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Today's Topics:
1. beeping noise in background (Kim Jones)
2. RE: What's up with the Manager Interface?!?! (Douglas Garstang)
3. g726 voice prompts (Eric Bishop)
4. Re: What's up with the Manager Interface?!?! (Tony Mountifield)
5. RE: What's up with the Manager Interface?!?! (Douglas Garstang)
6. Cisco 7940 Firmware 8.2 (James R. Stevens)
7. Re: voicemail.conf locking problem (Michiel van Baak)
8. Re: What's up with the Manager Interface?!?! (Richard Lyman)
9. Call recording with Asterisk BE (Ed Nu?ez)
10. Re: voicemail.conf locking problem (Tzafrir Cohen)
11. Re: How to park calls on a specific extension (Steve Sobol)
12. RE: What's up with the Manager Interface?!?! (Douglas Garstang)
13. Call dropping (Ed Nu?ez)
14. Re: How to park calls on a specific extension (Steve Sobol)
15. Re: SIP Port 5060 (Joseph)
16. RE: What's up with the Manager Interface?!?! (Douglas Garstang)
17. Setting RTP ports for Asterisk? (Vincent Delporte)
18. Re: Re: What's up with the Manager Interface?!?! (Richard Lyman)
19. Re: What's up with the Manager Interface?!?! (Richard Lyman)
20. Re: How to park calls on a specific extension (Ira)
21. Re: SIP Port 5060 (Andrew Joakimsen)
22. Re: Siemens Gigaset C450 IP vs S450 IP (Andrew Joakimsen)
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Message: 1
Date: Wed, 29 Nov 2006 16:06:57 -0600
From: "Kim Jones" <[EMAIL PROTECTED]>
Subject: [asterisk-users] beeping noise in background
To: <[email protected]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
I have asterisk 1.2.12.1 running with several client phone options. Our
echo cancellation is finally working great. The only problem I seem to
be having is there is background noise including beeping sounds at
regular intervals no matter which phone we use. Does anyone know why?
We are using a diqium tdm card.
Thanks
Kim
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Message: 2
Date: Wed, 29 Nov 2006 15:16:45 -0700
From: "Douglas Garstang" <[EMAIL PROTECTED]>
Subject: RE: [asterisk-users] What's up with the Manager Interface?!?!
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
Message-ID:
<[EMAIL PROTECTED]>
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-----Original Message-----
From: Steve Edwards [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 29, 2006 2:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] What's up with the Manager Interface?!?!
On Wed, 29 Nov 2006, Douglas Garstang wrote:
Grrrr. Here's another example...
Action: Command
Command: sip show peer 2944093
Response: Follows
Privilege: Command
* Name : 2944093
Secret : <Set>
MD5Secret : <Not set>
Context : 180o_CallStart
Subscr.Cont. : 180o_WatchBLF
Why the HELL is there an asterisk before 'Name'? Now I have
to strip the bloody thing out!
And why is there TWO empty lines before it?
Good grief!
Doug.
Would it be a better use of your time to "fix" the offending modules
rather than kludge your code to handle the inconsistencies?
Is AMI spec'd or would that be the first step?
Steve,
No... I'm not a C programmer. A standard interface would be a first step. :)
Doug.
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Message: 3
Date: Thu, 30 Nov 2006 09:19:13 +1100
From: "Eric Bishop" <[EMAIL PROTECTED]>
Subject: [asterisk-users] g726 voice prompts
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[email protected]>
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"
Anyone know if it posible to make voice promps native g726 or g711 format?
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Message: 4
Date: Wed, 29 Nov 2006 22:19:16 +0000 (UTC)
From: [EMAIL PROTECTED] (Tony Mountifield)
Subject: [asterisk-users] Re: What's up with the Manager Interface?!?!
To: [email protected]
Message-ID: <[EMAIL PROTECTED]>
In article <[EMAIL PROTECTED]>,
Richard Lyman <[EMAIL PROTECTED]> wrote:
just wait till you get a 'hiccup' that causes a line to get cut off,
drop a char, and continue on next line. <G>
(examples below)
I've made heavy use of the Manager interface for over 2 years now, and
have never seen the kind of behaviour you described and showed examples
of. I would be more inclined to suspect the functions you are using to
read and collect the AMI output. Perhaps there's a buffer boundary
error or something.
Cheers
Tony
I have asterisk 1.2.13 running with several client phone options. I have a problem in configuring in asterisk. I configure asterisk meetme.conf and extension.conf, but when i transfer call to conference it give me this message and asterisk
kill it self.
Ouch ... error while writing audio data: : Broken pipe
If any one knows about this please help me to fix this.
Cheers
Ishanka
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