[asterisk-users] Re: asterisk-users Digest, Vol 28, Issue 152
Ishanka Anuradha Ranasooriya
ishanka at mnetplus.com
Wed Nov 29 20:15:33 MST 2006
asterisk-users-request at lists.digium.com wrote:
> Send asterisk-users mailing list submissions to
> asterisk-users at lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> asterisk-users-request at lists.digium.com
>
> You can reach the person managing the list at
> asterisk-users-owner at lists.digium.com
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> 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)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 29 Nov 2006 16:06:57 -0600
> From: "Kim Jones" <kim at anmsinc.com>
> Subject: [asterisk-users] beeping noise in background
> To: <asterisk-users at lists.digium.com>
> Message-ID: <000401c71402$aff83d40$052110ac at ANMS.pvt>
> 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
>
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061129/81c3ce79/attachment-0001.htm
>
> ------------------------------
>
> Message: 2
> Date: Wed, 29 Nov 2006 15:16:45 -0700
> From: "Douglas Garstang" <dgarstang at oneeighty.com>
> Subject: RE: [asterisk-users] What's up with the Manager Interface?!?!
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <645FEC31A18FE54A8721500CDD55A7B6035D0C0D at mail.oneeighty.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
>
>> -----Original Message-----
>> From: Steve Edwards [mailto:asterisk.org at sedwards.com]
>> 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.
>
>
> ------------------------------
>
> Message: 3
> Date: Thu, 30 Nov 2006 09:19:13 +1100
> From: "Eric Bishop" <asterisk.eric at gmail.com>
> Subject: [asterisk-users] g726 voice prompts
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <4acda1b40611291419r3eb8651en34e5e0c7570f1fb2 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Anyone know if it posible to make voice promps native g726 or g711 format?
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061129/4e4f4dfb/attachment-0001.htm
>
> ------------------------------
>
> Message: 4
> Date: Wed, 29 Nov 2006 22:19:16 +0000 (UTC)
> From: tony at softins.clara.co.uk (Tony Mountifield)
> Subject: [asterisk-users] Re: What's up with the Manager Interface?!?!
> To: asterisk-users at lists.digium.com
> Message-ID: <ekl114$94n$1 at softins.clara.co.uk>
>
> In article <456DDAE1.4050705 at dynx.net>,
> Richard Lyman <pchammer at dynx.net> 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061129/b199421b/attachment.htm
More information about the asterisk-users
mailing list