You can not rotate logs with out dropping calls, and if logs get a
little over 2Gbs Asterisk will crashes...
So I could figure out the average time between crashs just by log level
and call volume! LOL
This is with out running into a single bug. smile Thankful I can
restart Asterisk from
time
I think the key idea is to help newbies along as much as possible so they
don't have to revert to the list to obtain answers to their questions. This
will reduce list bandwidth, possibly significantly.
I see that we already have a four line digium footer on each and every
message. With
As a newcommer I can say that saying things like Check the archive and
whatnot do not help at all when your first exposure to the subject thread is
someone saying It's already been answered, check the archive and that
message is 6 months old! Worst of all there are no hints on searching for
Mark Spencer wrote:
Why don't we just add it on the DIgium list server, wouldn't that make
more sense, to have a single place for all list memberships?
Yes, please. Doing that makes it easier to find it.
/Olle
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On Wed, Nov 19, 2003 at 06:56:04PM -0600, Rich Adamson wrote:
I've got an X100P a cisco 7960. if i call from an analog line via the
x100p to the cisco, there is an overly audible echo on the cisco. If i
make a call from a cisco to cisco, there is no echo. zapata.conf has
Ray Burkholder wrote:
I think the key idea is to help newbies along as much as possible so they
don't have to revert to the list to obtain answers to their questions. This
will reduce list bandwidth, possibly significantly.
I see that we already have a four line digium footer on each and every
Mark has been very emphatic about call features not
belonging in the dialplan.
Hmm.. I read this message, and the couple that came after it and still have
mixed thoughts..
My initial thought was that it didn't make sense - I'd rather have control
over which codes are used to activate which
www.loligo.com/asterisk/
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tddy
Sent: Thursday, November 20, 2003
5:01 AM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Help
configuring CISCO 7960
Yes, I went to cisco site but I
could
At 21:17 19-11-2003 +0100, you wrote:
Looking forward to more of these examples, Florian!
*grin*
I understand I have to find more information on the local channel construct
where you use @pbx for some reason I do not understand...
/Olle
Oh, thats just a matter of convenience on my part. It could
Robert G. Werner wrote:
Btw, I encourage those of the Wiki readership who can spell their way
out of a wet paper sack to not hesitate in fixing typos. Mis-spellings
really do make docs somewhat suspect, to some types of people. Not me,
of course. ;-)
Thank you! My native language is not
Hi,
At 16:48 19-11-2003 -0600, you wrote:
And you will have control of your devices. The problem is that many
of the call features are intrisically related to how the channel type
signals to devices. Hence, the functionality needs to be at the
channel driver level.
Actually, I don't agree. It
Hi again all,
I have searched the list for help with my problem but I can´t find an
answer. I only manage to get one port of my TDM400P card working.
When I do dmesg I get following, seems like four discovered ports:
---
Zapata Telephony Interface Registered on major 196
PCI: Found
JanM wrote:
Hi again all,
I have searched the list for help with my problem but I can´t find an
answer. I only manage to get one port of my TDM400P card working.
When I do dmesg I get following, seems like four discovered ports:
---
Zapata Telephony Interface Registered on major 196
Hi all,
I have purchased few TDM400P from Digium, but it happened that one of my my board and 3 cards went out of order and yesterday another working FXS module is out of order ... i.e. I am having ProSlic Errors and light for that module is not comming up at the back of the card. I am using
On Thu, 20 Nov 2003 00:47:08 -0500, Dorian Gray wrote
I yammered:
of public resources such as this list. put that FAQ in the list
subscribe welcome message or the list sig or the asterisk README or
handbook or all of the above...
er, in case it wasn't obvious: s/that FAQ/a link to that
Hi!
It looks like RTP has a real problem with timing if it is not receiving
RTP packets. If the outside call that is placed on hold is not generating
any audio, the sip/fxo gateway does not send * RTP packets.
Is this valid?
Yep, unfortunately. That's why for example in X-Lite you'll need to
It seems to me that ITSP's like to use a US dialing code eg 1-xxx
Wouldn't it be cool to have an Internet dialing code??
I don't know what the structures are or how the allocations work but it
would be so cool to know that 1-xxx was USA , 44-xxx was UK and yy-xxx
was an internet phone.. That
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michiel Betel
Sent: den 20 november 2003 11:24
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Still TDM400P problem
JanM wrote:
Hi again all,
I have searched the list for help with
Next configuration must work:
zaptel.conf
fxoks=1-4
loadzone=fi
defaultzone=fi
Zapata.conf
[channels]
group=1
context=internt
signalling=fxo_ks
channel=1-4
Srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de JanM
Enviado el: jueves, 20 de
will this port sort out UK caller id?
--- Original Message ---
From: Mark Spencer [EMAIL PROTECTED]
Sent: Wed, 19 Nov 2003 17:58:01 -0600 (CST)
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] 4 Port FXO cards
We *are* making progress, and i have a running prototype, however the
production
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Sergio Serrano Revuelto
Sent: den 20 november 2003 12:41
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Still TDM400P problem
Next configuration must work:
zaptel.conf
fxoks=1-4
Wouldn't it be cool to have an Internet dialing code??
The ITU reserved a code for international networks. It is 882 follow by
two digits to distinguish the networks. Last time I checked it was
difficult to apply unless you were a multinational corporation.
Thilo
I am curious as well if UK caller ID will be supported.
Anyone else out there with the same requirement?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Boardman
Sent: Thursday, November 20, 2003 11:06 AM
To: [EMAIL PROTECTED]
Subject: Re:
Hans-Henrik Andresen wrote:
How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ?
The same way you recieve videos through your fax machine.. :)
No, it can't be done..
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http://www.cisco.com/en/US/customer/products/sw/iosswrel/ps1839/products_feature_guide09186a0080087edb.html
Andrey.
We're having an issue with connecting a Cisco ITS installation to * such
that DTMF tones are passed to *. DTMF tones aren't passed to voicemail or
to any of the interfaces
Hi,
I have a problem with dialling internationals numbers, and I don't now what
is the cause.
I have one asterisk with a e100p card connected to the Telco
(spain/telefonica) and it can dial local and national numbers without
problems but when I try to dial a international number it hangs-up. I
Hans-Henrik Andresen wrote:
How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ?
The same way you recieve videos through your fax machine.. :)
HMM. greate sarcasm.
I had read about a driver for asterisk for voicemodems, that why i'm asking.
So if anyone had tried this, or
Good idea :)
Also, Oftel is planning 055. code specifically to be used for VOIP...
Ta
SJ
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Hans-Henrik Andresen wrote:
Hans-Henrik Andresen wrote:
How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ?
The same way you recieve videos through your fax machine.. :)
HMM. greate sarcasm.
I had read about a driver for asterisk for voicemodems, that why
Hi,
- Original Message -
From: C M [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, November 20, 2003 2:49 PM
Subject: [Asterisk-Users] iaxComm new version installation problem
hi,
i am trying to install iaxcomm-win-20031117.zip in my
windows xp machine. i am really messed
Hi Folks,
I'm trying to stablish a H323 connection from a Linejack/pstngw box
to asterisk.
The connection starts but doesn't complete appearing the following
error in my asterisk console.
H225 Answer: H225Failed to get initial Q.931 PDU,
connection not started.
What does it mean?
THanks
sorry dan,
i downloaded the file from
http://iaxclient.sourceforge.net/iaxcomm/index.htmland
file iaxcomm-win-20031117.zip but it does not seem to
work... it just disappears in the background.. i can
see it running in the task manager thing. and my
computer gets really slow.
cm
--- Dan [EMAIL
Hi Antonio,
This is often a pain with ISDN. What works varies from place to place.
Ahm the wonders of standards :-). Setting the dial plan to international
is probably right. When you do this you may need to drop the 00 prefix,
and start with the country code. Then again, you may not. It
I would be happy with the features working for sip and it doesn't bother me
where they are implemented as long as they work.
Thanks
John
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, November 19, 2003 1:25 PM
Subject: Re:
I'm going to add my two cents to this conversation as its now taken many
turns. This thread has produced quite a bit of good dialog, even though
some of it may not be viewed as such.
I've been playing with Asterisk now for a couple of months both at home
and at work in our Test Environment. I
On Thursday 20 November 2003 02:45, Florian Overkamp wrote:
At 16:48 19-11-2003 -0600, you wrote:
And you will have control of your devices. The problem is that many
of the call features are intrisically related to how the channel
type signals to devices. Hence, the functionality needs to
Hi,
- Original Message -
From: C M [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, November 20, 2003 3:26 PM
Subject: Re: [Asterisk-Users] iaxComm new version installation problem
sorry dan,
i downloaded the file from
http://iaxclient.sourceforge.net/iaxcomm/index.htmland
Hi--
If I have a voicemail box with a number of 1, and say (among other
things) in zapata.conf:
mailbox = 1
group = 3
context = workext
callerid = Steves Extension(999)999-
channel = 6
(assuming I have a TDM 4 port card, and 2 FXO T100P's )
Yes, I really have a mailbox, number 1. From
On Thursday 20 November 2003 01:54, Paul Crick wrote:
Mark has been very emphatic about call features not
belonging in the dialplan.
Hmm.. I read this message, and the couple that came after it and
still have mixed thoughts..
My initial thought was that it didn't make sense - I'd rather
On Thursday 20 November 2003 02:11 am, Ray Burkholder wrote:
I think the key idea is to help newbies along as much as possible so they
don't have to revert to the list to obtain answers to their questions. This
will reduce list bandwidth, possibly significantly.
I see that we already have a
On Wed, Nov 19, 2003 at 11:22:01PM -0800, Andrew Nelson wrote:
As a newcommer I can say that saying things like Check the archive and
whatnot do not help at all when your first exposure to the subject thread is
someone saying It's already been answered, check the archive and that
message is
On Thu, 2003-11-20 at 06:55, WipeOut wrote:
Hans-Henrik Andresen wrote:
Hans-Henrik Andresen wrote:
How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ?
The same way you recieve videos through your fax machine.. :)
HMM. greate sarcasm.
I had read about a
Amen! While -dev and -users may be a little too sparse, perhaps adding a
-business list would be beneficial for discussing those types of issues.
However business-related issues are not so common at this point, so perhaps
a list devoted to NONTECHNICAL discussion (-nontech?) would be
I'm using firmware wb.00.14 without any problems, but I'm pretty sure it's a
development release and
not official as I got it for test from our PQA department at work.
Marius
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Todd
Sent: 18. november
You can not rotate logs with out dropping calls, and if logs get a
little over 2Gbs Asterisk will crashes...
Why not? Why are the logfiles kept open for the entire life of Asterisk?
Hell even my heavily loaded qmail server isn't this braindead in that
regard.
If * can be patched to open,
Lots of people seem to want this, so I've stuck it up here:
- http://almaw.com/ethereal-iax2-plugin-0.1.zip
Note that it currently only does IAX-2. I might expand it to cope with
IAX-1 at a later date, but no promises. It's fairly basic - unzip the
file and follow the README instructions.
It's compiled on Nov 4th.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of ZyXEL -
Marius Ronningen
Sent: 20. november 2003 15:42
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Wifi600 problem
I'm using firmware wb.00.14 without any problems, but I'm
Would a -softphone and -hardphone be too granular? Sometimes I just
don't have the energy to sift through hundreds of messages...
Of course, the danger becomes making it too granular and losing out on
people who can help. I like the helpful nature of most of this list.
I want to thank everyone
Michael Ulitskiy wrote:
Hi,
I have asterisk functioning as SIP to H.323 gateway for local SIP endpoints and I have H.323 to PSTN
gateway (Lucent MAX TNT) connecting my LAN VOIP to PSTN via PRI.
Everything works fine with one exception. I seem to be unable to figure out why I cannot hear
PSTN
So far it seems like the proposed candidates for new lists are:
asterisk-newbies (perhaps a better word?)
Maybe asterisk-install ?
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Hi!
Now given the above, how do we encourage newbies to look for past
answers first?
Answer: By making the FAQ part of the * install and display that very
document on screen after make install. I'd say 95% of the new users
first get * onto their box and THEN start to ask questions.
To my
Mark Spencer wrote:
Amen! While -dev and -users may be a little too sparse, perhaps adding a
-business list would be beneficial for discussing those types of issues.
However business-related issues are not so common at this point, so perhaps
a list devoted to NONTECHNICAL discussion (-nontech?)
I made a patch sometimes ago, that allows this
Look into the asterisk cli:
'show application dial' ...
basically you should be able to do
exten = blah,1,Dial(SIP/blah,30,A(/path/to/file)r)
this patch has been added to cvs, so
you should already have that.
Matteo.
Il mer, 2003-11-19 alle 20:51,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 20 November 2003 04:38, John Todd wrote:
I'm using ATA-186 devices, with RFC2833 DTMF encoding. I am
having problems with routines that input long strings of numbers, in
that I am getting more than a small number of double digit
I'm not sure if I am wording this correctly, but I'll try.
I have a Cisco 2621 w/ a couple FXO and FXS ports. I have a couple cheap
analog phones plugged into the FXS ports. I am able to get * to ring those
phones when a call comes in, but I cannot get the phones to dial out. I
guess it's all
So far it seems like the proposed candidates for new lists are:
asterisk-newbies (perhaps a better word?)
asterisk-nontech
asterisk-biz
The amount of mail on asterisk-users is more than even *I* can read in a
day, and my job is 100% asterisk. There probably is a justification for
a new
On Wednesday 19 November 2003 15:16, John Todd wrote:
On Wednesday 19 November 2003 10:11, John Todd wrote:
after testing with a MGCP phone (Swissvoice ip10s) I found the
following ASTERISK-based codes (VERTICAL SERVICE CODES) to
work - I assume that most of those will also work with
Andrew Kohlsmith wrote:
You can not rotate logs with out dropping calls, and if logs get a
little over 2Gbs Asterisk will crashes...
Why not? Why are the logfiles kept open for the entire life of Asterisk?
Hell even my heavily loaded qmail server isn't this braindead in that
regard.
Maybe
Hi!
How do I use a voicemodem/soundcard to PSTN-gateway - is it possible ?
The same way you recieve videos through your fax machine.. :)
No, it can't be done..
If you skip the voicemodem part it _can_ be done; start with make
samples or take a look at:
I have the following setup..
[extensions]
; all extensions defined here.
exten = 1234,
exten = 1235,
[dial-out]
; PSTN dialout config
ignorepat = 9
exten = _9,
exten = h,
[local]
; phone context in sip.conf is here..
include = extensions
include = dialout
The question is where
works great here!
I can analize each iax2 packets easily now.
good work.
Matteo
Il gio, 2003-11-20 alle 15:46, Alastair Maw ha scritto:
Lots of people seem to want this, so I've stuck it up here:
- http://almaw.com/ethereal-iax2-plugin-0.1.zip
Note that it currently only does IAX-2. I
On Thu, 2003-11-20 at 07:45, Joe Dennick wrote:
I'm going to add my two cents to this conversation as its now taken many
turns. This thread has produced quite a bit of good dialog, even though
some of it may not be viewed as such.
So, the point that I'm trying to get to is that the product
Hi.
Il gio, 2003-11-20 alle 15:55, Philipp von Klitzing ha scritto:
Answer: By making the FAQ part of the * install and display that very
document on screen after make install. I'd say 95% of the new users
first get * onto their box and THEN start to ask questions.
I agree with that.
On 20/11/03 14:58, WipeOut wrote:
I am not sure a newbies list would help all that much, all that would
happen is that they would cross post to both lists and we would get
everything twice..
To a certain extent this is true. Newbie lists also inevitably become
filled with people with less
What's been suggested for a FAQ and other much needed information for such
a _HIGHLY_ technical software product has been proven thousands of times in
the past few months.
I think you are trying to make more of an issue out of this than there
is. I don't think you have seen anyone here
Hi there,
see subject.
I'd like to be able to use the vmbox prompt of VoiceMailMain2 and use
1234 and 4321 to point to the same mailbox. Will it be sufficient to
create a soft link for 4321 -- 1234 in /var/spool/asterisk/default or
will I get myself into horrible trouble?
Background: I
Mark Spencer wrote:
Amen! While -dev and -users may be a little too sparse, perhaps adding a
-business list would be beneficial for discussing those types of issues.
However business-related issues are not so common at this point, so perhaps
a list devoted to NONTECHNICAL discussion (-nontech?)
Hi all-
HELP!
This is actually a revisit of a problem that I had earlier with E400P's at a
customer site. Customer still gets locked up channel problem, but has
learned to live with it (channels clear themselves after several minutes).
The symptoms, which I believe are directly related:
I'm
Hey all...I'm trying to use gnophone to connect to my asterisk box
behind my firewall..I thought I could just setup a tunnel with something
like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone
to connect to localhost:5036 but I never see anything happen on the
asterisk
Forgive my inexperience...but when does a newsgroup, or series of
newsgroups become preferable to a list?
Michael
On Thu, 20 Nov 2003 17:50:41 +1100, Adam Goryachev wrote:
I agree that a nontech list would be fantastic. The only problem I have
with multiple lists is where people post the same
At 3:58 PM +0100 11/20/03, Tais M. Hansen wrote:
On Thursday 20 November 2003 04:38, John Todd wrote:
I'm using ATA-186 devices, with RFC2833 DTMF encoding. I am
having problems with routines that input long strings of numbers, in
that I am getting more than a small number of double digit
This is a password-protected document (CCO account required.) Can
you refer to a non-password protected URL for the sake of the
archives?
JT
http://www.cisco.com/en/US/customer/products/sw/iosswrel/ps1839/products_feature_guide09186a0080087edb.html
Andrey.
We're having an issue with
Hi!
So far it seems like the proposed candidates for new lists are:
asterisk-newbies (perhaps a better word?)
asterisk-nontech
asterisk-biz
Hm... who will answer the newbie questions then? Newbies?
Not sure the -biz part will make sense, but I guess it won't hurt much to
have it and then
It seems to me that ITSP's like to use a US dialing code eg 1-xxx
Wouldn't it be cool to have an Internet dialing code??
I don't know what the structures are or how the allocations work but
it would be so cool to know that 1-xxx was USA , 44-xxx was UK and
yy-xxx was an internet phone.. That
Joseph Finley wrote:
I'm not sure if I am wording this correctly, but I'll try.
I have a Cisco 2621 w/ a couple FXO and FXS ports. I have a couple cheap
analog phones plugged into the FXS ports. I am able to get * to ring those
phones when a call comes in, but I cannot get the phones to dial
So far it seems like the proposed candidates for new lists are:
asterisk-newbies (perhaps a better word?)
asterisk-nontech
asterisk-biz
The amount of mail on asterisk-users is more than even *I* can read in a
day, and my job is 100% asterisk. There probably is a justification for
a new
At 9:37 AM -0500 11/20/03, Andrew Kohlsmith wrote:
You can not rotate logs with out dropping calls, and if logs get a
little over 2Gbs Asterisk will crashes...
Why not? Why are the logfiles kept open for the entire life of Asterisk?
Hell even my heavily loaded qmail server isn't this
Are you using SIP or MGCP for the 7960...?
This will be a big difference not only in the phone configuration, but also
in the settings for *...
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www.bkw.org/~brian/cisco/ata.html
check connectmode and audiomode.. I don't have this problem on mine.
bkw
On Thu, 20 Nov 2003, Tais M. Hansen wrote:
On Thursday 20 November 2003 04:38, John Todd wrote:
I'm using ATA-186 devices, with RFC2833 DTMF encoding. I am
having problems with
Just a had to put in a few points on this...
First, it is correct that there is no cause to be rude, either by repling
rudely or posting without doing any research. I think that a response
directing them to the proper resources is better than not responding at all.
Second, one
At 07:26 AM 11/20/2003, you wrote:
Probably too late to ask for, but for us reversal polarity detection
(far end answer supervision) is very important for billing and pre-paid
purpose.
Don't the X100P cards already support this? I believe it's called KewlStart.
--Ernest
So far it seems like the proposed candidates for new lists are:
asterisk-newbies (perhaps a better word?)
asterisk-nontech
asterisk-biz
Any others as well? If we were to add another list, I *believe* we could
automatically subscribe everyone in -users to -whatever to help seed it a
Let me clarify my feelings:
I believe the API should look something like this:
struct ast_features {
/* Private data for features, which ones are enabled, state
information, etc */
};
/* Apply var/value pair to the feature set, return 0 on success, -1 if
this isn't a
Hi Gurus,
I we seen references to 'codec pass through feature' in the mailing list.
SIP to SIP and SIP to chan_h323 as well. Could someone help me to understand
this feature, or point me to some examples etc.
Appreciate any pointers here.
Thanks a bunch
Sathya
I have the following setup..
[extensions]
; all extensions defined here.
exten = 1234,
exten = 1235,
[dial-out]
; PSTN dialout config
ignorepat = 9
exten = _9,
exten = h,
[local]
; phone context in sip.conf is here..
include = extensions
include = dialout
The question is where
Anyone using a X100P in India? Does it work?
Thanks
-Suresh
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Your inheritied context is including the exten = h,... for dial-out internal
because your sip.conf is pulling both via your local context.
Something like this should fix it:
[local]
include = extensions
exten = _9,,1,Goto(dial-out,${EXTEN},1)
That will only execute the exten = h,... entry
Hi,
This is a minor bugfix release of asterisk-oh323.
The fastStart mode now is working (it was broken in 0.5.6).
Download:
http://www.inaccessnetworks.com/projects/asterisk-oh323
Regards,
Michael.
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Michael,
I've sent all info off-list.
Thanks.
Michael
On Thursday 20 November 2003 09:53 am, Michael Manousos wrote:
Michael Ulitskiy wrote:
Hi,
I have asterisk functioning as SIP to H.323 gateway for local SIP endpoints and I
have H.323 to PSTN
gateway (Lucent MAX TNT) connecting
On Thu, 2003-11-20 at 09:01, Andrew Kohlsmith wrote:
So far it seems like the proposed candidates for new lists are:
asterisk-newbies (perhaps a better word?)
asterisk-nontech
asterisk-biz
The amount of mail on asterisk-users is more than even *I* can read in a
day, and my job is
Hi,
On Thu, 20 Nov 2003 at 08:44, Chris Hirsch wrote:
Anybody have any ideas?
Asterisk uses UDP, but ssh can only forward TCP ports.
cu
Reinhard
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This may be better off on the developer list, but I thought I would see
if I was way off-base before I went there. I am working on a manager
CTI client (currently for windows but with hopes of porting it elsewhere
later).
[Hold/Reconnect]
I have many of the features working. I can originate
thx. it solved my problem. why not put the working app
in the website so that ppl won't get my kind of
problem
cm
--- Dan [EMAIL PROTECTED] wrote:
Hi,
- Original Message -
From: C M [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, November 20, 2003 3:26 PM
Subject: Re:
whatnot do not help at all when your first exposure to the
subject thread is
someone saying It's already been answered, check the
archive and that
message is 6 months old! Worst of all there are no hints
on searching for
this information. You know in such situations it's helpful
On Thu, 2003-11-20 at 09:44, Chris Hirsch wrote:
Hey all...I'm trying to use gnophone to connect to my asterisk box
behind my firewall..I thought I could just setup a tunnel with something
like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone
to connect to localhost:5036
On Thu, 2003-11-20 at 09:51, Michael Graves wrote:
Forgive my inexperience...but when does a newsgroup, or series of
newsgroups become preferable to a list?
Not. Mailing lists are better suited for long term archival
too(opinion).
There has been discussion about this before. Newsgroups are
Chris Hirsch wrote:
Hey all...I'm trying to use gnophone to connect to my asterisk box
behind my firewall..I thought I could just setup a tunnel with something
like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone
to connect to localhost:5036 but I never see anything happen
On Thursday 20 November 2003 09:51, Michael Graves wrote:
Forgive my inexperience...but when does a newsgroup, or series of
newsgroups become preferable to a list?
When hell freezes over or spammers stop spewing on newsgroups,
whichever comes first. Seriously, we've had this discussion before.
On Thursday 20 November 2003 09:38, Philipp von Klitzing wrote:
I'd like to be able to use the vmbox prompt of VoiceMailMain2 and
use 1234 and 4321 to point to the same mailbox. Will it be
sufficient to create a soft link for 4321 -- 1234 in
/var/spool/asterisk/default or will I get myself
Title: Zaptel DAX?
I could swear that I remember seeing some announcement somewhere that Zaptel now supported drop-and-insert across spans on a TE410P, but now I can't find it. Am I imagining this? We just got our TE410 up and running, and if we could cross-connect digital channels with it, I
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