I'm having a strange problem that I can't quite explain. I feel
like I'm missing something simple but I can't quite find it.
I'm not getting call-waiting CID whenever the incoming call is
delivered over IAX. However, when the same caller, coming in over
IAX, hits an empty ZAP channel,
I use the following:
exten = s,n,GotoIf($[${CALLERID(num)} : [0-9]{10}]?label)
Cheers,
Chad
On Nov 11, 2005, at 2:02 PM, Adam Robins wrote:
I am trying to test whether a callerid number is a valid ten digit
number. I'm a total novice with regular expressions.
I've tried:
exten =
on other broadband networks.
On 11/3/05, Chad Scott [EMAIL PROTECTED] wrote:
All,
I've been pushing hard for the use of Asterisk for the corporate
phone solution at the company I work for. Unfortunately, this
decision is completely out of my hands, although I've been applying
gentle influence
Hrm. Perhaps I should have actually responded off-list... DOH! :D
On Nov 7, 2005, at 9:11 AM, Chad Scott wrote:
Matt,
Sorry for the response off-list...
Would you be willing to talk to the powers that be for about 30
minutes about your experiences with Asterisk? I don't know what
All,
I've been pushing hard for the use of Asterisk for the corporate
phone solution at the company I work for. Unfortunately, this
decision is completely out of my hands, although I've been applying
gentle influence and pressure where I can.
The management for this project would like
My abandon count seems to increment by two whenever there's an
abandon... has anyone else noticed this?
I'm on (more or less) the latest CVS... probably about a month old.
I can't find anything in bugs.digium.com about it.
Thanks,
Chad
___
Noah, I could really use 1.3.4... however, a better question might be
how are *you* getting 1.3.4? I can't seem to get this from anyone,
including my reseller.
On Jan 26, 2005, at 8:42 AM, Noah Miller wrote:
Hi Chris -
I am getting to my wits end with these phones (and so is my boss). I
am
Maybe a corrupted voicemail directory? Or maybe the files are numbered
incorrectly?
Put the system into verbose mode and see what happens on the console
when you call it... that should help diagnose the problem.
On Dec 1, 2004, at 9:47 AM, Matt Hess wrote:
I'm having a problem with voicemail
On Nov 28, 2004, at 9:45 AM, Peter Svensson wrote:
Fair enough. If my unserstanding is correct perhaps someone can add a
note
to the wiki? It is not totally obvious.
Peter, why don't *you* add a note to the Wiki?
This is a community-supported project, and you're the community.
X-Lite doesn't support MGCP. You have to get the pro version for
that.
On Nov 27, 2004, at 9:05 AM, Leonardo J. Tramontina wrote:
I meant the MGCP messages were not happening, as SIP were when a
call happens... I can't watch them working... Did you understand what
is happening? Unfortunately
Read up on SetGroup and CheckGroup.
On Nov 22, 2004, at 9:57 AM, ismaelg wrote:
Hello,
I am trying a couple of days before to set up asterisk to redirects an
incoming call if the extension dialed is busy without success.
I just try to use 'Gotoif' command, with bad luck, it can't do what i
I'd recommend reading through the included sample queues.conf file
carefully... what you want is easily done.
You can either do a make samples (which will overwrite your existing
configuration) or look in the configs subdirectory in the asterisk
source tree for queues.conf.sample.
Cheers,
I thought Asterisk would indicate ringing on the PRI if it hadn't
answered it yet. If you answer the PRI and then Dial() don't you get
silence?
Since he says he's played a welcome message, I'd think the line has
been answered and therefore he has to indicate ringing to the dialing
party with
In queues.conf, try:
member = Local/extension@extension-context
So, for instance, something like:
member = Local/[EMAIL PROTECTED]
Cheers,
Chad
On Nov 15, 2004, at 7:59 AM, Shawn Dillon wrote:
I have the AgentCallBackLogin working well when the support
technician
logs into the queue manually.
It's been awhile since I've played with X-Lite, but I think it
absolutely *has* to use the MD5 auth stuff.
Use md5secret rather than secret in sip.conf. You'll have to MD5 hash
your password... there's documentation on this in the Wiki.
-Chad
On Nov 10, 2004, at 9:25 AM, Ashling O'Driscoll
Use Goto(context, extension, priority).
So, for instance:
[sip-inbound]
whatever-extension-used-inbound = Goto(other-context, other-extension,
1)
Hope that helps.
On Oct 25, 2004, at 11:34 AM, spkao wrote:
Hi,
Does anyone know if there is a way to have a sip or iax inbound call
dial an
Rather than use AgentCallbackLogin, try AddQueueMember and
RemoveQueueMember.
On Oct 21, 2004, at 1:22 PM, Jolan Luff wrote:
Hi,
I am trying to setup two extensions by which agents can automatically
login and logoff with asterisk-1.0.0. My extensions.conf looks like
this:
exten =
You're not going to be able to achieve what you want quite in the way
you visualize it.
The best advice I can give is to transfer the call to your mobile phone
and take the call with you that way. This prevents the caller from
being able to continue the call by simply staying on the line.
On
Anything that can generate g729 encoded data requires a license.
Period.
You may be able to find a utility to convert between GSM and g729, but
it will have a license associated with it.
If you're trying to avoid the licensing costs, just use another CODEC.
Your phone must support
The card requires no power from the phone company... it is powered from
the PCI bus.
However, I'm not sure if the card can detect if the line is there or
not... I've never tried it. Plug a line into it and see what you get!
:)
On Tue, 2004-10-19 at 12:55, Alex van Es wrote:
Hi all,
I just
The audio is carried on two RTP streams: one for each direction. Is it
possible those streams are being blocked by a firewall or something of
the sort?
The attempting native bridge message means that Asterisk is bridging
the two calls together without doing any codec translation... uLaw to
How about:
exten = s,1,GotoIf($[${CALLERIDNUM} : ^(904|321|407|252)[0-9]{7}$] ?
2:3)
exten = s,2,Goto(somewhere,s,1)
exten = s,3,DoWhateverElse
On Oct 15, 2004, at 7:21 AM, Ben Wern wrote:
That's pretty good..
I have a similar situation, where I need to match all the area codes
in a particular
You might have silence suppression turned on in the soft phone... turn
it off.
If that's not the culprit, use a different codec... maybe the soft
phone just doesn't speak GSM right.
On Oct 15, 2004, at 6:16 AM, CHAUVELIN Samuel wrote:
I use my asterisk to SIP H323 Gateway.
Softphone SIP -
Does the option A(filename) not work for you?
On Oct 15, 2004, at 5:38 AM, Robinson Tim-W10277 wrote:
It has all gone very quiet - I still need this...I spent a fair bit of
time looking at it but never got it to work. Needs someone with a bit
more of an understanding of Asterisk's architecture
Apparently you did not read my entire message.
I specifically stated it would be non-optimal and a bare-bones solution.
While 600 ohms may be the characteristic impedance of the wire run, a
mismatch at either end will change the impedance of the entire path to
some value that is the related to
I think the Polycom phones do it via a lookup in the directory stored
in the phone. At least, that's how I read it from the Admin Guide.
On Oct 12, 2004, at 11:19 PM, Brent Franks wrote:
Hi Joe,
The Polycom IP phones support this, however currently there is no
support for it in *.
I don't think
I'm not aware of anyone making something like this for telephone
systems... all the ones I've seen are RF matching networks for
transmitters and receivers. You'd likely have to build it.
On Oct 13, 2004, at 7:43 AM, Jayson Vantuyl wrote:
On Wed, Oct 13, 2004 at 07:12:12AM -0700, Chad Scott
On Oct 10, 2004, at 10:08 AM, Wolf Paul wrote:
dean collins [EMAIL PROTECTED] wrote:
However, those of us not working with hefty corporate budgets may not
have the option of spending
$100 for a test machine when there's a more cost effective option
available.
I'd seriously suggest, in your
If you're certain it is an impedance problem and the impedance of your
line is lower than that of the CO, you can increase the impedance of
your line by putting a potentiometer in-line and adjusting it until the
sidetone disappears. This is a bare-bones solution and decreases the
efficiency
You shouldn't be seeing these sorts of problems regardless of how the
zaptel drivers are performing. The kernel should panic and the system
halt if things go awry. If the system hard locks, that's almost always
a hardware issue of some sort.
What your problem sounds like is some conflict on
Impedance is the measure of total opposition (resistance, capacitance,
and inductance) to alternating current flow. Adding resistance will
raise the impedance of the line.
On Oct 12, 2004, at 12:58 PM, Rich Adamson wrote:
Impedance does not equal resistance. Apples and oranges.
If I recall correctly, the National ISDN protocol (NI2, I think) has
the capability of forwarding CID NAME to the provider who can then do
whatever they want with that information (including simply discard it).
On Sep 16, 2004, at 5:29 AM, Brandon Patterson (peering) wrote:
The owner of the
On Sep 13, 2004, at 11:32 AM, Andrew wrote:
I'm a bit new to the terminology. Let me ask my question more simply,
even though I think you already answered that it should
work
I want to receive calls into the Asterisk PBX via a cheap POTS-PBX
method, such as a WinModem or other FXS endpoint on
Is this problem more-or-less continuous or does it happen occasionally?
If the latter, does it always happen on channel one?
At first glance, this looks like some sort of line hit, maybe a loss of
sync?
On Sep 10, 2004, at 8:58 AM, Claus Futtrup wrote:
Hi Guys,
Im having some problems with a
Do these two events coincide? If so, I'd suspect memory problems.
If they don't coincide, I'd still suspect memory, but I'd also look at
IRQ sharing issues.
On Fri, 2004-09-03 at 09:16, Daniel Jimenez wrote:
To top this off, I also get PRI errors
Sep 3 10:56:52 NOTICE[196620]:
For what it's worth, Asterisk compiles and runs fine on my Fedora Core
2 system right out of the box.
I hope that helps to narrow down your issue.
On Jul 7, 2004, at 1:35 AM, Xavier Olivella wrote:
When installing asterisk, I follow de Getting Started manual and i
get the
following error while
Once you have downloaded the CVS once, subsequent updates will take up
far less bandwidth than downloading the .tar.gz regularly.
On Jul 5, 2004, at 7:30 PM, [EMAIL PROTECTED] wrote:
I have not high speed connection (i can't use cvs method), i was
download in my office the .tar.gz file and
This looks like a job for AGI...
I'd do something like
exten = _0207XXX,1,Dial(SIP/$EXTEN},15)
exten = _0207XXX,2AGI('missed-call-email.agi')
exten = _0207XXX,3,Voicemail(u${EXTEN:4})
exten = _0207XXX,4,Hangup
exten = _0207XXX,102,AGI('missed-call-email.agi')
...etc...
On Jul
Why don't you include a context on a schedule?
I have a support queue included only at certain times, such as monday
through friday, 6a to 6p. All other times I include a context that
sends that extension straight to voicemail.
Check out http://www.voip-info.org/wiki-Asterisk+tips+openhours
On
I've got another issue I can't quite figure out and a search of the
archives and Google turn up nothing...
Say a call comes in (these are all via SIP) and is sent into a
Queue(myqueue,t,,,300). Note the t to allow whomever receives the
call to transfer it.
The call is enqueued, and the logged
Thanks for the pointer... I don't think the device is the issue because
these types of transfers work in any other case.
For instance, say I call Bob from my SIP phone. Bob answers and decides
he's going to transfer me to Mike. He hits 'transfer,' dials Mike,
talks for a bit, then hits
I think I've narrowed this down via experimentation...
This seems to happen *only* when a call is sent to a phone out of the
queue and you attempt to put it back in the queue.
All the other use cases I've said seem to work during testing (from the
queue to party A, transferred to party B).
So,
;
free_user(vmu);
--- 1768,1773
On Jun 26, 2004, at 8:52 PM, Chad Scott wrote:
I've been banging my head on a brick wall for about an hour now trying
to understand why the following doesn't work (which is even provided
as an example in the distribution!).
The goal is to create
I'm using HEAD right out of the repository. It also doesn't appear the
'o' extension is working either.
Is this just something that's unfinished in HEAD?
On Jun 27, 2004, at 6:59 AM, Philipp von Klitzing wrote:
Hi!
I've been banging my head on a brick wall for about an hour now trying
to
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