow the call to originate there, the result is
the same.

Call is marked as answered and billsec is set to duration, or duration - 1.

Is that behaviour that has been observed before?

On 8 February 2010 17:00, Frank Church  wrote:
> The behaviour of my Asterisk appears to have changed suddenly without
> any apparent cause.
> The version is use is 1.4.27.1
>
> When a call is not answered billsec is set to duration, and calls are
> charged. I can't see any change I could have
> made to cause this problem. Is it something already known in Asterisk?
>
> /voipfc
>

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Re: [asterisk-users] SIP tunnel

[Scott L. Lykens]

> My idea is to use a well know port like port 80 (that is not blocked).
Skype for example uses this port.

If you are in a situation where the ISP/government is blocking VoIP you
are probably going to have to encrypt it to get it through, and that may
not even work. I have a client who has facilities in Belize where BTL
apparently employs quite sophisticated deep packet inspection... SIP or
IAX on any port combination would drop about half a second after the
media starts. IPSec over UDP/IKE were completely blocked as well. I
ended up using IPSEC over TCP as it was not interfered with.

If the ISP or government are not the problem, only firewalls... IIRC in
a typical NAT setup you could have the client register to you using IAX
- This will keep the port open through the NAT device so you can send
calls to them without them having to map ports in their firewall.

sl

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Re: [asterisk-users] app_dial.c: Unable to create channel oftype 'Zap' (cause 34 - Circuit/channel congestion)

[Tzafrir Cohen]

On Fri, Feb 12, 2010 at 11:15:17AM +1100, Lee, John (Sydney) wrote:
> 
> > What is the output of 'cat /proc/dahdi/1' ?
> I did not record it but it just shows every channel as 'red alarm'.

How many channels?

E1 or T1?

> 
> > What do you have in /etc/zaptel.conf ?
> loadzone=au
> defaultzone=au
> #
> # For OnRamp 10
> #
> span=1,1,0,ccs,hdb3,crc4
> bchan=1-10
> unused=11-15,17-31
> dchan=16
> #
> # Rhino 24-port Channel Bank
> #
> span=2,0,0,esf,b8zs
> fxols=32-55

So span 1 is E1 and span 2 is T1. Are you sure things weren't confused
somehow?

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Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

[Tzafrir Cohen]

On Fri, Feb 12, 2010 at 12:28:53PM +, Brian wrote:
> On Fri, 2010-02-12 at 14:15 +0200, Tzafrir Cohen wrote:
> {snip}
> > So the questions to ask are, I believe:
> > 
> > * Should asterisk here be run as root? If so: why?
> It suits me to run it that way on the device concerned.

Could you please be more specific?

> > * Where should the astvarrundir be?
> If you leave the defaults as they are (dictated by (!) in asterisk.conf)
> then the behaviour you get is Asterisk looking to /var/run/asterisk
> which the OS deletes on reboot. The question is why is this the default
> behaviour when it breaks systems that clear /var/run

Because we should make it simple to run Asterisk as non-root.

> 
> This was - btw - a compile from source, not packaged offering.

Right. But packagers have a way of running into platform-specific bugs.

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Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

[Brian]

On Fri, 2010-02-12 at 14:15 +0200, Tzafrir Cohen wrote:
{snip}
> So the questions to ask are, I believe:
> 
> * Should asterisk here be run as root? If so: why?
It suits me to run it that way on the device concerned.
> * Where should the astvarrundir be?
If you leave the defaults as they are (dictated by (!) in asterisk.conf)
then the behaviour you get is Asterisk looking to /var/run/asterisk
which the OS deletes on reboot. The question is why is this the default
behaviour when it breaks systems that clear /var/run

This was - btw - a compile from source, not packaged offering.
> 
> -- 
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> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
> 
I disagree. 


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Re: [asterisk-users] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory

[Tzafrir Cohen]

On Thu, Feb 11, 2010 at 07:36:23AM -0500, Leif Madsen wrote:
> Jason Parker wrote:
> > Brian wrote:
> >> Each time the server is rebooted Asterisk duly
> >> deletes the manually created /var/run/asterisk directory - quite why it
> >> does this I just don't know - perhaps it is a bug?
> > 
> > Your assumption is incorrect.  Some Linux distributions will empty 
> > /var/run/ on 
> > boot, just as they do with /tmp/.  I do believe you're right, however, in 
> > suggesting that there is a bug in Asterisk.  It appears that Asterisk 
> > creates 
> > /var/run/asterisk/ during install and assumes that it will always exist.
> > 
> > Some of the sample init scripts (Debian) create that directory before 
> > starting 
> > Asterisk.  This should be done in all of them (or in Asterisk itself, 
> > maybe?).
> > 
> > Please report an issue on http://issues.asterisk.org/
> 
> For future reference, the issue reported is #16802.
> 
> https://issues.asterisk.org/view.php?id=16802

The report in the bug is not clear.

If you run Asterisk as non-root, asterisk should have write permissions
to the astvarrun directory. Thus it should be the subdirectory
/var/run/asterisk .

Packages on Debian has long ago defaulted to even prevent running as
root, and had astvarrundir set to /var/run/asterisk as a compile-time
default for quite some time.

The Ubuntu packages, based mostly on the Debian ones, had to face the
fact that Ubuntu deletes everything under /var/run by default at boot.
Thus the init.d script was fixed to (re)create that directory.

This fix made it eventually into the Debian packages, and also
(eventually, and independently) into the standard "debian" asterisk
init.d script.

So the questions to ask are, I believe:

* Should asterisk here be run as root? If so: why?
* Where should the astvarrundir be?

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Re: [asterisk-users] Can an agent Login to a queue and be paused

[Lenz Emilitri]

In this case, I suggest you modify the login script so that your agents
always start paused. It should be trivial to do.
l.




2010/2/8 Robert Grignon 

>  Not a bad idea... We use queuemetrics and the login is done via Web GUI.
> I could easily just send it to pause upon login...
>


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Re: [asterisk-users] Asterisk -> SIP-ROUTER -> Internet = no audio

[Yves Arikoglu]

thanks brian,

yes, i am aware that sip is only responsible for signalling and therefor 
my conclusion was, that it
has got something to do with nat / firewall / the router...
meanwhile i´ve got it solved... although the sip-provider tried to 
convince me, that the misconfiguration
is on my asterisks´ side, i penetrated the support until they looked 
over it again and... what should i
say... finally they had to admit, that the router had a wrong acesslist. 
they corrected it and now it works.

yves

Brian schrieb:
> On Fri, 2010-02-12 at 02:18 +0100, Yves Arikoglu wrote:
>   
>> Hi,
>>
>> I am breaking my fingers in configuring an asterisk (1.6) to 
>> successfully transmit audio with the following setup:
>>
>> asterisk, resides in local network, ip is 10.26.208.252
>> versatel business router (directly connected to a dsl, configured by 
>> sip-provider), WAN ip 89.244.13.25
>> versatel sip-proxy ip 89.244.13.10
>>
>>
>> in sip.conf I have:
>> [general]
>> bindaddr=0.0.0.0
>> externip=89.244.13.25
>> localnet=10.26.208.0/255.255.252.0
>> nat=yes
>> qualify=yes
>>
>>
>> the local sip phones register correctly and can make calls between each 
>> other with audio.
>> the local sip phones CAN make outbound calls via the sip-provider... 
>> will say, destination phone rings, but there is no audio (on both legs)
>> after pickup...
>> external phones can call my sip-number... the call comes into the 
>> asterisk, the sip-extension rings, but after pickup... no audio at all.
>> even if i route the call from external to a queue or something else... i 
>> see, that asterisk is playing voicefiles, but the caller does not hear
>> anything.
>> because sip-signalling works in any ways, but audio not, i think its got 
>> something to do with nat... but there is no firewall between asterisk
>> and the router or between the router and the internetconnection from 
>> versatel... and i already tried millions of combinations of using
>> nat=yes/no/route, qualify=yes/no, canreinvite=yes/no and and and and i´m 
>> stuck as i was never ever stuck before :-(
>>
>> any hints? anybody?
>>
>> 
> You are aware that SIP only sets up, monitors and takes the call down?
> The audio stream is RDP and on higher ports. My guess is that the audio
> stream on inbound calls is not arriving where it should be - or is
> blocked. This could be router or nat, but one thing jumps out to me:
> Does your Asterisk Server itself have something set up in the built in
> iptables firewall blocking udp inbound traffic in the port range
> 15000:2? The output of the command 'iptables -nvL' will tell you
> pretty quickly.
>
> HTH.
>
>
>
>   


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Re: [asterisk-users] SIP RTP ports not released when channel is hung up

[Armin Schindler]

On Fri, 12 Feb 2010, Armin Schindler wrote:
 I had a look at
   netstat -nuap
 and it shows that a lot of ports are still assigned, even if there is no
 channel in use.
 But "sip show channels" show a lot of (unused) entries with no
 codec/Format and "Last Message" like INVITE, REGISTER, OPTIONS.
>> REGISTER and OPTIONS allocate no RTP ports, so those are not a problem. If
>> you have a SIP channel that has a last message being INVITE and still say
>> you have no calls, you have a problem right there.
>
> I just see these entries with "sip show channels", but cannot tell if
> e.g. the REGISTER listed channels have RTP ports allocated.
> Who can I find out which SIP channel allocated which port?
> Or which SIP channel belongs to the ports I see with 'netstat -nuap'?

I just made a test to confirm:
After a restart of asterisk (to have a clean state with no sip channels 
activ and no RTP port allocated), I can confirm that:
- REGISTER and OPTION listed sip channels don't use RTP ports
- after some calls (e.g. SIP to SIP) the RTP ports are freed immediately
   (looks like this is the case on hangup before answer).
- after some other calls, the RTP ports are freed after about 20-30 seconds
   after hangup.
So basically all is correct.

> I do have a sip channels like
>  172.21.4.1146660430c3a638e  00102/0  0x0 (nothing)No   Init: 
> INVITE
> in 'sip show channels' and they don't go away for a long time.
> Shouldn't there be a timeout to destroy such a channel even if somehow
> the phone was 'disconnected' in during a call?
>
>>> If the channels exists even after 32 seconds after BYE, and BYE was
>>> signaled correctly, I would file a bug report.

It really looks like that there is a case where the sip channel is not
destroyed and that is the cause of the problem.
I will try to reproduce this.
Any ideas?

Armin


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Re: [asterisk-users] SIP RTP ports not released when channel is hung up

[Armin Schindler]

>>> using Asterisk 1.4.28, I encountered a problem with SIP
>>> RTP port allocation.
>>>
>>> I found some entries in mailinglist and bugtracker regarding
>>> this issue, but only old ones.
>>>
>>> My rtp.conf has
>>>   [general]
>>>   rtpstart=3
>>>   rtpend=30100
>>>
>>> so 100 ports available. I know that up to 4 ports per channel can be used
>>> and so up to 25 channels are possible.
> 4 ports only if you use audio and video. We use two ports per RTP stream - 
> and send on two ports, but this is for incoming media.
> So 100 ports is enough for 50 audio calls.

Even if it isn't a video call, I think as soon as videosupport is activated,
the additional 2 ports are allocated.

>>> But even earlier I often get the error about "No RTP ports remaining".
>>>
>>> I had a look at
>>>   netstat -nuap
>>> and it shows that a lot of ports are still assigned, even if there is no
>>> channel in use.
>>> But "sip show channels" show a lot of (unused) entries with no
>>> codec/Format and "Last Message" like INVITE, REGISTER, OPTIONS.
> REGISTER and OPTIONS allocate no RTP ports, so those are not a problem. If 
> you have a SIP channel that has a last message being INVITE and still say 
> you have no calls, you have a problem right there.

I just see these entries with "sip show channels", but cannot tell if
e.g. the REGISTER listed channels have RTP ports allocated.
Who can I find out which SIP channel allocated which port?
Or which SIP channel belongs to the ports I see with 'netstat -nuap'?

I do have a sip channels like
  172.21.4.1146660430c3a638e  00102/0  0x0 (nothing)No   Init: 
INVITE
in 'sip show channels' and they don't go away for a long time.
Shouldn't there be a timeout to destroy such a channel even if somehow
the phone was 'disconnected' in during a call?

>> If the channels exists even after 32 seconds after BYE, and BYE was
>> signaled correctly, I would file a bug report.
>
> Yes, the RTP ports should be closed at least at that point, when we destroy 
> the SIP channel. Anything else is a bug. I am not really sure about when 
> they're closed, but I'm trying to understand that in my RTCP adventures 
> since I want to change it.

Before filing a bug, I would like to be sure that I have checked all 
possibilities here.

To me it looks like that some special event leaves a sip channel activ and 
not be destroyed. So when Asterisk runs for a longer time, more and more 
channels like this occur.
Ayn idea how to check this?

Armin


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Re: [asterisk-users] Asterisk -> SIP-ROUTER -> Internet = no audio

[Brian]

On Fri, 2010-02-12 at 02:18 +0100, Yves Arikoglu wrote:
> Hi,
> 
> I am breaking my fingers in configuring an asterisk (1.6) to 
> successfully transmit audio with the following setup:
> 
> asterisk, resides in local network, ip is 10.26.208.252
> versatel business router (directly connected to a dsl, configured by 
> sip-provider), WAN ip 89.244.13.25
> versatel sip-proxy ip 89.244.13.10
> 
> 
> in sip.conf I have:
> [general]
> bindaddr=0.0.0.0
> externip=89.244.13.25
> localnet=10.26.208.0/255.255.252.0
> nat=yes
> qualify=yes
> 
> 
> the local sip phones register correctly and can make calls between each 
> other with audio.
> the local sip phones CAN make outbound calls via the sip-provider... 
> will say, destination phone rings, but there is no audio (on both legs)
> after pickup...
> external phones can call my sip-number... the call comes into the 
> asterisk, the sip-extension rings, but after pickup... no audio at all.
> even if i route the call from external to a queue or something else... i 
> see, that asterisk is playing voicefiles, but the caller does not hear
> anything.
> because sip-signalling works in any ways, but audio not, i think its got 
> something to do with nat... but there is no firewall between asterisk
> and the router or between the router and the internetconnection from 
> versatel... and i already tried millions of combinations of using
> nat=yes/no/route, qualify=yes/no, canreinvite=yes/no and and and and i´m 
> stuck as i was never ever stuck before :-(
> 
> any hints? anybody?
> 
You are aware that SIP only sets up, monitors and takes the call down?
The audio stream is RDP and on higher ports. My guess is that the audio
stream on inbound calls is not arriving where it should be - or is
blocked. This could be router or nat, but one thing jumps out to me:
Does your Asterisk Server itself have something set up in the built in
iptables firewall blocking udp inbound traffic in the port range
15000:2? The output of the command 'iptables -nvL' will tell you
pretty quickly.

HTH.



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Re: [asterisk-users] [Fwd: SIP tunnel]

[mosbah.abdelkader]

Thank you very much. This is a good tip. I will see openvpn.

Please have a look at the scientific miracles of CORAN: http://www.55a.net/

Thank you again.

On Thu, Feb 11, 2010 at 10:58 PM, Hans Witvliet  wrote:

>  Forwarded Message 
> > From: mosbah.abdelkader 
> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> > 
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] SIP tunnel
> > Date: Thu, 11 Feb 2010 14:37:24 +0100
> >
> > Hello,
>
> > I have the following situation: A firewall is blocking all SIP and RTP
> > traffic in the side of some of my clients. My clients cannot change
> > settings of the firewall.
> > I need to solve this problem and I need some help from you.
> > I have this idea: implement a SIP user agent which does not use well
> > known SIP ports (uses http port 80 for example) and use other ports
> > that are not blocked by the firewall for RTP (FTP, https,
> > ssh, ...ports). Then, configure Asterisk to use the same ports to
> > interact with the client.
> > Is this idea feasible? if not what are the problems? please give me
> > your opinions about the situation?
>
> > _
>
> I would rather suggest to build a ssl-tunnel between those locations.
> Have a look at openvpn,..
> Firewalls seldom block port-443 as it is also used for https...
>



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