We have a server with a TE120 on a partial PRI trunk that several times
a day declares the PRI trunk down and stops handling calls until the
asterisk is stopped, the zaptel/te120 modules reloaded, and asterisk
started.
Just before things go down, the log shows the following error:
ERROR[9424]
Tell me when to stop laughing. Multiple channels and unlimited minutes ? No
sane person will give that to you.
On Dec 30, 2007 2:16 AM, Steve Finkelstein [EMAIL PROTECTED] wrote:
Hi all,
I have a budget to work with and was wondering if there are any folks
providing SIP/IAX2 trunking for
Justin Case wrote:
Tell me when to stop laughing. Multiple channels and unlimited minutes ?
No sane person will give that to you.
Yap I agree...
but but for about $900 per month one could get T1 (24 channels)
unlimited in/out as far I seen last time our providers rates.
Senad
On Dec
Senad,
Mind if I ask who that provider is?
Thanks.
Sent from my iPhone
On Dec 31, 2007, at 8:10 AM, Senad Jordanovic [EMAIL PROTECTED] wrote:
Justin Case wrote:
Tell me when to stop laughing. Multiple channels and unlimited
minutes ?
No sane person will give that to you.
Yap I
Hello All,
We need IP Phones in Lahore, Pakistan. Preferred brands are Atcom, Polycom
and Grandstream. However any other good brand is also acceptable. Our client
is interested in cheaper phones. Can anyone provide in Pakistan ?
Regards,
--
Kashif Naeem
Director
Hadi Telecom
Doug Lytle wrote:
Jay Moore wrote:
Hi list.
I'm new to IVRs and trying to set up one that toggles an auto-forward
flag on or off for specific accounts.
Why don't you post what you've currently written and we'll go from there?
Doug
Actually, after switching to AEL, I think I
I need the digit map to call China. Example number:
011-86-10-6887-
011-International (obvious)
86 is country code (China)
10 is city code (Beijing)
Last 8 digits are the number.
I tried using 011xxx.T but it always asks me to enter more digits. Tried
some variations as well,
On Dec 31, 2007, at 11:36 AM, Michael Munger wrote:
I need the digit map to call China. Example number:
011-86-10-6887-
011-International (obvious)
86 is country code (China)
10 is city code (Beijing)
Last 8 digits are the number.
I tried using 011xxx.T but it always asks
Hi all,
I've been working on this for days and can't find a solution. I need to use
AddQueueMember for my agent logins to my Queues -- but a number of my agents
are outside the main server, which is connected to my asterisk network over
IAX2. I can't just do a AddQueueMember(queuename) because
Here is some information I received from my account rep at Digium regarding
this information:
-- Digium --
That's news to me as well as the rest of the sales team. We were told that
users cannot change the 1gb flash card.
I just spoke with one of the Sales
It is when you type 'make install' that these directories get created.
'make linux26' IS obsolete as another poster mentioned.
broadband Voice wrote:
I successfully obtained the Asterisk code and extracted them into
/usr/src. When I make and install asterisk, zaptel, libpri etc. Are
they
On Mon, 2007-12-31 at 12:02 -0600, Gregory Malsack wrote:
Here is some information I received from my account rep at Digium regarding
this information:
-- Digium --
That's news to me as well as the rest of the sales team. We were told that
users cannot
Jerry Jones wrote:
Yours should work if you wait long enough for t to timeout.
I think your digit map needs a T on the end of it if you want to allow
timeouts for that match.
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I would GUESS that if this line is removed, asterisk is settling on slin
codec for the channel and does not try to negotiate anything better?
Hence it will work without it.
Mojo
Bhrugu Mehta wrote:
hi, all
I have test echo application for just fun.
I can'nt understand why this is used
Brian J. Murrell wrote:
And how long will that flash card last with the log and vmail churn?
Flash devices have a limited number of writes you can do to a single
cell before it wears out and cannot be written to any more.
So does one have to throw out the whole appliance when one wears out
I don't understand the
USERS vs PEER vs FRIENDS. I just use Peer for everything. Has to do
with can I only contact you or can you contact me too? ... Peer does
it all.
RealTime does have an issue. If you don't turn on caching, then it holds no
state information. So if you think you're going
That was one of the many iterations I tried already. It seems to respond
in that it recognizes that I am dialing 01186106887, but then it
only connects me to a dial tone and says Enter More Digits.
There has to be something simple I am over looking here. I understand
regular expressions,
Michael Munger wrote:
only connects me to a dial tone and says Enter More Digits.
It actually says this?
I would say then it's not the phone, but your phone system's
programming. The Polycoms don't verbally say anything, at least not the
ones I deal with.
Doug
--
Ben Franklin
Doug Lytle wrote:
Michael Munger wrote:
only connects me to a dial tone and says Enter More Digits.
It actually says this?
I would say then it's not the phone, but your phone system's
programming. The Polycoms don't verbally say anything, at least not the
ones I deal
Gregory Malsack wrote:
-- Digium --
That's news to me as well as the rest of the sales team. We were told that
users cannot change the 1gb flash card.
I just spoke with one of the Sales Engineers and he stated that it is
apparently possible to change
Brian J. Murrell wrote:
And how long will that flash card last with the log and vmail churn?
Flash devices have a limited number of writes you can do to a single
cell before it wears out and cannot be written to any more.
All modern flash cards (not flash chips, which are lower level) have
Mojo with Horan Company, LLC wrote:
So try: 011XXT in your digit map, meaning 011 plus at least six
digits, consider it good
Err duh, that's ten X's not six :) To account for the Tajikistan
example plus a little bit of local number.
Really, it's dead simple to just do it like
Kevin P. Fleming wrote:
the Linux kernel on the AA50 does not have NFS
support nor SMB support, and there are no userspace tools present to
handle NFS or SMB mounting of filesystems.
FUSE? But it's probably not on the appliance.
Regards,
Philipp Kempgen
On Mon, 2007-12-31 at 14:16 -0600, Kevin P. Fleming wrote:
No. The files to repopulate the CF card are available to users who have
active support subscriptions and they can replace the card. Users can
also, of course, make a backup copy of the card on a new card when they
receive the unit and
Hey all,
I've setup my asterisk install on a CentOS5 server, I've got a few
IAX2 and SIP softphone clients connected on the same subnet and at
least 1 external IAX2 softphone. However I'm having some difficulty
getting the Polycom hardphone to function correctly. Watching the logs
and debug trace
On Mon, 2007-12-31 at 21:13 +, Glenn Gillen wrote:
I'm having some difficulty getting the Polycom hardphone to function
correctly. Watching the logs and debug trace it:
- Registers correctly
- Is able to make calls to other peers
However it is not able to answer calls made to it.
At 14:27 12/31/2007, Mojo with Horan Company, LLC wrote:
Mojo with Horan Company, LLC wrote:
So try: 011XXT in your digit map, meaning 011 plus at least six
digits, consider it good
Err duh, that's ten X's not six :) To account for the Tajikistan
example plus a little bit of
On Sunday 30 December 2007 14:40:40 Mindaugas Kezys wrote:
Thank you!
Will it come to 1.4.16.3 or 1.4.17?
Yes, it will.
--
Tilghman
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asterisk-users mailing list
To UNSUBSCRIBE
I have been trying to track down the cause/fix for a problem and I am out of
ideas... I am hoping one of you can point me in the right direction.
The symptom is that when a calls is placed from an internal extension
through an analog line to a number on the pstn the caller can hear the
callee but
glenn,
check your handset cord... it might be plugged into the wrong port in
the back of the phone. perhaps the headset jack...
daveC
Glenn Gillen wrote:
Hey all,
I've setup my asterisk install on a CentOS5 server, I've got a few
IAX2 and SIP softphone clients connected on the same subnet
Doug wrote:
At 14:27 12/31/2007, Mojo with Horan Company, LLC wrote:
Mojo with Horan Company, LLC wrote:
So try: 011XXT in your digit map, meaning 011 plus at least six
digits, consider it good
Err duh, that's ten X's not six :) To account for the Tajikistan
example plus a
On Dec 28, 2007 8:28 PM, Al lists [EMAIL PROTECTED] wrote:
what method is preferred:
haylafax and Iaxmodem or spnadsp for faxing.
What are you trying to do and do you have a T1 or ISDN line?
___
--Bandwidth and Colocation Provided by
at this time is terminating a SIP trunk,
each DID will get its own fax box.
I guess at this time i'm looking to find a tutorial for installing iaxmodem
and hylafax as it seems to be the answer.
On Dec 31, 2007 9:11 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
On Dec 28, 2007 8:28 PM, Al lists
Unless your provider provides a T.38 gateway, fax over SIP is pretty
much guaranteed to be unusable. Often you can get away with it over a
LAN using G711a or G711u, but any of the lower bandwidth codecs /won't/
be able to properly handle fax calls.
Whilst I haven't used it myself, I believe
If by fax box you mean an ATA with a fax machine attached them
Asterisk 1.4 with T38 passthrough should work if the SIP provider has
T.38 capabilites.
If by fax box you mean a 'faxmail inbox' then no Asterisk cannot
help you terminate that from SIP. Get a Cisco gateway, make sure your
provider
Rob Hillis wrote:
Last time I heard IAXModem didn't support T.38 because the IAX2
protocol didn't support T.38 - whether that's still the case or not, I
don't know.
There are actually two reasons. One is that T.38 over IAX is not
defined. The other is the current T.38 termination support in
Brian Alexander wrote:
I have been trying to track down the cause/fix for a problem and I am
out of ideas... I am hoping one of you can point me in the right direction.
The symptom is that when a calls is placed from an internal extension
through an analog line to a number on the pstn the
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