Eric Wieling wrote:
Erick Weber V. wrote:
I'll Like to now how to insert a pause on a SIP string. I have a ATA
186 and
a FXS => FXO converter so I will like to program a extension that can be
dialed and it will dial the ATA extention, wait for dial tone and then
dial
the phone number.
You ca
On Wed, 2004-04-14 at 07:55, I wrote:
> Did any of you ever experience missing "180 ringing" messages when
> dialing from a sip agent through an asterisk box?
Reading up some more on this it now appears to me this is RFC compliant
behaviour as a 183 Session Progress is being sent instead. So, my
Did any of you ever experience missing "180 ringing" messages when
dialing from a sip agent through an asterisk box? I tried both a Zap and
another SIP channel for dialout. Using a ISDN zap channel the call setup
would look like
SIP-UA * PSTN
| ||
|INVITE->
Matt White wrote:
James H. Thompson wrote:
Would it make any sense to store the voice mail formatted as a email
msg in a Maildir directory
structure.
Then you could also retreive them with an email client.
As an extension of this thought, how about going one step further
and storing the voicema
James H. Thompson wrote:
Would it make any sense to store the voice mail formatted as a email msg in a Maildir
directory
structure.
Then you could also retreive them with an email client.
As an extension of this thought, how about going one step further
and storing the voicemail on an imap server
On Mon, Apr 12, 2004 at 02:19:23PM -0400, James H. Cloos Jr. spake thusly:
> Yes, there would be. This is the same issue as using nfs mail spools
> with maildir style storage. W/o locking there is no way to guarantee
> that two servers do not create the same vm file on top of one another.
The Ma
On Mon, 12 Apr 2004, Scott Laird wrote:
> I mean, even G.711 is only ~80 kbps (including overhead), so you
> should be able to run hundreds of simultaneous conversations on 100
> Mbps Ethernet without running out of bandwidth.
Disclaimer: I do not build networks for a living -- the following co
I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833)
and OpenPhone (H.323, G.711).
But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I
press any key on analogue phone connected to ATA, Asterisk shows following message:
-- Exec
Hopefully they won't mandate a specific amount of "Canadian content" per
call. ;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of TC
Sent: Tuesday, April 13, 2004 7:40 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Vonage goes to .ca
FYI
http://www.it
Keep in mind that if this is a Local T1 you CAN have people calling you with
no callerid. Make sure to include that possibility in your dialplan:
With CallerID:
exten => _*XX*916222,1,Dial(
Without CallerID:
exten => _**916222,1,Dial(
MATT---
-Original Message-
Hi,
Yet another question, now that I have callerid working correctly, I'm
trying to work out how to utilise the different numbers I have. I have a
100 number range allocated to my E1/PRI/OnRamp service.
My incoming calls are handled like this:
Advertised/published number is an analogue line termi
On Wed, 14 Apr 2004, Gary wrote:
> they actually send the caller-id info after the SECOND ring.
That depends. ;) If you define ring as 'single burst of ring voltage',
then it does come after the second ring. That's how * will have to look
for it, after all.
It changes if you have Distinctive
On Wed, 2004-04-14 at 00:52, Vic Cross wrote:
> G'day Adam,
>
> This drove me nuts for a few days just recently (only fixed it yesterday
> in fact, and I've not had a chance to update any doco anywhere yet).
>
> On Wed, 14 Apr 2004, Adam Goryachev wrote:
>
> > Actually, now that I look at the
On Wed, 2004-04-14 at 10:32, Mike Machado wrote:
> I found ways to do substrings. So this is what I did. I changed the zap
> channels to come into a context called 'fixup' and then jump into
> default after doing the parsing and setting of CID.
>
> [fixup]
>
It was probably because I emailed everyone in the company every day until
it got fixed...
My next step was to start calling executives on their cell phones.
;-)
To be fair, it took them a while to get back to my original email, but
after I emailed the CEO, they were very responsive and sent me
I found ways to do substrings. So this is what I did. I changed the zap
channels to come into a context called 'fixup' and then jump into
default after doing the parsing and setting of CID.
[fixup]
> map it in the dial plan. Are there substr functions I can use? Can I
Look at the wiki/tiki www.voip-info.org and search for cmd cut or at
your console do a show application cut.
If you still don't know what to do, do a search on the wiki for
variables or read the README.variables in the asteris
I am setting up a box with a T100P. Everything is going well. The
company I am working with has their one phone switch gear. They
provisioned me a E&M Wink T1. Cannot do PRI unfortunately. We chose E&M
so we could pass an unlimited number of DIDs to the trunk as apposed to
FXS loopstart signaling
Hi Michael,
I tried to reply to your prive e-mail, but it seems like your mail
service blacklists the whole of Hong Kong. :-\
Regards,
Steve
Michael Welter wrote:
Just when I thought I couldn't be wrong, I was wrong. We have
woodpeckers that drill into the arial telephone cables, and water
s
On Tue, Apr 13, 2004 at 07:09:57PM -0400, Jon Pounder wrote:
> > FYI
> > http://www.itbusiness.ca/index.asp?theaction=61&sid=55298
> > did not like this by line in the story
> > t"he CRTC has said it will likely regulate voice over IP the same as other
> > phone services."
> I noticed exactly the
In case anyone was wondering, I managed to solve the issue. Turned out
to be a problem with one of the X100P cards conflicting IRQ's with my
ethernet card. This was causing zaptel to see only one X100P and then
TDM400P, so the second channel was the TDM400P, which is why I was
getting the error
> FYI
> http://www.itbusiness.ca/index.asp?theaction=61&sid=55298
>
> did not like this by line in the story
> t"he CRTC has said it will likely regulate voice over IP the same as other
> phone services."
>
I noticed exactly the same thing and thought the same when I read it earlier.
CRTC knows
they actually send the caller-id info after the SECOND ring.
Now of course if the au indications were changed to combine the first
and second ring to appear as one ring, no other changes would be needed
??
Gary
On Wed, 14 Apr 2004 00:36:16 +1000, Duane wrote:
>Adam Goryachev wrote:
>> I am in A
Sean Garland wrote:
In almost all my calls now, I am getting beeps and loud and soft parts
of a conversation. It is getting very irritating. Has anyone had this
happen? How do I get rid of it?
Set relaxdtmf=no in /etc/asterisk/zapata.conf.
___
Asteri
FYI
http://www.itbusiness.ca/index.asp?theaction=61&sid=55298
did not like this by line in the story
t"he CRTC has said it will likely regulate voice over IP the same as other
phone services."
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lis
Brian Cuthie wrote:
Tor Houghton wrote:
On Tue, Apr 13, 2004 at 02:56:48PM -0400, Jeremy McNamara wrote:
Use IAX2, it is a better IAX protocol.
Jeremy McNamara
P.S. If you really must have it, dig thru the channels/Makefile, but
there is zero reason to use it any longer.
Well, I use
In almost all my calls now, I am getting beeps and loud and soft parts
of a conversation. It is getting very irritating. Has anyone had this
happen? How do I get rid of it?
Thanks
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
205 N. Mt. Shasta Blvd. Suite 100
Mt. Shasta, CA 96067
Could you post a link?
Thanks!
On Apr 13, 2004, at 2:59 PM, Nick Knight wrote:
Just a quick note, I have been putting together a TAPI driver for
Asterisk, this enables the user to perform things like click to dial
from any TAPI enabled app (such as outlook or ACT etc). At the moment
it
is very
Erick Weber V. wrote:
I'll Like to now how to insert a pause on a SIP string. I have a ATA 186 and
a FXS => FXO converter so I will like to program a extension that can be
dialed and it will dial the ATA extention, wait for dial tone and then dial
the phone number.
You cannot put pauses in any dia
They just updated their software and that seems to have resolved the DTMF
issues, at least for me.
Chris.
On Tue, 13 Apr 2004, Andrew Kohlsmith wrote:
> > 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones?
> > I have had a terrible time getting a hold of anyone over there,
On Tue, Apr 13, 2004 at 04:58:19PM -0400, James Golovich wrote:
>
> # If you really want IAX1 uncomment the following, but it is
> # unmaintained
> #
> #CHANNEL_LIBS+=chan_iax.so
>
Thanks all, I'll move to IAX2 after I've tested the notransfer option.
Tor
__
Yeah, tried this.
Seems that the Zapateller code is not written correctly.
The problem is that if one does this
exten => s,1,Zapateller(answer|nocallerid)
Then the call is answered by Zapateller regardless of the callerID state.
The tones are played if there is no caller id. The problem with th
Since Asterisk 1.0 will be released soon I am wondering if Digium runs
CVS stable on IAXtel and Digium's own PBX. If they are, then great! It
will get a good workout. If not, then WHY? A great way for a product
to get bugs fixed are for the group that codes the product to run it in
a production
On Tue, 13 Apr 2004, Eric Wieling wrote:
> Tor Houghton wrote:
>
> > Well, I use IAX1 between the clients on the inside of the NAT to my local
> > Asterisk, and IAX2 between the local Asterisk and my remote Asterisk.
> > Previously (I have not tried yet with current version), when both clients
Has anyone had any luck with voice recognisation using sphinx, if yes then
could u please send some pointers.
does the eagi app for sphinx really work cause i'v tierd it and sphix dosent
seem to do anything
--
regards
Vikram (http://www.vicramresearch.com)
__
Tor Houghton wrote:
Well, I use IAX1 between the clients on the inside of the NAT to my local
Asterisk, and IAX2 between the local Asterisk and my remote Asterisk.
Previously (I have not tried yet with current version), when both clients
and Asterisk used IAX2, the clients would communicate direct
Mine, too, are fixed...I was in much the same boat as the original
poster...an old DID in 212 worked with DTMF, two much newer ones in 213
and 818 (new markets, apparently) didn't until this morning.
On Tue, 13 Apr 2004 16:02:37 -0400, "Robert Jackson"
<[EMAIL PROTECTED]> said:
> Very cool. I am
Very cool. I am just glad they got it fixed.
-Original Message-
From: Isaac McDonald [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 13, 2004 3:56 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoicePulse Connect Problems
It works now! I did nothing on my end either. VP must mon
Hello all,
Just a quick note, I have been putting together a TAPI driver for
Asterisk, this enables the user to perform things like click to dial
from any TAPI enabled app (such as outlook or ACT etc). At the moment it
is very basic and can only perform click to dial but further
functionality w
It works now! I did nothing on my end either. VP must monitor this list.
Isaac
Robert Jackson wrote:
Just a quick couple of questions for ya'll.
1) Does anyone know if VoicePulse Connect will be supporting dtmf tones?
I have had a terrible time getting a hold of anyone over there, and I
need
On Tue, 2004-04-13 at 20:13, Tor Houghton wrote:
> Well, I use IAX1 between the clients on the inside of the NAT to my local
> Asterisk, and IAX2 between the local Asterisk and my remote Asterisk.
> Previously (I have not tried yet with current version), when both clients
> and Asterisk used IAX2,
Tor Houghton wrote:
On Tue, Apr 13, 2004 at 02:56:48PM -0400, Jeremy McNamara wrote:
Use IAX2, it is a better IAX protocol.
Jeremy McNamara
P.S. If you really must have it, dig thru the channels/Makefile, but
there is zero reason to use it any longer.
Well, I use IAX1 between the clien
On Tue, 2004-04-13 at 16:11, Benjamin Wakefield wrote:
> On the tiki it says for international digits, I can dump them in the
> "digits/au" directory.
> I tried that -- just because, I also made a copy in "au/digits".
> When the queue announces the position I it says:
> -- Started music on hold
Isaac:
The DID is in Ocala, FL. I am not sure if it is a new market or
not. I have not heard anything from their support folks either, but I
just checked the line again and it is working. I did nothing to fix it.
I just don't understand. If you don't mind give yours a try again and
let
Hi!
> On Tue, 13 Apr 2004, Dmitry Mishchenko waxed:
>
> > In other words can I receive information which we are usually getting in CDRs
> > during the time when the call is still active?
>
> Yes, via the manager interface. Check manager.conf, it
> lets * talk on port 5038.
The other option is
On Tue, Apr 13, 2004 at 02:56:48PM -0400, Jeremy McNamara wrote:
>
> Use IAX2, it is a better IAX protocol.
>
>
> Jeremy McNamara
>
>
> P.S. If you really must have it, dig thru the channels/Makefile, but
> there is zero reason to use it any longer.
>
Well, I use IAX1 between the clients on
Tor Houghton wrote:
Hi,
I just upgraded to the recent CVS, and IAX1 no longer seems to be available.
Is there a way to reenable it?
Use IAX2, it is a better IAX protocol.
Jeremy McNamara
P.S. If you really must have it, dig thru the channels/Makefile, but
there is zero reason to use it any
> Another thing to try is to disable call waiting on the
> [EMAIL PROTECTED] phone (if call waiting is enabled, it's doing
> what you've asked it to)...
>
Yep, except on the Polycom, we have found no way to disable
call-waiting.
WW
Willy Wouters
ypOne Publishing
___
> 2) I think you should have been looking at incominglimit,
> not outgoinglimit, or possibly both of them together in
> some combination.
>
Another perspective issue. Apparantly 'incoming' means into
the [*] box, and outgoing is leaving the [*]. In any case,
I tried both, but 'outgoing' is co
Hi,
I just upgraded to the recent CVS, and IAX1 no longer seems to be available.
Is there a way to reenable it?
Tor
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update optio
Scott Laird wrote:
There shouldn't be much that needs tuned, unless your network is
overloaded and dropping packets. If that's happening, then you're
going to need to dig in and take a look at QoS on Linux *and* on your
switches and routers, but odds are that won't be a problem on most
LANs.
Andrew:
It didn't work, the problem is that * stays on priority 1 until you hangup
and the it pass to priority 2 so what I think is that it has to be all in
the priority 1 line
Hope we can figure it out
Erick
- Original Message -
From: "Andrew Thompson" <[EMAIL PROTECTED]>
To: <[EMAIL P
Andrew
Thanks for your answer
I'll test this conf an I'll post it so you know if it works
Thanks
Erick
- Original Message -
From: "Andrew Thompson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, April 13, 2004 12:48 PM
Subject: RE: [Asterisk-Users] FXS => FXO Converter Pro
Robert Jackson wrote:
Just a quick couple of questions for ya'll.
1) Does anyone know if VoicePulse Connect will be supporting dtmf tones?
I have had a terrible time getting a hold of anyone over there, and I
need this functionality before I can migrate to * completely.
2) Are there currently a
I have 2 asterisk systems connected with an iax2 trunk. The first has
SIP phones and x100 line cards, the second at a remote location has a
TDM with zap extensions. When calls are parked by the zap extensions at
the second system, the calls are parked on the second system so users at
the first serv
I am confused as well. They also made it clear that the contract terms
included recouping the cost of the ConnectReach for them, so I doubt
that TWTC is offering it at no extra cost. My contention with that, of
course, is, "why not take my ConnectReach and give it to someone else...
you already hav
Erick Weber V. wrote:
> Hello:
>
> I have a ATA 186 and a FXS => FXO converter so I will like to program
> a extension that can be dialed and it will dial the ATA extention #,
> wait for dial tone and then dial the phone number.
Unfortunately I don't believe there is a concept of "wait for dia
Hi,
i install the Asterisk PBX on a linux machine with i4l to connect to PSTN
(EuroISDN). And i configure a very simple dial plan in extension.conf.
After this, i connect with a SIP program to asterisk and would call my
cellular phone, but got this error:
-- Executing Ringing("SIP/ACzerniak
> 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones?
> I have had a terrible time getting a hold of anyone over there, and I
> need this functionality before I can migrate to * completely.
Works just fine for me. Don't send in-band DTMF if you're not using the
alaw/ulaw/sli
That would be a valid configuration, and yes yellow is an option for setting a yellow
alarm when no channels are open.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel
Sent: Tuesday, April 13, 2004 10:57 AM
To: [EMAIL PROTECTED]
Subject: RE: [
The lastest CVS's versions (both stable and head), the 'r' option in
app_dial doesn't work with SIP and Re-invites. I've heard reports that it's
not working with IAX2 either.. I'm using Cisco gateway and cisco ATA's and
I am doing re-invites, and it's worked up till this point.. What's going on?
The GS phones do not currently support conferencing on the phones using
the conference button. You'll probably have better luck setting up a
conference room, help with which I'm absolutely worthless... The
on-phone conferencing should be addressed in a future GS firmware
revision.
HTH,
Ryan
On
Il 17:28, martedì 13 aprile 2004, Robert Siedl ha scritto:
> Hi List,
>
> I have asterisk running on my server and work with 2 cisco ata und 1x
> snom device. I can intern call it´s fine. But wenn i make a extern call,
> I have many quality troubles. The extern user hear me good, but I hear
> him b
Assuming that it is a Welltech gw, the setting is peer-to-peer mode.
Jorge
John Bittner wrote:
Hi,
Anyone get the Immix C3-FXO Sip gateway to work with asterisk. I have it
working for outbound calls but cant get it to work for inbound calls. The
unit has an built-in greeting and it keeps picking
On Fri, 2004-04-09 at 02:56, Alessio Focardi wrote:
> HI,
>
> quick and simple question: is it possible to use inband dtmf with g729?
Absolutely not.
> What I would like to do is to have sip clients connected to asterisk and a zaptel
> card to make pstn phone calls.
>
> My concern is to allow
Hello:
I have a ATA 186 and a FXS => FXO converter so I will like to program a
extension that can be dialed and it will dial the ATA extention #, wait for
dial tone and then dial the phone number.
Thanks
Erick
___
Asterisk-Users mailing list
[EMAIL
Just a quick couple of questions for ya'll.
1) Does anyone know if VoicePulse Connect will be supporting dtmf tones?
I have had a terrible time getting a hold of anyone over there, and I
need this functionality before I can migrate to * completely.
2) Are there currently any problems with inbou
Since * does video over sip has anyone tried to configure voicemail2 to be able to
leave a video message?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Stephen Davies [EMAIL PROTECTED] wrote:
> > Has anybody got any experience using an X100P on an NTL phone line in
> > the UK (I'm in an ex Cable & Wireless area if that makes any
> > difference).
> >
> Indeed the call end termination doesn't work on an NTL line. I'm not
> so sure it works too we
On Tue, 13 Apr 2004, Dmitry Mishchenko waxed:
> In other words can I receive information which we are usually getting in CDRs
> during the time when the call is still active?
Yes, via the manager interface. Check manager.conf, it
lets * talk on port 5038.
--Chris
--
Chris Maj, Rochester
cma
According to voip-info.org,
"3 way calling: Normally implemented by the phone"
I am using a Grand Stream 100 and not able to make this work. I can dial out
to 1st number then with the flash button I am able to dial out again to a
2nd number. I am not able to bind them together into 1 conversation
On Mon, 12 Apr 2004, Dragan Mickovic waxed:
> Is it possible for asterisk to do an sql query in order to
> get the member list of a call queue?
No, you will have to write code besides SQL in order to do
it. To go the C route, try modifiying app_queue. To use a
different language, you could code
Duane wrote:
Adam Goryachev wrote:
I am in Australia, which I think expects callerid at a different time to
other countries Although other people have told me callerid is
working correctly for them
Forgot to mention there is a patch for this, but it won't patch
cleanly against current
Alessio Focardi wrote:
HI,
quick and simple question: is it possible to use inband dtmf with g729?
What I would like to do is to have sip clients connected to asterisk and a zaptel
card to make pstn phone calls.
My concern is to allow sip users to use digits for call destinations that
do req
If you're reading the digest of the Asterisk-users mailing list:
* Please always strip the parts of the message you're not replying to
- do not resend the whole digest!
* Please always change the subject so it reflects your message
- Do not send a message with a subject of
RE: Asterisk-Use
Hi List,
I have asterisk running on my server and work with 2 cisco ata und 1x
snom device. I can intern call it´s fine. But wenn i make a extern call,
I have many quality troubles. The extern user hear me good, but I hear
him bad (robotics). I work with SIP an ALAW protocol.
Where can i look thi
Good Morning,
I'm working with a queue at the moment and I having trouble with my
digits.
Australia is my example.
On the tiki it says for international digits, I can dump them in the
"digits/au" directory.
I tried that -- just because, I also made a copy in "au/digits".
When the queue announc
Andrew Thompson wrote:
[EMAIL PROTECTED] wrote:
Another observation of something which doesn't work:
exten => 3200,1,Dial(SIP/3200,20,tTr)
exten => 3200,2,Playback(tt-weasels)
exten => 3200,3,Hangup
exten => 3200,102,Dial(SIP/3201,20,tTr)
exten => 3200,103,Playback(tt-weasels)
exten => 3200,10
On Tue, 13 Apr 2004, Andrew Thompson wrote:
> Two things:
>
> 1) Have you looked at call queue's?
>
> 2) I think you should have been looking at incominglimit, not outgoinglimit,
> or possibly both of them together in some combination.
>
In response to [EMAIL PROTECTED], who wrote:
> >
> > The
That makes sense. I've always found the following wording (from the sample
file) confusing, not clear about whether "this" and "this span" referred to
the span connection on the card, or the span itself:
# The timing parameter determines the selection of primary, secondary, and
# so on sync sourc
G'day Adam,
This drove me nuts for a few days just recently (only fixed it yesterday
in fact, and I've not had a chance to update any doco anywhere yet).
On Wed, 14 Apr 2004, Adam Goryachev wrote:
> Actually, now that I look at the file again, I can also see:
> Line: 80
> /* Typically, how man
> -Original Message-
> From: Gregory Junker
>
> On Mon, 2004-04-12 at 11:28 -0400, Troy Settle wrote:
> > At this point, I'm using straight Asterisk, with a a PSTN
> gateway at a data
> > POP passing calls via IAX to my PBX here in the office.
>
> Who is the PSTN gateway provider?
>
I had exactly the same question. There is some really useful documentation
on voip-info.org regarding the extensions.conf syntax.
-Original Message-
From: Brian Cuthie [mailto:[EMAIL PROTECTED]
Sent: 13 April 2004 15:19
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Dial Plan For
Hi Mike-
It sounds like, from the discussion here, that your setup is already
correct. Must be something else causing the occasional red alarm! Should
not occur...
Cheers
Scott
-Original Message-
From: Mike Sturdee [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 13, 2004 7:08 AM
To: S
Hello Pertti,
we would be interessted to, if you could send further informations...
Thanks
Regards
Felix Deierlein
[EMAIL PROTECTED]
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Pertti
Pikkarainen
Gesendet: Samstag, 10. April 2004 11:26
An
Adam Goryachev wrote:
I am in Australia, which I think expects callerid at a different time to
other countries Although other people have told me callerid is
working correctly for them
Forgot to mention there is a patch for this, but it won't patch cleanly
against current CVS...
http://b
Adam Goryachev wrote:
I am in Australia, which I think expects callerid at a different time to
other countries Although other people have told me callerid is
working correctly for them
From what I've been able to guess at Telstra sends a short ~50ms chirp
to the phone, the caller id and th
I have two grandstream budtetone-100 and cisco 7960g phones. When I talk via speaker
phone on either of the phones I get a lot of echo. Any suggestions? Also how do I turn
on the mark echo canceller.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [E
Scott Laird wrote:
> Since the system clock ticks at 1 kHz in 2.6, is there any reason why
> it can't be used (more or less) directly for timing in 2.6? That'd be
> a lot easier then hooking into a 1 kHz USB interrupt source.
Would someone who is familiar with the 2.6 series kernel please comment
On Tue, 13 Apr 2004 [EMAIL PROTECTED] wrote:
> I have no reason to doubt what you wrote, so I already
> changed the timing parameter for my system ;). I did have
> it set as span=1,0,0, ... Now, please, in what scenario
> would one select option '0' and win what scenario would one
> use option
Stephen Davies wrote:
Hi Alex,
Indeed the call end termination doesn't work on an NTL line. I'm not
so sure it works too well on other lines either.
I did some work a while back to add detection of the UK busy/hangup
signal on the line, but I never got it working well enough to depend
on it. Th
have to dial the entire number, like 1 + area code + number. I'd like to
eliminate this by having the user just dial 9 + 7 digit number, and have
asterisk put the 1 + area code (which is in a variable in
extensions.conf)
in front of it prior to sending the request to Voice Pulse. Is this
possib
[EMAIL PROTECTED] wrote:
> Another observation of something which doesn't work:
>
> exten => 3200,1,Dial(SIP/3200,20,tTr)
> exten => 3200,2,Playback(tt-weasels)
> exten => 3200,3,Hangup
> exten => 3200,102,Dial(SIP/3201,20,tTr)
> exten => 3200,103,Playback(tt-weasels)
> exten => 3200,104,Hangup
>
Try something like this:
exten => _9NXX,1,Dial,IAX2/[EMAIL PROTECTED]/1${NPA}${EXTEN:1}
...
-brian
Nik Martin wrote:
In the absence of "The Definitive Guide to Asterisk Dial Plans" book, I'd
like to do something possibly unique with the formatting of extensions in my
dial plan, and am having
Some people have some really wacky ideas about how sampled systems work :-)
Regards,
Steve
Michael Welter wrote:
Just when I thought I couldn't be wrong, I was wrong. We have
woodpeckers that drill into the arial telephone cables, and water
seeps through the holes and partially grounds the tip
Ahh, the {EXTEN:1} must serve to skip the 1st character of the extension,
stripping the 9 off
That's what I needed, thanks!
-Original Message-
From: Austin M. Brower [mailto:[EMAIL PROTECTED] On Behalf Of
Austin M. Brower
Sent: Tuesday, April 13, 2004 9:11 AM
To: Nik Martin
Subject: Re: [A
Dear Nick
Very usefull function driver how can i try it?
Thanks in advance
Dimitri
On Monday 12 April 2004 21:00, Nick Knight wrote:
> Hello all,
>
>
>
> Just a quick note, I have been putting together a TAPI driver for
> Asterisk, this enables the user to perform things like click to dial
On the other end of our PRI line would be a telco switch.
On Mon, 12 Apr 2004, Scott Stingel wrote:
> Mike-
>
> You didn't say what's at the other end of your PRI line, but you might try
> having the other end be the timing sync source. Try: span=1,0,0,esf,b8zs
> instead. Maybe that will help.
Well, Once upon a time, I had problems receiving callerid, and then one
day, Mark was logged into my asterisk box helping with something else,
and I asked him about this, and he showed me a nice tweak to some source
file that made it work.
Some time later, I must have done hundreds of CVS updates,
Don & others,
Thank you for your answer. The fog maybe lifting ;).
The zaptel.conf file has the following in its comments:
#
# The timing parameter determines the selection of primary,
secondary, and
# so on sync sources. If this span should be considered a
primary sync
# source, then give it a v
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