Nonsense.
If you have access to the firmware, they're fantastic phones and the
best phones I've ever used.
If you buy the phones new from an authorised dealer, buying the
Smartnet contract will cost a few dollars and only take a couple of
days to process giving you full access to Cisco's website
yes its possible port 5061 for SIP
On Thu, 15 Jul 2004 smadi wrote :
hi;
does anyone know how can i start asterisk with on a port that is not the default 5060?
thanks
m. smadi
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I am sending this again since I haven't get it back for twelve hours:
When I originate call to IAXComm, more or less one of tree calls fails
for no aparent reason. Originating calls to SIP clients works as
expected. Anybody has similar problems? Is it asterisk or client problem?
Asterisk log:
Jul
On Wed, 2004-07-14 at 14:33, Dameon D. Welch-Abernathy wrote:
I experience a some echo. It can be minimized by adjusting the SPA to
PSTN Gain and PSTN to SPA Gain values to quieter values. I believe it
can range from -12 to 12. 12 is about twice as loud as 6, -12 is twice
as quiet as -6.
I
On Wed, 2004-07-14 at 16:56, Kevin Walsh wrote:
I'll rush out and buy one for use at home as soon as they support the
UK (BT) phone system for Caller*ID and distinctive ringing etc. (as
the SPA-2000 does for UK phone handsets).
The FXS port on the SPA-3000 supports that stuff, but the FXO
Hello All,
I have a very small setup of 4 users and a X100P. Asterisk is currently
running on an Athlon 1800 but the server it is running on is also our
imap/web/mail/development/samba server and we are having a few issues
with asterisk which I believe is down to to many tasks.
What I intend to
Hi all,
I noticed that all incoming calls come from the user [EMAIL PROTECTED], so
I just can't hit the Call button on my SJphone for Linux to return the
call...
Is there any way to configure Asterisk to show the real [EMAIL PROTECTED] ?
Thanks and regards,
Martin
i use a p2 400 here and it has problems with the scheduling but for 1
or 2 calls that would be ok. Depending on the volume you expect at 1
time adress the hardware according. I'd suggest atleast a 1ghz or so
On Thu, 15 Jul 2004 08:11:43 +0100, Simon Chappell
[EMAIL PROTECTED] wrote:
Hello All,
I contacted the Australian company about 6 weeks ago and they told me it was still not
ready and wouldnt be ready for 2-3 months.
-ken
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Martin
Sent: Wednesday, July 14, 2004 9:05 PM
To: [EMAIL
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dear all.
We are currently using either Grandstream BT100 phones or SNOM 200.
The BT100 comes with a 10mbit ethernet port and the snom with 2x100mbit
port
Problem with the SNOM is that they are expensive and I don't really
like their design: often
By the way: To be legal you also need to buy a SIP license (150 US$ list
price). Cisco spare components are sold without any license or software...
It would had been better you had bought a phone with a software license.
Regards
Sascha
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
Hello everybody,
The problem that I had withj loading oh323 module was finally solved thanks to the help of a kind member of this list, it was due to a problem with redhat 9.0 , the update was the magic solution...
Thank you for your support
Best regards,
Soumaya
Créez gratuitement votre
Try to compile with lastest CVS
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Martin
List-Petersen
Enviado el: jueves, 15 de julio de 2004 1:12
Para: [EMAIL PROTECTED]
CC: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] Chan_Capi 0.3.4a error
I have to agree that the hardware is fanatastic ... but the functionality is rather
poor
--- example A ---
receive call A
answer A
Receive call B
answer B
You can't link/transfer these 2 damned calls.
--- example B ---
You first have to push new call before dialing a number.
this seems to be a
Hi friends
I have some doubt in connecting my firefly3rd party softphone from windows machine to
asterisk server in linux .
My asterisk is behind the Port Restricted NAT. I am using STUN server to cross the
firewall.
My STUN server is running in Linux machine.
When my firefly3rd
Hi Chris,
Chris A. Icide wrote:
According to the wiki at voip-info.org, the dial structure for using
oh323 without a gatekeeper is:
OH323/exten@host:port
or
OH323/exten
The second option is valid only in the case where a gatekeeper is used.
NOTE: OpenH323 library v1.12.0 has a bug in the
Uniden-200
- Original Message -
From: Jean-Yves Avenard [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 15, 2004 12:17 PM
Subject: [Asterisk-Users] SIP phones recommendations
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dear all.
We are currently using either Grandstream
Hi,
for SIP account you can use this: http://www.freeworldialup.com/
for a UK number try this: http://www.calluk.comthe numbers are
free.
Regards,
Yiannis
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Johannes van
HulstSent: 14 July
On Thu, 15 Jul 2004 10:08:41 +0200, Michael Devenijn
[EMAIL PROTECTED] wrote:
--- example B ---
You first have to push new call before dialing a number.
this seems to be a detail but 1° why ? and 2° try to migrate users from another
system ...
I don't know how your handsets are setup, but I
Uniden UIP-200
you can follow this thread in the list: Cheap (US$120 or less) SIP Phones
- Original Message -
From: shabanip [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 15, 2004 1:21 PM
Subject: Re: [Asterisk-Users] SIP phones recommendations
Uniden-200
- Original
Consider the Uniden UIP200, I believe it meets all your criteria.
http://www.voip-info.org/wiki-Uniden
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: Jean-Yves Avenard [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 14, 2004 9:47 PM
Subject:
like i said this is a detail look at the other comments
but to clearify my point :
Try to dial a number without pushing a button or picking up the handset. (dial your
number and then pickup the handset or indeed with a dialplan it uses automatically the
speakerphone)
This is so easy to
Seshu Kanuri,
I don't want to look one of those which asserts their work is their own,
otherwise I would not shared it with the community...
By looking at the link you sent to the mailing-list and the comment that
you attached there was nothing to make me believe that you brought
changes... If
Hi,
I'm trying to get zapras do work, I had downloaded the pppd-source and the 2
patches.
I succefull compiled and install the patched version of pppd, but got this
error in message-log
Jul 15 11:43:32 voip1 pppd[9296]: In file /etc/ppp/filters: unrecognized
option 'active-filter'
Jul 15
Also www.voiptalk.org
They can do SIP and IAX.
HTH
Chris
--
Chris
--
E Mail: [EMAIL PROTECTED]
SIP: [EMAIL PROTECTED]
IAXTEL: 17003366726
On Wed, 14 Jul 2004, Dameon D. Welch-Abernathy wrote:
On Wed, 2004-07-14 at 11:41, Johannes van Hulst wrote:
Can
Title: Message
HI,
If you
are looking for a UK number just for testing/personal use, I canprovide
you one.
For
commercial purpose, numbers should be available by middle of
september.
Ta
SJ
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Sascha Knific wrote:
By the way: To be legal you also need to buy a SIP license (150 US$ list
price). Cisco spare components are sold without any license or software...
It would had been better you had bought a phone with a software license.
Regards
Sascha
but it says Callmanager Licence - I
How is the sound quality? Have you ever used a BT headset?? I have
wasted too much money on BT in the last couple of years. If it still
uses analog transmission for audio, I would skip this. I know there are
digital audio devices around... but you still have the normal BT
bandwidth
Having a couple of issues here, but cant seem to find anything to useful in
the WIKI or elsewhere. Here is my setup;
spa2000s and spa1000s -- Asterisk -- Cisco AS5350 -- PSTN
Problem 1: DTMF does not seem to work now for outgoing calls from the
sipuras to the PSTN. I am using rfc2833 for
Trying to get callerID working on a T1 with TE405P using
em_w signaling.
What format does Feature D expect the callerID and the DNIS to
be?
Ex: *008005551212##9991237654# or how should the callerID be
formated?
The provider can provide the ani spill after the wink, before the
call is answered.
I'd be very interested in the outcoming of this.
Greetings, Holger
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If you babelfish yourself through it, http://www.sipgate.de does it.
Why babelfish? Decent people speak german.
:-)
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I just downloaded chan_capi.0.3.4a.tar.gz but it will not compile on my
system (Suse 9.1).
Search the mailing list archive. The answer to your question is right
there.
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Try to dial a number without pushing a button or picking up the
handset. (dial your number and then pickup the handset or indeed with a
dialplan it uses automatically the speakerphone)
This is so easy to make it work but why does Cisco not do it ??
Grandstream can't do it, either.
With my
hello asterisk people ;)
i have a problem disconecting me if i talk to someone thru isdn hfc
based card (from sip phone sipura 2000 to telco), i get diconnected in
2-4 minutes randomly .. ?
anyone has a same problem ?
where to look?
- latest asterisk CVS + bristuff 0.1.0 same with 0.0.2 ..
[EMAIL PROTECTED] wrote:
BUT, I am now able to call both phones at the same time and they
both ring. This sounds like Asterisk is blocking multiple
connections from the same IP Address? How do i go about fixing
this?
Your NAT bindings are probably expiring. You need to have your phones
Well, I have run into a bit of a snag. It seems that once the phone
registers after I have set call waiting = no, on the phone, it changes
back to yes. After a real quick test before the phone re-registers if I
have the phone set to call waiting=no the call still tries to route
through to the
Hello Soumaya,
It's great that you solved the problem.
But I still don't know how to do. What's the
problem with redhat 9.0? Could you tell me more
details?
Thanks a lot
Rui
Fathallah Soumaya wrote:
Hello everybody,
The problem that I had withj loading oh323 module
was finally
ruixun wu wrote:
Hello Soumaya,
It's great that you solved the problem.
But I still don't know how to do. What's the
problem with redhat 9.0? Could you tell me more
details?
Thanks a lot
Rui
as I am the kind member I can tell you also:
get the latest glibc and libssl updates and recompile
Gustavo,
Als je duits nog een beetje goed is heb ik hier een duitse provider voor ons
MVG Han
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Holger Schurig
Sent: Thursday, July 15, 2004 9:40 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Where
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
BUT, I am now able to call both phones at the same time and they
both ring. This sounds like Asterisk is blocking multiple
connections from the same IP Address? How do i go about fixing
this?
Your NAT bindings are probably expiring. You
This message does not indicate major trouble.
It simply means that the STUN protocol is not using message integrity
generation/checking
which is a checksum method using HMAC hashes.
STUN works ok without it.
muralikrishnan lakshmanan wrote:
Hi friends
I have some doubt in connecting my
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 15 July 2004 03:11 am, Simon Chappell wrote:
Hello All,
I have a very small setup of 4 users and a X100P. Asterisk is currently
running on an Athlon 1800 but the server it is running on is also our
imap/web/mail/development/samba
On Thu, 2004-07-15 at 04:53, Hans-Henrik Andresen wrote:
Jul 15 02:43:57 voip1 kernel: Zaptel: Zaptel PPP support not compiled in
It's thrue, I have'nt pathced and compiled the kernel, but Can't find
anything about it - no reademe.
The source code is your friend. Learning to use grep and at
I'm a bit displeased at the way this happened. I received an email from
VoicePulse. Here's some excerpts:
--
We're sending you this important update so you can take advantage of
improvements we've
been making to your VoicePulse Connect! service.
We've been working hard on
Note The previous method for terminating IAX2 calls using Connect!
will
cease to be available at midnight (GMT) on August 15th, 2004. The
message I got was at 1:51 AM EST. That means I was given negative 5
hours and 51 minutes to make this change.
Check your clock. It's still July.
Andrew
[EMAIL PROTECTED] wrote (on Jul 15):
I'm a bit displeased at the way this happened. I received an email from
VoicePulse. Here's some excerpts:
Hm, I've not seen that yet. It looks like good news to me!
Note The previous method for terminating IAX2 calls using Connect! will
cease to be
Note The previous method for terminating IAX2 calls using Connect! will
cease to be available at midnight (GMT) on August 15th, 2004. The
message I got was at 1:51 AM EST. That means I was given negative 5
hours and 51 minutes to make this change.
Is it August yet?
On Thu, 15 Jul 2004 [EMAIL PROTECTED] wrote:
Note The previous method for terminating IAX2 calls using Connect! will
cease to be available at midnight (GMT) on August 15th, 2004. The
message I got was at 1:51 AM EST. That means I was given negative 5
hours and 51 minutes to make this
On Thu, 15 Jul 2004, Chris Glover wrote:
On Thu, 15 Jul 2004 [EMAIL PROTECTED] wrote:
Note The previous method for terminating IAX2 calls using Connect! will
cease to be available at midnight (GMT) on August 15th, 2004. The
message I got was at 1:51 AM EST. That means I was given
can i start asterisk on port 5090 rather than 5060? and if yes how?
m. smadi
muralikrishnan lakshmanan wrote:
yes its possible port 5061 for SIP
On Thu, 15 Jul 2004 smadi wrote :
hi;
does anyone know how can i start asterisk with on a port that is not the default 5060?
thanks
m. smadi
I am pretty CERTAIN that Asterisk is not allowing multiple logins
from the same IP Address. If i sit and watch the CLI sip debug, watch
one phone register and call the other one, it goes through the
Retransmitting to find the phone. Once the other phone re-registers
it rings again, but
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Yager
Sent: Thursday, July 15, 2004 10:31 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoicePulse changes
Note The previous method for terminating IAX2 calls using Connect!
I'm very close to making this work in the crossover wine emulator on
linux. Currently I am getting an error when trying to download the
config directly from an ip address. See attached snapshot for details.
When installing the program I had to choose win2k as the emulated OS.
Regards,
Maron
May want to think about adding this to your dial plan somewhere :-)
exten = s,1,SayUnixTime(,,ABdYIMp)
ken
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Thursday, July 15, 2004 9:24 AM
To: [EMAIL PROTECTED]
Subject:
and the attachment is here :)
Maron Kristófersson wrote:
I'm very close to making this work in the crossover wine emulator on
linux. Currently I am getting an error when trying to download the
config directly from an ip address. See attached snapshot for details.
When installing the program I
Might take a look at sip.conf towards the top:
[general]
port = 5060 ; Port to bind to (SIP is 5060)
I'd strongly suggest you look at /usr/src/asterisk/configs and all
of the samples within that directory. Surprising what can be found
there.
can i start
Hi,
I know we've talked about this phone to death. I have pretty good voice
quality, with and without WEP enabled, using the G729a codec and DLink
Netgear access points.
I am facing one obstacle that is driving me insane.
Does anyone have the call hold feature working? If you do... how did
Managed to set up a
Budge Tone 100 on our VOIP network.
When I place an
outgoing call, I get a dialtone, but no ringtone (ie the sound I hear when the
telephone I have dialed is ringing).
Any
ideas?
Note The previous method for terminating IAX2 calls using Connect!
will
cease to be available at midnight (GMT) on August 15th, 2004. The
message I got was at 1:51 AM EST. That means I was given
negative 5
hours and 51 minutes to make this change.
Check your clock.
FWIW, I've seen several cases similar to this (not with asterisk though)
where the nat box itself had a limit (bug or otherwise, unknown) as to
how many devices behind the box could use a specific port number.
Ahhh, thats right! We have seen this many times before. Change the
source port of
On Wed, 14 Jul 2004, xfastjackx wrote:
I will receive my CISCO 7960G tomorrow. [...] so could please someone
send me the SIP-firmware?
Search the list:
http://www.google.com/search?hl=enlr=ie=UTF-8q=site%3Alists.digium.com+cisco+firmware+sip
--
Lista asterisk em portugues:
Enable RTP. Keep alive problem from your ATA. Check that configuration
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, July 14, 2004 10:57 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] why ata stop
I started having problems with their IAX termination service last night. I
couldn't make any outbound calls but could receive inbound. I made the
changes to the configs that were e-mailed to me and now it is working
fine.
- Jason
I'm a bit displeased at the way this happened. I received an
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Rich Adamson
Sent: Thursday, July 15, 2004 12:26 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] VoicePulse changes
[...]
You should be far more disturbed with their comment about stable
Hi,
I finally got Asterisk installed using the German installation CD from
http://www.asterisk.de.ms.
I got two SIP phones working (SIPPS)
asterisk*CLI sip show inuse
UsernameincomingLimit outgoingLimit
56780 N/A 0
Hi to all,
I successfully installed the astcc module
but I can't connect to the database.
I use for settings the ip address of
my asterisk box, the user name mysql.
When I tryed to create the database
I received an internal error, then I manually created the database but
now it does not connect.
Welcome back to July. How's the future?
There's one rather reliable, albeit not very popular provider with DIDs
in the Philly area: Vonage. Their softphone service works without
problems on my asterisk system, even allowing for multiple simultaneous
incoming calls, eliminating the need for
Remco Barende wrote:
On Thu, 15 Jul 2004, Chris Glover wrote:
On Thu, 15 Jul 2004 [EMAIL PROTECTED] wrote:
Note The previous method for terminating IAX2 calls using Connect! will
cease to be available at midnight (GMT) on August 15th, 2004. The
message I got was at 1:51 AM EST. That means I was
Maron Kristófersson wrote:
I'm very close to making this work in the crossover wine emulator on
linux. Currently I am getting an error when trying to download the
config directly from an ip address. See attached snapshot for details.
Post me the snapshot directly. It won't come through the
Maron Kristófersson wrote:
OOPS! Ignore that last post, I found the snapshot in your next post.
SRB
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Is it possible to configure a couple of SIP accounts for incoming
calls and then forward this calls to different SIP phones?
Like
Registry [EMAIL PROTECTED]
Registry [EMAIL PROTECTED]
Forward 123 to [EMAIL PROTECTED]
Forward 440 to [EMAIL PROTECTED]
Regards,
Han
[EMAIL PROTECTED] wrote:
FWIW, I've seen several cases similar to this (not with asterisk
though) where the nat box itself had a limit (bug or otherwise,
unknown) as to how many devices behind the box could use a specific
port number.
Ahhh, thats right! We have seen this many times
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 08 July 2004 05:45 pm, Stephen R. Besch wrote:
The most recent version of GSConfigure is available at
www.buffalo.edu/~sbesch Several serious bugs that kept the program from
getting started have been ferreted out and corrected with the
I have an asterisk box setup with a T1 card (hooked to a pri from the
telco). Sometimes when people call a number that rings down that pri,
the first ring is really really long, like 3 normal rings put together.
My indications.conf is the default that comes out after you do a make
install. Is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 15 July 2004 10:50 am, Remco Barende wrote:
On Thu, 15 Jul 2004, Chris Glover wrote:
On Thu, 15 Jul 2004 [EMAIL PROTECTED] wrote:
Note The previous method for terminating IAX2 calls using Connect!
will cease to be available at
Hi,
I was wondering if there is an app available that would let me run
queries on MySQL or Postgress database through the extensions.conf file
in Asterisk.
My goal is to be able to run a simple query on a database that could
return a value in an ${var} variable, or even the number or rows that
After having just seen this in -dev and responding to it, I have to
say... PLEASE do NOT cross post. Almost every person on -dev is also on
-user. Post to one place, and wait for your answer. If someone directs
you to the other, then fine. Respect our time and resources by not
doubling up on your
This one is for the archives.
I got a call today that the * at one of my clients was not working. The
switchboard is set up to ring for a while and then the rest of the
phones start up if the switchboard doesn't pick up. This was not
happening. Instead the mobile phone of one of the people
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I know there is probably a page in wiki that answers this but I could not
find it. I was wondering if there is a way that a user can exit the
directory buy pressing a button like * and be returned back to the auto
attendant. What I have in my mind
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
FWIW, I've seen several cases similar to this (not with asterisk
though) where the nat box itself had a limit (bug or otherwise,
unknown) as to how many devices behind the box could use a specific
port number.
Ahhh, thats right! We have
On Thu, 2004-07-15 at 14:13, Joshua McClintock wrote:
I have an asterisk box setup with a T1 card (hooked to a pri from the
telco). Sometimes when people call a number that rings down that pri,
the first ring is really really long, like 3 normal rings put together.
My indications.conf is
My apologies, I was wondering about that.
Brian.
On 15-Jul-04, at 2:48 PM, Steven Critchfield wrote:
After having just seen this in -dev and responding to it, I have to
say... PLEASE do NOT cross post. Almost every person on -dev is also on
-user. Post to one place, and wait for your answer. If
Hi!
I was wondering if there is an app available that would let me run
queries on MySQL or Postgress database through the extensions.conf file
in Asterisk.
Search for res_odbc
http://www.voip-info.org/tiki-index.php?page=Asterisk%20res_config
Cheers, Philipp
Hi,
The national PSTN is built up by a Siemens Eriksson digital PBX
which, in most cases, ends up in analogue interfaces with tone
dialing. It proved a hard job for me to find the most important tone
signals and information messages going out of the PSTN. However, I
dont know where and how to
They are logging in with different ports. Could asterisk be lying and
actually giving them the same port? or getting confused? can i specify what
source port the phone uses on the asterisk box? or is this something that
has to be done on the phone?
The source ports have to be configured on
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 15 July 2004 03:05 pm, Chris Mader wrote:
I know there is probably a page in wiki that answers this but I could not
find it. I was wondering if there is a way that a user can exit the
directory buy pressing a button like * and be
In some (many?) countries, there are analog lines with reversal polarity
as answer supervision. When the called party hangup the polarity back to
normal.
In this case analog lines are reliable as digital ones from billing
point of view.
BTW, reversal polarity is not a obsolete technology. Here
On Wed, 14 Jul 2004, Kris Boutilier waxed:
Is it enough to simply plug an incoming T1 line in to a Digium T100p card or
should I pass the connection through some form of local CSU to provide
isolation, buffering, local diags and so on?
Plug it right in there.
--Chris
--
Chris Maj,
I just added under the sip.conf different ports
[2815551212]
type=friend
secret=cytel
nat=yes
canreinvite=no
host=dynamic
dtmfmode=rfc2833
context=cytelmain
port=5999
stopped/started the server and interesting enoughits not taking that
port.
2815551212/2815 164.66.67.68 D N
On Wed, 14 Jul 2004, Hall, Eric M. waxed:
FC1
What I don't understand is why it works using the -vgcd but not
when just running asterisk ?
Are there any log messages about the mp3 player not being
spawned ? Like Fork failed or unable to spawn mp3player
?
I am unfamiliar with how FC1
On Wed, Jul 14, 2004 at 10:14:19AM -0700, Bob Knight wrote:
[EMAIL PROTECTED] wrote:
From the CLI and during a call I want to be able to:
*** Pulse the outgoing line and record at least 50 ms of the incoming
line.
The pulse waveform must be specifiable as a series of amplitudes
-Original Message-
The bad part is that starting with SP2 on w2k ms EULA has
changed to include
your agreement to let microsoft not only see, what you have
on your computer,
but also install software on it. This has caused a big
corporate hold on
updating beyond SP2. The
+3, Funny
-Original Message-
Maybe if you circle the globe enough times, crossing the
international
date line each time, of course, it would be possible to get to August
15th yesterday ;-)
SRB
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Sorry to ask this question here since it's related to IRC and not
Asterisk, but I am having trouble logging into the #asterisk IRC channel
on freenode and was wondering if anyone else has had this problem and
solved it.
So here's the situation: Whenever I try to login to the #asterisk
channel I
Nathan Alpert wrote:
Sorry to ask this question here since it's related to IRC and not
Asterisk, but I am having trouble logging into the #asterisk IRC channel
on freenode and was wondering if anyone else has had this problem and
solved it.
So here's the situation: Whenever I try to login to the
Hi,
Thanks for you reply.
I download glibc-2.3.2.tar.gz and
glibc-linuxthreads-2.3.2.tar.gz. The configuration
process was fine, no error occured(I typed
glibc-2.3.2/configure --enable-add-ons=linuxthreads).
But there was a strange things happened in make
process. The make process kept
Hi all,
are there any news about bristuff 0.0.3, which compiles against CVS HEAD?
Any informations regarding the timeframe of appearance would be appreciated...
Greetings
Bjoern
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
Sent: Thursday, July 15, 2004 1:00 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] VoicePulse changes
Welcome back to July. How's the future?
There's one rather reliable, albeit
It's got nothing to do with IdentD and everything to do with registering
your nick on the net/node.
Mike ;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Nathan Alpert
Sent: Thursday, July 15, 2004 2:54 PM
To: [EMAIL PROTECTED]
Subject:
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