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<a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-request@lists.digium.com">asterisk-users-request@lists.digium.com</a> wrote:
<blockquote cite="mid20061130014842.C60F72FC4BB@lists.digium.com"
type="cite">
<pre wrap="">Send asterisk-users mailing list submissions to
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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" <a class="moz-txt-link-rfc2396E" href="mailto:kim@anmsinc.com"><kim@anmsinc.com></a>
Subject: [asterisk-users] beeping noise in background
To: <a class="moz-txt-link-rfc2396E" href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></a>
Message-ID: <a class="moz-txt-link-rfc2396E" href="mailto:000401c71402$aff83d40$052110ac@ANMS.pvt"><000401c71402$aff83d40$052110ac@ANMS.pvt></a>
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" <a class="moz-txt-link-rfc2396E" href="mailto:dgarstang@oneeighty.com"><dgarstang@oneeighty.com></a>
Subject: RE: [asterisk-users] What's up with the Manager Interface?!?!
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<a class="moz-txt-link-rfc2396E" href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></a>
Message-ID:
<a class="moz-txt-link-rfc2396E" href="mailto:645FEC31A18FE54A8721500CDD55A7B6035D0C0D@mail.oneeighty.com"><645FEC31A18FE54A8721500CDD55A7B6035D0C0D@mail.oneeighty.com></a>
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</pre>
<blockquote type="cite">
<pre wrap="">-----Original Message-----
From: Steve Edwards [<a class="moz-txt-link-freetext" href="mailto:asterisk.org@sedwards.com">mailto:asterisk.org@sedwards.com</a>]
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:
</pre>
<blockquote type="cite">
<pre wrap="">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
</pre>
</blockquote>
<pre wrap="">to strip the bloody thing out!
</pre>
<blockquote type="cite">
<pre wrap="">And why is there TWO empty lines before it?
Good grief!
Doug.
</pre>
</blockquote>
<pre wrap="">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?
</pre>
</blockquote>
<pre wrap=""><!---->
Steve,
No... I'm not a C programmer. A standard interface would be a first step. :)
Doug.
------------------------------
Message: 3
Date: Thu, 30 Nov 2006 09:19:13 +1100
From: "Eric Bishop" <a class="moz-txt-link-rfc2396E" href="mailto:asterisk.eric@gmail.com"><asterisk.eric@gmail.com></a>
Subject: [asterisk-users] g726 voice prompts
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<a class="moz-txt-link-rfc2396E" href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></a>
Message-ID:
<a class="moz-txt-link-rfc2396E" href="mailto:4acda1b40611291419r3eb8651en34e5e0c7570f1fb2@mail.gmail.com"><4acda1b40611291419r3eb8651en34e5e0c7570f1fb2@mail.gmail.com></a>
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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: <a class="moz-txt-link-abbreviated" href="mailto:tony@softins.clara.co.uk">tony@softins.clara.co.uk</a> (Tony Mountifield)
Subject: [asterisk-users] Re: What's up with the Manager Interface?!?!
To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>
Message-ID: <a class="moz-txt-link-rfc2396E" href="mailto:ekl114$94n$1@softins.clara.co.uk"><ekl114$94n$1@softins.clara.co.uk></a>
In article <a class="moz-txt-link-rfc2396E" href="mailto:456DDAE1.4050705@dynx.net"><456DDAE1.4050705@dynx.net></a>,
Richard Lyman <a class="moz-txt-link-rfc2396E" href="mailto:pchammer@dynx.net"><pchammer@dynx.net></a> wrote:
</pre>
<blockquote type="cite">
<pre wrap="">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)
</pre>
</blockquote>
<pre wrap=""><!---->
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
</pre>
</blockquote>
<pre wrap="">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.
</pre>
<big>Ouch ... error while writing audio data: : Broken pipe<br>
<br>
If any one knows about this please help me to fix this.<br>
<br>
</big>
<pre wrap="">Cheers
Ishanka
</pre>
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