As an example, here is my custom digitmap:
digitmap
dialplan.digitmap=9,911|9,411|0T|00T|1xx|9,011x.T|9,1[2-9]xx[2-9]xx|9,[2-9]xx|*7x|7x|*1xx|*8
The | is used to separate different entries. The comma means that it'll
keep providing the dial tone after hitting 9. If you see a T after
Hey all,
I'm having problems where there is significant static when making SIP -
PSTN calls. SIP - SIP and SIP - VM calls are totally clear and fine.
Here's the setup:
Polycom 601,501, and ten 301s.
Digum 2400 TDM card w/echo cancelling, 12 FXO ports.
The TDM card is on IRQ 5 with nothing
?site=sr:SEARCH:MAIN_RSLT_PG
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Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
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about them:
http://www.vegastream.com/vega400.asp
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Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
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Post your extensions.conf and what's on the CLI (asterisk -r)
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Roman Volf
Keystreams
be negative.
defaults to now.
timezone: timezone, see /usr/share/zoneinfo for a list.
defaults to machine default.
format: a format the time is to be said in. See voicemail.conf.
defaults to "ABdY 'digits/at' IMp"
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Roman Volf
Keystream
Krystian Filiks wrote:
What about plain g729?
My main concern is the Hardware, anyone that can tell me if this
Supermicro 6014H-32 is stable and sutible for asterisk?
Supermicro Superservers are traditionally extremely stable and reliable.
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Roman Volf
Keystreams Internet Solutions
[EMAIL
.
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Keystreams Internet Solutions
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Sounds like you are missing the mysql client libraries.
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Keystreams Internet Solutions
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Please stop double posting your questions. This will not help you get
any answers.
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Roman
Have you looked here:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Jason Brown wrote:
Here is something I wasnt quite expecting from a business deployment,
and dont have an answer for. Maybe one of you do
Have you tried putting both access points on the same channel?
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Jim Meehan wrote:
I've got a Hitachi WIP-5000 phone. Seems to work well with my Asterisk setup,
except for a few annoyances:
1) If the phone has been sitting unused
I setup this google group because Google seemed to be good at
threading the topics from the list. I have noticed that many threads
don't go as well as planned and wind up in the wrong place.
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Oh and it was just a test to see how it worked. Pretty easy to setup
Asterisk-users
On Apr 8, 2005 8:47 PM, Roman Volf [EMAIL PROTECTED] wrote:
I setup this google group because Google seemed to be good at
threading the topics from the list. I have noticed that many threads
don't go as well
This should work fine.
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Matt Riddell wrote:
Peter Bowyer wrote:
exten 456,1,Background(Please-set-time-mmddhhmm)
exten _.,1,System (date ${EXTEN})
If I dial 456 I get the message, so I type 04021305 (2nd April, 13:05).
On the console
Or if google is too complex, http://asterisk.keystreams.com
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Robert Webb wrote:
On Tue, 15 Mar 2005 11:56:18 -0500
Fabian Borot [EMAIL PROTECTED] wrote:
Hello all
I have been learning * from almost 1 month now. It looks really
powerfull
Because SIP works with things other than Asterisk. IAX does not.
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Joseph wrote:
I'm just curious why Sipura isn't using free IAX protocol with their
devices instead of SIP?
With IAX NAT traversal would have been easier, so why
It would be helpful if you pasted the relevant sections of sip.conf and
extensions.conf
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Im new to Astererisk. I compiled the latest CVS and setup the server. It
looks like things are working. I'm running kphone
In case you didn't get the last 5 responses, you just need to create an
alias for the two email accounts.
But honestly people, do you not read the rest of the thread before
responding? Its already been answered.
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
C F wrote:
yes, create
of 2005. It will *not* be updated in real time (at
least not for now)
Please direct flames/questions/comments to [EMAIL PROTECTED]
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Keystreams Internet Solutions
[EMAIL PROTECTED]
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Have you seen the user guide?
http://www.sipura.com/Documents/SPA841UserGuide.pdf
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
Scott Bussinger wrote:
There isn't an Admin Guide for the SPA-841 as far as I know.
However, I have found that the Admin Guides for their other
products
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