Displaying 20 results from an estimated 508 matches for "riddel".
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riddell
2004 Sep 01
3
Distinctive rings
Is it possible to allow distinctive rings work for FXS ports as well?
I need a certain FXS extension to ring a distinctive double ring.
I modified zapata.conf appropriately for dring1,dring2 and it just
Seems to ignore my updates.
Do distinctive rings only work for FXO ports?
Paul Seniuk
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2008 Mar 10
11
Microsoft Office Communications Server
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Has anyone done any integration with this?
All I know so far is that it appears to use some non standard form of SIP.
Any pointers?
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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Version: Gn...
2005 Jan 17
3
On Hold music
...need a sound card in the asterisk
box to use this hold music feature.
Hope this helps.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Computer
Onsite Support
Sent: Monday, January 17, 2005 10:51 PM
To: matt.riddell@sineapps.com; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] On Hold music
Please be more specific regarding symbolic link of mpg321 so I can
troubleshoot it myself. The strength thing is that I tried this in three
other different computers and can't get...
2008 Jan 23
5
Snom 320 Lost Settings
...as anyone ever seen an Snom320 lose settings?
It's been working fine for months and then I got a call this morning
saying that it was asking for country, timezone etc.
I logged in remotely, and it had lost the server address, username,
password, mailbox and ringtone.
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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Version: Gn...
2005 Sep 21
7
add 0 (zero) to incoming callerID - how?
I have an asterisk box and SIP / IAX2 phones.
To call out, users have to add 0 (zero) before a real telephone number.
That means, that if they want to call someone that has a number 123456,
they have to call 0-123456.
Simple, right?
This has a serious drawback though - when someone calls us from the
number 123456, we see the callerID 123456, and we're unable to use the
callback/redial
2006 Apr 26
6
Sphinx2
I have a gateway, which I call from my mobile phone (free of charge,
since it is the same phone company).
This gateway gives me a dial tone. I can than dial to any extension
number or even other gateways, ....
It is getting more a trouble to remember all the numbers, or to key in
all the long phone numbers when you got the dialtone.
I was thinking of using for this Sphinx2. How can I
2009 Sep 02
1
AMI Originate Commands executed in sequential Order problem
...: Contents of asterisk-users digest..."
Today's Topics:
1. Inquiry:Problem with Call Parking (hadi motamedi)
2. Re: Asterisk Web Meetme module not loading (Glen)
3. Re: Inquiry:Problem with Call Parking (Lee, John (Sydney))
4. Re: Asterisk Web Meetme module not loading (Matt Riddell)
5. Re: Inquiry:Problem with Call Parking (hadi motamedi)
6. Re: Inquiry:Problem with Call Parking (Matt Riddell)
7. Re: Inquiry:Problem with Call Parking (Darrick Hartman)
8. Re: Asterisk Web Meetme module not loading (Glen Ganderton)
9. Re: Inquiry:Problem with Call Parking (Lee,...
2009 Aug 31
4
Inquiry:How to hide Caller Id
Dear All
Can you please do me favor and let me know how I can hide the subs number
being displayed on his phone when he goes off hook ? I mean when the subs
goes off hook he sees his assigned number on his phone and I need to disable
this feature . I don't know from which configuration file this feature is
coming so please let me know how can I disable it .
Regards
H.Motamedi
--------------
2005 May 25
4
SER Help
Hi,
I'm looking for a tutorial or installation guide for
SER to be used with asterisk to solve the remote SIP
agent problem. All the documents available are for
large scale installation.
Any help is highly appreciated.
Regards.
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2007 Jan 02
5
Call connected, cannot hear or speak - $20 for fix
I am able to get this script to dial, but I am unable to talk or hear
anything. The script asks for the number to call and the the caller id to
display (if user is not at their normal extension). Once submitted, the
external extension receives a call, once answered the call is then placed to
the dentition number.
The script works as the call is place, but I cannot hear or say anything.
Any one
2004 Sep 17
2
Re: Asterisk-Users Digest, Vol 2, Issue 163
...hen
with this the fax machine report "Not Response". I modified the audio level
in zapata.conf and after that the fax machine report "Commnunication Error".
Do you an idea what could be ?
Thanks,
Angel.
> Message: 3
> Date: Sat, 18 Sep 2004 00:48:23 +1200
> From: matt.riddell@sineapps.com
> Subject: Re:Re: [Asterisk-Users] Fax and Asterisk
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Message-ID: <414B85D7.29122.5EA177@localhost>
> Content-Type: text/plain; charset=US-ASCII
>
> On 17...
2006 Sep 22
6
Digium G.729 codec binaries updated for Asterisk 1.4 beta
The x86 and x86_64 Digium G.729 codec binaries have been updated for use with the Asterisk 1.4 beta (which should also work on current svn trunk).
Anybody that is using the older modules with the 1.4 beta (or svn trunk newer than several days ago) is strongly encouraged to upgrade immediately, to avoid potential issues.
--
Jason Parker
Digium
2006 Sep 22
6
Digium G.729 codec binaries updated for Asterisk 1.4 beta
The x86 and x86_64 Digium G.729 codec binaries have been updated for use with the Asterisk 1.4 beta (which should also work on current svn trunk).
Anybody that is using the older modules with the 1.4 beta (or svn trunk newer than several days ago) is strongly encouraged to upgrade immediately, to avoid potential issues.
--
Jason Parker
Digium
2005 Sep 20
6
iax2 trunking wackyness
Hi
I was doing some bandwidth testing, and my incomming usage is
36% more than my outgoing bandwidth.
The setup is IAX2 trunking using GSM codec.
Is there any obvious reason I am overlooking to figure out why
there is such a big difference between the two.?
I am using CVS-head September 3rd, maybe there is a version
skew?
Any suggestions will be appreciated.
Thanks
Clive
2010 Mar 18
3
Free Daily Asterisk News iPhone and iPod Touch app
Hi all,
I've released another free app for the iPhone and iPod touch - this one
lets you read the Daily Asterisk News.
Hope you enjoy it :D
http://www.venturevoip.com/news.php?rssid=2371
--
Cheers,
Matt Riddell
Managing Director
_______________________________________________
http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/exchange.php (Full ITSP Solution)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
2019 Jan 11
2
[asterisk-app-dev] Multiple ChannelDestroyed events for the same channel
...ng once to the event:
ari.on('ChannelDestroyed', channelDestroyed);
Is this normal?
I’m writing like a CDR on channel destroyed so don’t want to write it multiple times.
Should I keep an array of channels and only write if I haven’t seen the event for that channel before?
Cheers,
Matt Riddell
_______________________________________________
asterisk-app-dev mailing list
asterisk-app-dev at lists.digium.com
http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
2016 Oct 17
2
Streaming for ASR
Matt Riddell wrote:
>
>> On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradovera at gmail.com
>> <mailto:luca.pradovera at gmail.com>> wrote:
>>
>> I have been working on designs for two different projects, where both
>> of them would need to use the IBM Watson st...
2005 Mar 15
2
Wiki down: Is there another source for documentation?
As the title suggests, I was wondering if there was another source of
documentation for Asterisk.
Related: If one wanted to contribute to documentation, who would one
contact?
Thanks!
Sean
2005 Sep 30
2
Why does the s extension not work in my extensions.conf file
Hello
In my extensions.conf file:
[frompstnisdn]
exten => s,1,Dial(SIP/200&SIP/202,20)
exten => s,2,Voicemail(su200)
exten => s,3,Hangup
I use the s, start, extension to handle incoming calls.
In my zapata.conf:
context=frompstnisdn
This works ok on another asterisk box I setup. But on incoming calls I get:
-- Extension '787367' in context 'frompstnisdn'
2005 Aug 28
2
Need quote for Asterisk and billing remote install
Please send me a quote for remote installation of
Asterisk, GUI administration, and billing for calling
card, caller ID based prepaid, and postpaid.
Off list please.
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