AUC is Friday at 12:30 PM EDT. See http://x2z.eu for how to join.
http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622
We'll be talking about SIP/IAX providers and I'd like to hear more
about asterisk appliances like the Digium and the new D-Link.
If the Digium guys are around as they usually
Maybe I missed something here. In my understanding, the only parties in the
call at DTMF stage are the originator and Asterisk. The destination is not
in the picture yet. Is this correct? What is the purpose of the said DTMF
sequence? Do you have a sample dial plan?
No, the problem is to
On 5/2/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Wilson Pickett [EMAIL PROTECTED]
Date: Wed, 2 May 2007 15:30:21 +0200
Is there a way to do the following scenario?
1) my asterisk box receives an incoming call from a toll free number
provider such as nufone, voicepulse, etc.
2) It then dials
Is there a way to do the following scenario?
1) my asterisk box receives an incoming call from a toll free number
provider such as nufone, voicepulse, etc.
2) It then dials a number via SIP and outputs a DTMF sequence.
ok, that part we do every day.
3) After DTMF though, is it possible to
Interesting - I wonder if this is due to different models being marketed
in different places? I have a Nokia 7110 from around 1999 - no jacks
whatsoever.
I think it'd ude to Nokia and others realizing they could make money
grow on trees by not allowing standard headsets to be connected but
I have 4-5 different Nokias, none have a 2.5mm jack. Nothing that even
remotely resembles a jack.
The older ones did have 2.5 jacks
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where are the out of office replies when they're needed?
On 4/30/07, Dovid B [EMAIL PROTECTED] wrote:
I love these :)
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 27,
I think you need to explain control the call pass through those
cards a little please.
On 4/27/07, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi all,
I have 2 cards, they are x100p and TDM400p (2 FXO and 2 FXS), in a
server. Is it possible to control the call pass through those cards?
Any
AUC is Friday at 12:30 PM EDT. See http://x2z.eu
Hi,
One of our guests this week will be Jay Phillips to tell us about
Adhearsion. Haven't heard about the open-source Adhearsion? Look here:
http://www.linuxjournal.com/article/9519
Be with us to ask Jay questions. If you can't be there,
On 4/25/07, Stefan Wintermeyer [EMAIL PROTECTED] wrote:
ssh would be a good start. In case you are not familiar with Linux,
just go for a traditional PBX. Not kidding!
I respectfully disagree! Learn linux, you don't need to be a major
guru to install linux, then asterisk. Of course, a friend
Hello again,
Mark Spencer will be joining us for questions on at least one of these
conferences (we've discussed this and he is definitely onboard with
the concept), but his schedule is such that we can't *promise* he'll
be there this week. I haven't heard back from him about today, but
I'll try
Per,
Have you tried the smsq app? That's what I use to send SMS, I don't
think I ever got the SMS app to work for sending. It receives fine.
/usr/bin/smsq --motx-channel=ZAP/1/0809101000 $RECIPIENT message here
works fine here.
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On 4/15/07, Gordon Henderson [EMAIL PROTECTED] wrote:
That looks fine to me. (if a little busy, if this were a production
server, I'd go into the BIOS and disable as much as I could and compile up
a custom kernel)
Also maybe remove the sound card for a test just to see if that makes
any
The 9 may be legacy, but it is somewhat important!
It's also geo or culture specific. I've been in may offices where you
dial 0 for outside. I have a system in our office where there are
provider codes to force use of one specific one, access codes that
will give a ZAP dialtone of one of two
I love this thread, especially when it came to the chicken boner
part of the discussion - brings back NANAE with a smile - and I'm glad
no one found it off-topic, I think it's well worth talking about (the
suit, not the chicken boners) as this may have an effect on some of
what we do.
I hope to
Just one last reminder to please join us at 12:30 PM EDT today, Friday
to talk about asterisk, ask questions, talk about your solutions,
share your experience as users, developpers or providers at the first
weekly Asterisk Users Conference/Podcast:
Check http://x2z.eu for technical info. The
I am starting an asterisk users live conference call on Talkshoe, a
robust voIP conferencing platform I use for several podcasts. Although
I have spoken to Mark Spencer and a Digium VP about this idea, they
have nothing to do with it for the moment. They may wish to come on
board later if enough
Jay,
Just for the record, I own 3 BT102 and all three have stopped working
for various different reasons. This make me think that um... they're
not very good. Two had hardware problems, one of those was minor
(handset cord) and one will not work no matter what firmware I use.
Grandstream tried
Further disclaimer, there
is NO commercial intent behind this initiative. I only hope to bring
members of the user community together.
Someone kindly emailed privately about this. By the above disclaimer
I mean that I myself have nothing to sell in doing this not is it
meant to be a
This old post just saved my nerves! Please remember to post solutions
when you find them as this person did.
SPA-941, swapping in a D-Link switch and DSL modem to replace a
Linksys WAG54-G. This router is notorious for suddently losing the
ability to negotiate bit rates (thanks again, Google).
Dial(Zap/1-1,IAX2/yy:[EMAIL PROTECTED]/xx|60|r)
Looks right to me and the call seems to be accepted by FWD. What
codecs are you using?
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I discussed this with Digium techs, who recommended using the 1.4 version of
Zaptel with the 1.2 version of Asterisk, at least with my hardware
Hi Brent,
It seems like zaptel 1.2.15 and asterisk .16 are working now,
thanks to a quick hint from Tzafrir about changing a Makefile.
I would check to see if you had the latest version of the zaptel library for
the version of asterisk you are trying to compile.
Has this issue been resolved? I'm having the problem now with the code
base downloaded yesterday.
I should have said the code base for zaptel and asterisk I
Has this issue been resolved? I'm having the problem now with the code base
downloaded yesterday.
On 2/3/07, Erick Perez [EMAIL PROTECTED] wrote:
while compiling svn 53132 of asterisk branch 1.2
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3 -Iinclude
On 3/14/07, Kevin P. Fleming [EMAIL PROTECTED] wrote:
There is no need for any 'map'; any Asterisk 1.2.x release should be
usable with any Zaptel 1.2.x release, but of course we'd suggest using
the latest releases of both. There are no API changes or feature
additions (generally) in release
I'm used to seeing the same versioning (maybe I've been gone too long)
Is zaptel 1.2.15 the right one for asterisk 1.2.16 ?
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I have used www.voiptalk.org for a number of years with their IAX2
connectivity and they seem very reliable with no echo issues. They will also
Second that. Not cheap but reliable and been there for years.
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Hello,
A few weeks ago I enabled the dnsmgr. A few days ago I noticed we
could not reach any IAX2 peers in the USA. I did everything I could
think of including a full reboot to no avail. Re-commenting the enable
in dnsmgr.conf and restarting asterisk made things work again.
Have there been
exten =h,2,System(cp /etc/asterisk/callback.info
/var/spool/asterisk/outgoing)
You could run a script instead of the cp command in system and add the
wait in that.
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Hi,
I turned on dnsmgr in 1.2 and it worked for a few weeks. Suddenly, no
iax2 providers were working. All of them were unreachable. My own
fixed ip phones were. I disabled the dnsmgr and now the IAX providers
are working again. No big deal, but it's odd that this happened this
way. Anyone else
we are using asterisk in combination with the voicemail system. I´m just
wondering if it is possible to switch the voicemail to an I am on
holiday mode.
Just use Play(recordedmsg) instead of voicemail
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I'm thinking I have a faulty unit, but I would love to get some debug
information out of it. Can anybody give me any pointers or suggestions
on how to continue from here? I figure I'm calling digium come monday,
but I would like to have it figured out by then.
What phone are you trying to
Have you tried this?
2000 = 1234,User Nametz=eastern24
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How do I program the dialplan in extensions.conf to:
(a) try multiple provider to make an outgoing call based on current
latency between my * box and the different providers ?
(b) have if provider 1 goes down, then someone can still call me at
number xxx- but now come in through provider 2
(b) have if provider 1 goes down, then someone can still call me at
number xxx- but now come in through provider 2 or provider 3 ?
Oops, misread this one, yes you can have fallthrough numbers but this
must happen at the provider end, not yours.
___
On 7/12/06, Martin Joseph [EMAIL PROTECTED] wrote:
On Jul 12, 2006, at 10:18 AM, KC wrote:
I have the same problem before with 2 different providers. We resolved
this
by turning off qualify (qualify=no).
Of course this doesn't fix anything, it just stops the warnings from
showing up...
On 7/10/06, Ryder Brook [EMAIL PROTECTED] wrote:
and learning a lot and the stupid mistake was that the telephone that I was
calling from has caller id blocked. Well, the only satisfaction is that I
You should always have a way to test with a call that you know is
working, such as a cell phone
On 7/9/06, Ryder Brook [EMAIL PROTECTED] wrote:
I have 2 POTs line coming into Asterisk. We have callerid feature from
Verizon on one of the lines.
What interface are the lines connected to?
I am not able to track any CallerID coming in, in the log. I am pretty green
with asterisk, and it's
On 5/31/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Douglas Garstang wrote:
Oh Crud. So, if I want to jump to another extension or context, I have to
specify the full context, extension and priority? I can't specify a label? It's a
bit tricky trying to jump to a specific priority in an
I could be way off here, but I thought FXS ringing was signaled only
by a change in voltage on the pair, so I'm not sure how zaptel could
instruct the hardware device to send a different voltage? I think its
only capability with FXS is to fluctuate the voltage to support
distinctive rings.
On 5/24/06, Andy Jefferson [EMAIL PROTECTED] wrote:
Went to their site today. Site claims they are still in biz. What is
the story? What really happened to Nufone anyway?
The word dead isn't too accurate. If you pronounced dead and were
buried while in a temporary coma, you'd see that. or not
On 5/25/06, Akpome Akpoguma [EMAIL PROTECTED] wrote:
I was actually running record() application, when I pressed the # key to
interrupt the recording, it just doesnt stop
This can depend on features.conf, the codec used, the phone used, the
digitmap of the phone if there is one and several
On 4/18/06, Wes Baehr [EMAIL PROTECTED] wrote:
Well this is disappointing. Time to find somebody else...
From: NuFone Operations
Sent: Tuesday, April 18, 2006 3:44 PM
Subject: NuFone Update: DIDs
Effective 3pm EST Today, April 18th, 2006 Telesthetic, the carrier
snip
I received an email from
On 5/9/06, Alasdair Gow [EMAIL PROTECTED] wrote:
If you are in the same call group *8# should pick it up I think.
It should work if a channel is ringing but not if answered. Some
answering machines have the same problem, you can't pick up once the
message is being recorded (from the same
Hi,
I was wondering if anyone has experience with intercepting an incoming
call. For example, suppose I'm away and see an after hours incoming
call on ZAP going to vmail. I'd like to answer that call. I've seen a
couple of ways to do this, but I'd like to know what anyone out there
is using
Where can I get some asterisk books.. or tutorials..?
Excellent, though slightly dated, starter tutorial:
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
http://www.automated.it/guidetoasterisk.htm
On 5/9/06, Time Bandit [EMAIL PROTECTED] wrote:
Drag the call from the ZAP channel and drop it on your phone.
Interesting, never thought of that. But I'd like to be able to do it
without any computer on though, that is, by dialing an extension to
pick up ZAP/1 in my specific case.
I
anyone using hardware IAX phones with asterisk ?
I've googled on this issue and found several hardware phones which
support IAX protocol, but before paying money I'd like to know more
about what people experiencing with them.
I have had three of them for neary two years. Here's an executive
No, you have to kill the op_server app and restart it
This is incorrect. You can just send it the HUP (Hangup) signal and it
will reload it's configuration files.
Isn't that what HUP does? :)
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On 4/28/06, Matt [EMAIL PROTECTED] wrote:
Well services broke. It's down.. DIDs ring fast busy.Does anyone
know the details of why nufone did not have backup providers? How can
someone lose a contract with a CLEC like that?! Is there more to this
story then we know?
Ok, NOW you can yell
On 4/29/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
I've been playing around with a new system I'm going to install in
another office. In setting up the Polycom's, I accidently used a new
power supply from a new 601 (24VDC) with an 600. The 600 only require
On 4/18/06, Brian Capouch [EMAIL PROTECTED] wrote:
Folks, please.
This thread has the potential to become a torrent.
It belongs on -biz as it has nothing to do with Asterisk itself.
And why all these cock robin the sky is falling statements. Nothing
has been interrupted yet and until it
is it possible to force FOP to reload its configuration files
(op_buttons.cfg and op_style.cfg) while it is working? I tried to click
on the refresh icon but nothing happens.
No, you have to kill the op_server app and restart it
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You forgot need and please
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SPLORF!! You owe me a new keyboard and a clean screen
Here is my password for my bank account: ca$h-this
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At the moment, I can't seem to get more than one IAX client
registered behind NAT... am I correct in my above assumption or have I
missed something ?
I've used multiple hardware IAX phones behind NAT without a problem.
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I have several providers that do NOT allow using your own CID. On one
of these, I suppress outgoing CID simply because I don't want people
calling us back on that number.
When I dial out through this SIP provider to an incoming number of
another SIP provider (for testing dialplan stuff, since
I'm having an apparent issue where caller id name isn't coming through
my IAX2 channels. The name shows up in the asterisk cdr log, but my IAX2
application doesn't receive it. I'm running asterisk 1.2.4.
Just tried it on IAXCOMM with 1.2 and it worked fine.
Is this a known problem or config
This should be called auto-secretary or auto-receptionist or
something like that since it's exactly what a receptionist does.
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While I'm asking about the Polycom ip500, the answers for all phones
where mic/handset/headset levels are adjustable would be of interest
to many I'm sure.
For the ip500, the default value for the handset seems to be
voice.gain.tx.analog.handset=3
I've noticed that echo all but goes away when
I use 3 which is the default on my 501's and 600's
No echo here
Actually, admin docs warn us NOT to change this value, but I am not in
the US. I don't always have echo, but when there is echo it almost
always goes away by lowering the level into the handset (or headset
mic).
On 2/25/06, Anton Krall [EMAIL PROTECTED] wrote:
Does anybody know how to set polycom's default ring volume ? Everytime you
restart a polycom phone, ring defaults to a very low volume setting which is
kind of annoying having to set everytime you reboot.
IIRC, You have to set it in the XML file
On 2/23/06, Anton Krall [EMAIL PROTECTED] wrote:
Ken, Im having problems with the time on my polycoms, it doesn't matter
which sntp server and offset I enter, the phone wont take the offset into
account, Ive tried entering it directly on the phone and also on the .cfg
file but no luck, any
Dont know. All i know is that i had ulaw enabled in * and i was getting
errors relating to iLBC.
The first thing to check is whether the IAXy even does iLBC
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To
I just bought a new IAXy box and am only achieving one way calling.
Both iax.conf and the IAXy support ulaw and gsm. When I try to call,
Does the IAXy now support anything but ulaw or alaw? The original one didn't.
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I've been looking for someone whom speaks both with a Welsh accent and
also the language.
Check this: http://isdnvoice.com he says he has access to a whole
panoplie of Welsh speakers here:
http://isdnvoice.com/services.htm
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But I don't think Digium is in a hurry to implement such a
feature since it forces people to buy more licenses than they really
need to avoid dead calls.
I don't think they're in ahurry either, but I doubt that whatever
their commission on the $10/channel fee is has a big impact on their
Have you seen that 3 Asterisk servers were running during this show ?
François,
I was there (had a coffee with Dave in fact) but was wondering, there
was no official asterisk presence, was there? Maybe we should have
helped organize this as * is a Linux Solution
Right. My original question was about making Asterisk wait a number or
rings (or amount of time) before picking up a Zap line. If the
rings/time were not reached while the line is still ringing, do nothing.
As someone must have already said, it's not a good idea to share lines
with asterisk.
There is an issue here in France with our Siemens DECT phones that
required a patch to change the ring _frequency_. It was given here
ages ago, but now I can't find it.
Shame on me for not coming back!
//{20,7,RING_OSC,0x7EF0}, // changed to
{20,7,RING_OSC,0x7E6C}, // new value for 25hz
for
For a newbie, it's a start, but you are absolutely right. The work we
are doing with RTCP support will help in this, measuring quality per
call. Those of you that wants to test, please go to the bug tracker for
What qualify stats are good for is watching over a very long period
and comparing
However, for a small sub-set of users
that I work with, Skype is a channel that is preferred for audio in
some circumstances, and I feel that it's worthwhile to have some
ability to connect with users who have expressed that preference.
Thanks for your post, John.
I too encounter resistance
Has anyone had any experience with the Linksys SPA-941 when it comes to
multiple line appearences?
This is what the 841 manual says: (maybe the 941 is different?)
The SPA-841 does not support multiple calls on the same Line key. The
corresponding Line key blinks quickly in red on any incoming
You left the attribution off the quote you included with your mail.
That was Dovid Bender, right?
Brian, Yes, sorry... I believe so, yes.
Ira, you're right. At the time it was available it was a monumental
guide. Maybe there will be a newer version RSN? Still, for a number of
questions about
On 1/12/06, Colin Anderson [EMAIL PROTECTED] wrote:
... On my home setup the wife avoids using the SNOM's
because it looks uber-intimidating and things like call transfer, park etc
blows her mind, she doesn't get it. So I dusted off some Vista 350's I had...
(checking thread subject, looking
Please stop plugging the book. Its annoying. We know
its out there.
http://asteriskdocs.org deserves all mentions it receives and the
people behind it like Leif have done a great service to the community.
The entire book is still available online free so why stop plugging
it. For two years, the
I don't think you can beat the Polycom's for design, features, configuration
options and functionality tho. :)
Polycoms (I only have experience with a ip500) have many qualities.
However, I think it's only a matter of time before entries at the
$180-$200 price point begin beating it in many
I want to take this opportunity to thank you all for your emails of concern.
Reports of my death have been greatly exagerated.
This said, what a great singer, how many times have I been typing away
at the Midnight Hour, listening tot hat song in my head.
wp
This is a problem because many of my users are using GXP-2000s with
Early Dial enabled: I need Asterisk to go on rejecting the number with
484 address incomplete until it sees the final * digit.
Can anybody give me a clue how to accomplish this?
If the phone is even entry quality, it should
I second your opinion that Polycom needs to change their policy on this.
Strengthening the Reseller Channels is one of the more nonsensical
justifications for not publicly providing updates for their own product.
Especially in light of the fact that you can easily and legally get
the source
Thanks much, questions about rj11, 2.5mm and quick connector wiring
abound in google search results but to date I hadn't found an answer.
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Does anyone know where the order of the wires on this connector can be found?
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extensions.conf
[context]
exten = s,n,Set(DYNAMIC_FEATURES=zapflash)
exten = s,n,Dial(SIP/,15,tw)
features.conf
[applicationmap]
zapflash = *3,caller,flash,() needed a comma between flash an the
()
I Wonder (aloud) if there'd be a way to send the incoming call to another phone?
I am setting up a phone system for a small office.
The office will have 5-8 phones and a fax line.
There are 4 hunt lines coming into the office.
assumed that I would buy voip phones and connect
all the phones to a private ethernet network.
You can do this and may it would be the natural
It is not problem with outdated DNS cache. It takes approximatelly 2 minutes
Any other thoughts?
I guess you haven't read the earlier messages saying that asterisk has
chached the ip and needs to be restarted if that ip changes.
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i have an asterisk box behind the NAT ,when i try to
send calls through Sip to the voip provider server the
call is answered but in a one way calling,I hear the
voice of the other side just for 4 seconds and then
stop but the call do not hangup.
SOmetimes this can be due to the client using
Official Polycom view seems to be that you shouldn't work at night :)
The phones are crying out for a backlit LCD that only lights when
ambient light is low. I have a cheap radio/weather station with a
large LCD that does that.
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I have used both and at this point I would favor the 942. At least it
has a speakerphone. Its display also has more useful info although I
find it a bit darker but the text is sharper. Buttons on the Poly are
There is only one thing I don't like on the Sipura and that's the way
the phone lines
Is there a syntax I can use to set a variable based on the evaluation
of an expression? I need something that will work in 1.0.9 and 1.2.
Isn't this what you're looking for:
set(VARIABLE=$[NULL${something}=NULL]})
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My requirement is to have an extension 9000 ring on all of the phones'
second line.
I've configured this extension in asterisk (extensions.conf and sip.conf) as
I would any other extension.
Inside the SPA-841 interface, I configured Ext 2 with the appropriate SIP
information and set the line
Is there any ATA that supports iaxy protocol and a codec different
from g711 like gsm or g729 ??
If I understood the question, look at PA168x based ATA. I think they
support those codecs and more.
Here's a site with some info:
http://www.aussievoip.com.au/wiki-PA1688
I installed 1.2 tarball from
http://ftp.digium.com/pub/asterisk/asterisk-1.2.0.tar.gz yesterday.
I had it running for about 24 hours and everything was working
perfectly, ZAP, SIP, IAX.
I did a lot of experimenting with features.conf today. At first
nothing worked. Then suddenly, all the features
But I am still trying to find out what the Shared line appearance is
on these phones?
My impression was that this works on a compatible system to allow
several people to have line 2000 and see its status which I guessed
what what you were trying to do in the first place? As others have
said, I
How can I set up a group of outbound trunks which will rotate use
dependant on how many outbound calls need to be made.
You could do this by writing a simple (but laboriously long) macro to
try the accounts in order, dialing via the first available one.
There would be a dial() command followed
Yes. I believe the Cisco phones do conferencing in the same fashion. I'm
not 100% on whether or not the SPA-841 or the new SPA-941 does it.
The SPA-941 does conferencing and it works exactly like the transfer.
a soft button you hit twice, Conf once to dial the invited 3rd party
and once to
The useers are log in all right, but when i try to have a call between they,
it not work...
Read this
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf
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I would like to manipulate phone call direction to voicemail for lunch,
after hours etc, but am unsure how to do this. Could someone point me to a
howto or quickly explain the concept?
I would recommend checking a database value over the time based
GoToIfTime unless you are always go to and
But Grandstream 100 occasionally loses registration (have anyone found any
solution to this ?) and then AMP's dialparties.agi won't proceed call
even to voicemail, but will end it imediately (cause dialparties checks if
SIP extension is present - otherwise it deletes it from call group...)..
Can anyone please let me know if they have got UK CallerID working using a
X101P?
While you're waiting for a live answer, there are several threads on
this list you could search for.
Try this
http://www.google.com/search?hs=7Adhl=enq=asterisk+uk+calleridbtnG=Search
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