On Tue, 30 Nov 2004 16:21:02 -0600, Jay Milk [EMAIL PROTECTED] wrote:
2. I'm using Sipura SPA-2000s -- $50/port. Linksys PAP2-NA is also an
option if you can find them -- $30/port.
I'll have to look at the SPA-2000 and the Linksys. Thanks.
3. From what I've read on this list (go google a
what does this warning really mean?
does it have any side effect on my * box? 'cose I've recently had
random seg. faults on my box.
I'm using latest CVS -r v1-0
Dec 1 12:08:42 WARNING[6189]: Avoided deadlock for 'SIP/2502-6303', 10 retries!
Dec 1 12:08:43 WARNING[6189]: Avoided deadlock for
Dec 1 12:08:43 WARNING[6189]: channel.c:495 ast_channel_walk_locked:
Avoided deadlock for 'SIP/2502-6303', 10 retries!
Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked:
Avoided deadlock for 'SIP/2502-6303', 10 retries!
Dec 1 12:08:44 WARNING[6189]: channel.c:495
On Wed, 01 Dec 2004 09:46:14 +, Jean-Michel Hiver
[EMAIL PROTECTED] wrote:
2. What is a good, inexpensive FXS solution?
I simply got a budgeton sip phone. It's simple, it has nice fat buttons
and it sounds fine.
What I've encountered in discussions with home-office folks is they
have
Hi,
I saw your messages
related to the Nortel I 2004. I downloaded the code but I have some
trouble with the installation, could you give me some more
informations...
The trick is that I
cannot compile (the make). It must be because I didn't make the changes in
db.c but after reading
I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over
pstn is ok. Also inbound pstn calls get redirected to Asterisk y.y.y.y
But I want to setup VOIP sessions with local carrier. I added dial-peer
40 for this and enabled ip2ip on this peer. Calls will get routed trough
I would like to let my callers know what time it is before I switch them
to an extension number.
(They should know that it is 3 am in the morning, when they are calling me)
Is there such an application available?
bye
Ronald
___
Asterisk-Users mailing
at the same time I have also this notice log.
this makes my problem more meaningful.
i think it might be a bug inside *. (am i right?)
Dec 1 12:44:46 NOTICE[6189]: Disconnecting call 'SIP/2502-6303' for
lack of RTP activity in 4794 seconds
Dec 1 12:44:47 NOTICE[6189]: Disconnecting call
On Sun, 28 Nov 2004 14:10:29 -0300, Andres Junge [EMAIL PROTECTED] wrote:
Hello.
I have this problem. In my asterisk box, I was running debian woody with
asterisk package from backports.org. Last friday I upgraded from debian
to sarge and change from kernel 2.4.18-1-686 to kernel 2.6.8-1-686,
On Wed, 1 Dec 2004 20:13:16 +1000, Robert Barnes
[EMAIL PROTECTED] wrote:
This has happenned to me now too - so I doubt that your hardware is faulty...
Oops - wcfxs was renamed to wctdm some time ago... Working again now.
RAB
___
Asterisk-Users
Hello,
I just compiled and started Asterisk 1.0.2 following Getting Started
With Asterisk Version 0.1a from http://www.automated.it/guidetoasterisk.htm
I made only one change to default config files - I changed from using
oss to alsa.
I don't have any devices so far.
I started asterisk from
Hello,
Another warning I have.
I just compiled and started Asterisk 1.0.2 following Getting Started
With Asterisk Version 0.1a from http://www.automated.it/guidetoasterisk.htm
I made only one change to default config files - I changed from using
oss to alsa.
I don't have any devices so far.
I
On Tue, 2004-11-30 at 16:23 +0100, Dave Cotton wrote:
It looks like libtiff on certain distributions, certainly Mandrake
Cooker, is broken. I went through a lot of testing with Steve to get
faxes received properly and found that Mandrake had not applied a patch
highlighted on the Hylafax
--On Wednesday, December 01, 2004 12:00 AM -0600 Brent Clements
[EMAIL PROTECTED] wrote:
Ok, so I'm setting up my small office.
I have my asterisk machine setup and I have 3 sip phones connected as my
stations and a 4 port FXO card ready to go(planning on only using 2 lines
currently).
What
Hello,
Yet another warning I have.
I just compiled and started Asterisk 1.0.2 following Getting Started
With Asterisk Version 0.1a from http://www.automated.it/guidetoasterisk.htm
I made only one change to default config files - I changed from using
oss to alsa.
I don't have any devices so far.
Hello,
I just sent it with a wrong title... so once again:
I just compiled and started Asterisk 1.0.2 following Getting Started
With Asterisk Version 0.1a from http://www.automated.it/guidetoasterisk.htm
I made only one change to default config files - I changed from using
oss to alsa.
I don't
On Wed, 2004-12-01 at 11:24 +0100, Tomasz Chmielewski wrote:
Hello,
I just compiled and started Asterisk 1.0.2 following Getting Started
With Asterisk Version 0.1a from http://www.automated.it/guidetoasterisk.htm
I made only one change to default config files - I changed from using
oss
--On Wednesday, December 01, 2004 12:59 AM -0800 Lee [EMAIL PROTECTED]
wrote:
Guess I need to research DID providers a bit more, as I would want a
local number, or to keep my existing number...But I'm in a medium
sized city, and won't be surprised if local numbers aren't available.
Thanks
--
I
On Wed, 2004-12-01 at 11:26 +0100, Tomasz Chmielewski wrote:
[snip]
chan_iax.c:7507 load_module: Unable to open IAX timing interface: No
such file or directory
What does it mean? Is it something to worry about? How to get rid of it?
For these and many other basic questions first search
Hello,
Is there a list of software phones which will work with Asterisk?
For Linux and Windows?
I don't have any hardware yet, and before I buy anything I would like to
know how Asterisk really works (with software phones for example).
Tomek
___
Patrick wrote:
On Wed, 2004-12-01 at 11:26 +0100, Tomasz Chmielewski wrote:
[snip]
chan_iax.c:7507 load_module: Unable to open IAX timing interface: No
such file or directory
What does it mean? Is it something to worry about? How to get rid of it?
For these and many other basic questions first
hi, we have just received our first shipment of
digium cards, FXO + FXS combinations, and collected all the hardware for our
custom clone server which will house our test-bed for asterisk.
I'm based in Dhaka, Bangladesh so you will
understand we may not always be able to get all the
Hei!
Should be something like this:
exten = exten_number,1,Answer
exten = exten_number,2,DateTime()
exten = exten_number,3,Dial(SIP/exten_num,30,)
Your application may vary...
Rennes
Ronald Wiplinger wrote:
I would like to let my callers know what time it is before I switch
them to an extension
On Wed, 2004-12-01 at 12:07 +0100, Tomasz Chmielewski wrote:
What I found on voip-info.org was that I didn't have a working timer -
and I had to load ztdummy module. So I did (modprobe ztdummy), started
asterisk again, but I'm still getting the same error.
Had you actually compiled zaptel?
I noticed that I'm no longer able to receive calls from PSTN to my
SipGate DID number.
I changed the sip.conf and extension.conf as per
http://www.voip-info.org/tiki-index.php?page=Sipgate but the problem
remains...
However, I can receive calls from another sipgate user. The problem is
only
at the same time I have also this notice log.
this makes my problem more meaningful.
i think it might be a bug inside *. (am i right?)
Dec 1 12:44:46 NOTICE[6189]: Disconnecting call 'SIP/2502-6303' for
lack of RTP activity in 4794 seconds
Dec 1 12:44:47 NOTICE[6189]: Disconnecting
Dave Cotton wrote:
On Wed, 2004-12-01 at 12:07 +0100, Tomasz Chmielewski wrote:
What I found on voip-info.org was that I didn't have a working timer -
and I had to load ztdummy module. So I did (modprobe ztdummy), started
asterisk again, but I'm still getting the same error.
Had you actually
On Wed, 1 Dec 2004, Dave Cotton wrote:
On Wed, 2004-12-01 at 12:07 +0100, Tomasz Chmielewski wrote:
What I found on voip-info.org was that I didn't have a working timer -
and I had to load ztdummy module. So I did (modprobe ztdummy), started
asterisk again, but I'm still getting the
Peter Svensson wrote:
On Wed, 1 Dec 2004, Dave Cotton wrote:
On Wed, 2004-12-01 at 12:07 +0100, Tomasz Chmielewski wrote:
What I found on voip-info.org was that I didn't have a working timer -
and I had to load ztdummy module. So I did (modprobe ztdummy), started
asterisk again, but I'm still
Hello,
I had this problem a few months ago on a machine that I did a lot of
recording on. It was caused by slow disk access time. Asterisk would wait
for something to write to disk and basically freeze everything. It would
always eventually happen to the same machine no matter if I wiped it
can you tell
me what i need to do
thanks
Rodney Acosta Coya.
Dpto. Tecnologa de la
Informacin. [EMAIL PROTECTED] Tel:(53)(24) 62 611
-Mensaje original-De: Keith O'Brien
[mailto:[EMAIL PROTECTED]Enviado el: Martes, 30 de Noviembre
de 2004 10:20 p.m.Para:
[EMAIL
Hi List,
I have set up the following in my extensions.conf
; local numbers look like 0262XX
; but must be dialed 262 262XX
exten = _0262XX,1,Dial,IAX2/[EMAIL PROTECTED]/011262262${EXTEN:4}
exten = _0262XX,2,Dial,IAX2/[EMAIL PROTECTED]/011262262${EXTEN:4}
exten =
On Wed, 2004-12-01 at 13:01 +0100, Tomasz Chmielewski wrote:
So the only issue left I have is with this skinny not found when 0.0.0.0
is set in skinny.conf
in modules.conf
noload=chan_skinny.so
--
Dave Cotton [EMAIL PROTECTED]
___
On Wed, 2004-12-01 at 13:37 +0100, Dave Cotton wrote:
On Wed, 2004-12-01 at 13:01 +0100, Tomasz Chmielewski wrote:
So the only issue left I have is with this skinny not found when 0.0.0.0
is set in skinny.conf
in modules.conf
noload=chan_skinny.so
Oops
noload = chan_skinny.so
--
Dave Cotton wrote:
On Wed, 2004-12-01 at 13:37 +0100, Dave Cotton wrote:
On Wed, 2004-12-01 at 13:01 +0100, Tomasz Chmielewski wrote:
So the only issue left I have is with this skinny not found when 0.0.0.0
is set in skinny.conf
in modules.conf
noload=chan_skinny.so
Oops
noload = chan_skinny.so
but i have already an UltraWide 320 Scsi HardDisk installed on my * box.
seems that this won't be the cause of my problem at least.
i think that it should be something betweeen these two errors:
- NOTICE[6189]: Disconnecting call 'SIP/2502-6303' for lack of RTP
activity in 4794 seconds
-
jim wrote:
In this particular case what you'll probably need to do is run a sniffer
program such as Ethereal to see what the IAXy is up to. Whatever state
Yes,
worked perfectly. It had an ip address in
the 192.168.0. network - I assume its default.
Roger.
Hi,
What can it be when I can establish a connection between two Softphones but no
voice is transfered ?
thnx
Hugo,
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To UNSUBSCRIBE or update
Hiyall.
Just been testing my * to FWD connection again over IAX2
Well - I can get to 'tell me' and a UK freephone number so it would
appear its back up again. (ive not yet been lucky enough for anyone to
call me on it yet)
Thanks whoever :)
Wayne.
___
I'm looking to give the SPA-3000 a whirl as I'm having too much
difficulty with the irq sharing thing inside the box.
I'm reading the book but without having one in-hand to play with it
appears a little obtuse at this time. Before I drop down my money can
someone with some hands-on with one of
channel driver for the Cisco SCCP protocol used by ost of their phones
On Wed, 01 Dec 2004 13:46:55 +0100, Tomasz Chmielewski
[EMAIL PROTECTED] wrote:
Dave Cotton wrote:
On Wed, 2004-12-01 at 13:37 +0100, Dave Cotton wrote:
On Wed, 2004-12-01 at 13:01 +0100, Tomasz Chmielewski wrote:
Is it possible to terminate calls via SIP on a
Cisco AS5300? Did anyone do it?How? Do i need an special IOS
version?
Ive beentrying to compile the OpenH323
channel for the last month, but errors still happens.
Thanks in advance.
___
there is some kind of a list on www.voip-info.org. ut, testing with
software phones is rather only a feature test sincesound quality of
softphones is not really comparable to sound uality of hardhones.
Just my 2gr.
On Wed, 01 Dec 2004 11:42:14 +0100, Tomasz Chmielewski
[EMAIL PROTECTED] wrote:
Indeed they do - but if you want numbers, you need to say where you are -
there is no point our company supplying you with UK numbers or toll free, if
you actually US people to call them!
- Original Message -
From: Me [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Yes - all the recent IOS versions support SIP.
- Original Message -
From: Francisco [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 01, 2004 1:43 PM
Subject: [Asterisk-Users] Asterisk + AS5300
Is it possible to terminate calls via SIP on a Cisco AS5300? Did anyone do
So the only issue left I have is with this skinny not found when
0.0.0.0
is set in skinny.conf
in modules.conf
noload=chan_skinny.so
Oops
noload = chan_skinny.so
what's this skinny anyway?
Cisco's VoIP protocol, like SIP, or MGCP, but Cisco developed it
themselves, and it is the default
I am doing that actually, terminating calls via SIP on a
Cisco AS5300, and it is working good.
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de
FranciscoEnviado el: Miércoles, 01 de Diciembre de 2004 10:43
a.m.Para: [EMAIL PROTECTED]Asunto:
[Asterisk-Users] Asterisk + AS5300
On Wednesday 01 December 2004 14:43, Francisco wrote:
Is it possible to terminate calls via SIP on a Cisco AS5300? Did anyone do
it? How? Do i need an special IOS version? Ive been trying to compile the
OpenH323 channel for the last month, but errors still happens.
Yep, works fine, but only
Hi,
What can it be when I can establish a connection between two
Softphones but no voice is transfered ?
thnx
Hugo,
It could be a codec problem, or many other things - can you provide
more detail? What softphone is it? What codec(s) are you trying to
use? If it's a SIP softphone, what's your
Dear List,
I'm running an Asterisk 1.0 server with 4 HFC cards and bri-stuff behind an
Anlagenanschluß with 8 B-channels in Germany. It worked fine with Deutsche
Telekom, but since we switched to Arcor nothing works at all.
After some debugging, I called Arcor helpdesk who told me that they do
On Wednesday 01 December 2004 05:06, Jeff Owen wrote:
I'm sure this might not be the correct place to ask and I have done a
Google but I can't seem to find anything that says there is a VSP that will
work with * in the Ukraine.
I have a friend that lives in Kiev and basically want a phone
Hi all,
I'm diagnosing a problem related to PRI card. I would
like to know the following: assuming I've got a
working PRI card and correctly installed Linux drivers
and a PRI line connected to the card, even without
starting asterisk, shouldn't I hear a ring tone when I
dial the number? I'm
Can you post a sample of your configuration?
(sip.conf, extensions.conf and as5300 dial-peers)
Thanks!
boch.-
- Original Message -
From:
Sebastian Nocetti
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Wednesday, December 01, 2004 10:54
AM
Tracy R Reed wrote:
Some of you may recall that I have been working on building a box to
convert H323 to SIP. After a significant amount of outside help and
slicing and dicing of the ohh323 code to get it to compile on AMD64 we
finally got it working. Now we are working on improving the
Patrick wrote:
I'm running an Asterisk 1.0 server with 4 HFC cards and bri-stuff behind an
Anlagenanschluß with 8 B-channels in Germany. It worked fine with Deutsche
Telekom, but since we switched to Arcor nothing works at all.
After some debugging, I called Arcor helpdesk who told me that they do
Hi!
Some - few - providers are using IAX2 as a protocol. Most are using SIP.
I know that there are advantages of IAX2 regarding multiple connections.
But beside this I'm asking myself (and you all) why I should prefer IAX2
when my SIP connection is working.
Are there differences in the
Hi all,
I'm diagnosing a problem related to PRI card. I would
like to know the following: assuming I've got a
working PRI card and correctly installed Linux drivers
and a PRI line connected to the card, even without
starting asterisk, shouldn't I hear a ring tone when I
dial the number?
Patrick wrote:
Dear List,
I'm running an Asterisk 1.0 server with 4 HFC cards and bri-stuff behind an
Anlagenanschluß with 8 B-channels in Germany. It worked fine with Deutsche
Telekom, but since we switched to Arcor nothing works at all.
After some debugging, I called Arcor helpdesk who told me
We terminate calls on both a 5300 and a 7206. But then we discovered the
Digium PRI cards and we got rid of the 5300. :)
-Matthew
- Original Message -
From: Sebastian Nocetti [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent:
- Original Message -
From: Tomasz Chmielewski [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, December 01, 2004 4:42 AM
Subject: [Asterisk-Users] software phones for Asterisk - is there a
list?
Hello,
Is there a list of
Hello,
--- Tomasz Chmielewski [EMAIL PROTECTED] wrote:
Is there a list of software phones which will work with Asterisk?
See the 'SIP Phones (SIP User Agents)' section here:
http://pernau.at/kd/voip/bookmarks-sip-rtp-ua.html
Regards, Girish
Correct me if I am wrong, but G729 is not distributed with asterisk. By
default it is not available without a license.
http://www.voip-info.org/wiki-ITU+G.729
You have to compile and install the free implementation to test.
Otherwise it won't work...
Kinda hard to get asterisk to use a codecs
Hello,
I'd suggest posting a bug if you haven't already and if you have purchased
any Digium products I would recommend calling them as well. The
ast_channel_walk_locked error is a rare and hard to diagnose problem and the
bug trackers and Digium would be the best people to help you.
It might
df -i - you're out of inodes on that filesystem.
-Ronan
On Wed, 1 Dec 2004, Roger Schreiter wrote:
Hi,
after loading the zaptel driver wct4xxp I have strange
log lines in the syslog:
Out of storage space.
free tells, that more than 900 MB are still free.
Disk space is also available.
Hi,
I notice that SJPhone is registering to asterisk with an expires of 120
secs. However, when I invoke the command sip show peer [sip id]. I notice
that the output indicates the expires 427 and the expiry is 900. Can someone
explain these numbers to me?
I also notice that just before
Heya
I have my * box connected to the Telkom PSTN, and an analogy line with callerID subscription (yes we get charged extra :).
When i call the line, it rings once, a short pause, and then the continued ringing of the phone. Using an external callerID device, it shows the number of the call
Roger Hanson wrote:
- Original Message - From: Tomasz Chmielewski [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, December 01, 2004 4:42 AM
Subject: [Asterisk-Users] software phones for Asterisk - is there a list?
Hello,
Some - few - providers are using IAX2 as a protocol. Most are using SIP.
I know that there are advantages of IAX2 regarding multiple connections.
But beside this I'm asking myself (and you all) why I should prefer IAX2
when my SIP connection is working.
Are there differences in the
Daniel Eboa wrote:
Hello to all,
I have a strange behavior of my asterisk box. I'm running asterisk with
asterisk-oh323 channel driver and everything works very well.
But after few hours, my asterisk stop running and I have to restart it
by typing asterisk -vvvc. Most of the time I connect to my
Asterisk Monitor seems to be working fine. Though the
problem I am having is the two files (in out) muxing.
I added ,m to the string, yet the call records two files
still, and I get the resulting error, at the bottom.
monitor executing ( nice -n 19 soxmix
Repeat after me... WARNING != ERROR. This is just letting you know that it
walked the channel list and did avoid a dead lock by not trying to grab a
lock on a channel that's already locked.
if (ast_mutex_trylock(l-lock)) {
if (retries 10)
Hi
Thorston
It
could be the ver of Asterisk or the card driver, we have not used that
particular card but have had
issues
with that and found that the driver was the problem.
Doug
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Thorsten
I use asterisk-oh323-0.6.3b, pwlib-v1_6_6 and openh323-v1_13_5.
This is the complete error: H245:818c6c0 PWLIB Assertion Fail: file
transports.cxx, line 1637
Thanks.
Daniel
-Original Message-
From: Michael Manousos [mailto:[EMAIL PROTECTED]
Sent: mercredi 1 décembre 2004 16:47
To:
...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20041201/051dca
a7/attachment-0001.htm
--
Message: 8
Date: Wed, 01 Dec 2004 17:46:47 +0200
From: Michael Manousos [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk Process Stop After few
ok, this is my config.
sip.conf
[gw-as5300]
type=friendinsecure=yeshost=xxx.xxx.xx.xx
disallow=allallow=g729allow=ulawcanreinvite=noreinvite=nodtmfmode=rfc2833
extensions.conf
exten = _.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _.,2,Congestion
5300 dialpeer, by
default cisco creates a VOIP
Michael Vogel wrote:
Hi!
Some - few - providers are using IAX2 as a protocol. Most are using
SIP. I know that there are advantages of IAX2 regarding multiple
connections. But beside this I'm asking myself (and you all) why I
should prefer IAX2 when my SIP connection is working.
Are there
On Wed, 1 Dec 2004, Enoch Root wrote:
I'm diagnosing a problem related to PRI card. I would
like to know the following: assuming I've got a
working PRI card and correctly installed Linux drivers
and a PRI line connected to the card, even without
starting asterisk, shouldn't I hear a ring
On Wed, 1 Dec 2004, Steve Underwood wrote:
Patrick wrote:
I'm running an Asterisk 1.0 server with 4 HFC cards and bri-stuff behind an
Anlagenanschluß with 8 B-channels in Germany. It worked fine with Deutsche
Telekom, but since we switched to Arcor nothing works at all.
After some debugging,
In my experience with ast_channel_walk_locked every time this WARNING came
up all audio streams on my asterisk server stopped for between 1-10 seconds.
In my case, this WARNING was the only indication of something going wrong,
there was no other output anywhere that described something going wrong
To all:
I am researching the feasibility of replacing our current PBX (ATT
Partner Plus) with an * PBX. I have purchased an X101P card and I have
* running of a FC2 machine. The X101P is connected to an extension on
our current PBX. Many archives exists regarding X101P cards but I just
need
Have you checked if nice
allso exists?
It tries to move the
soxmix to the background
Van:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Craig Waddington
Verzonden: woensdag 1 december
2004 15:56
Aan:
[EMAIL PROTECTED]
Onderwerp: [Asterisk-Users]
Asterisk Call Monitor and
Please visit http://lists.digium.com to un-subscribe from the list.
bkw
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Rich Adamson wrote:
I've successfuly converted 7940 from call manager firmware version 3 to
SIP 7.3. just last week. You need to upgrade to firmware version 6
first, then upgrade to 7.
Also once you've upgraded the phone, you should remove firmware config
file from tftp server, otherwise the
Hello ,
Ill just started with asterisk and I would
liket to to dial between your two
phones with to cisco ATA 186 , but I have a problem
The two cisco ATA and the server in the same networks
and i have the ring in the phone but iam not able to place a call
Between the twe phone .
Or he has a Channelized T1 with inband signaling.
bkw
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Peter Svensson wrote:
On Wed, 1 Dec 2004, Steve Underwood wrote:
Patrick wrote:
I'm running an Asterisk 1.0 server with 4 HFC cards and bri-stuff behind an
Anlagenanschluß with 8 B-channels in Germany. It worked fine with Deutsche
Telekom, but since we switched to Arcor nothing works at
WipeOut schrieb:
[IAX] ... doesn't have as much of an overhead as SIP..
Overhead _during_ the call or before, when registering, etc.?
I have strange sound problems. I don't know if its a bandwith problem.
(Although I guess its a problem of a computer that is overloaded with tasks)
Bye!
Michael
May be just record words 'hours', 'minutes', 'o'clock' and use
Playback(nowis)
SayNumber(12)
Playback(hours)
Say(and)
Playback(30)
Playback(minutes)
Playback(oclock)
:)
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How about following the very easy-to-understand UNSUBSCRIBE procedure
outlined at the bottom of every message from this list? (Oh gawd, I
sound like Critchfield now :p )
Greg
To UNSUBSCRIBE or update options visit:
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Hello all,
I am looking for a
solution where people can use Asterisk as an answering machine where with a
password they can post voice message to a server in mp3 format. Its for
journalists that they are far from the office like correspondents
can put their job for the final diffusion
Hi Adamson,
Thanks for such a comprehensive answers. Below is some more data for your
feedback. I tried all, but it is still not working.
Any comments and advise based on below data?
0. The System is in Singapore.
1. I have an X100P Generic Clone Card bought over from eBay.
2. lspci output:
Some - few - providers are using IAX2 as a protocol. Most are using SIP.
I know that there are advantages of IAX2 regarding multiple connections.
But beside this I'm asking myself (and you all) why I should prefer IAX2
when my SIP connection is working.
some discussion of this a few months
I have this same problem with Cisco Phones. Someone will try and call an
extension and asterisk will say Can't create SIP channel and I immediately
do sip show peers and the phone in question still has an IP listed.
If I know for fact that my 10 phones will never change IP addresses, how can
I
WipeOut wrote:
Michael Vogel wrote:
Hi!
Some - few - providers are using IAX2 as a protocol. Most are using
SIP. I know that there are advantages of IAX2 regarding multiple
connections. But beside this I'm asking myself (and you all) why I
should prefer IAX2 when my SIP connection is working.
Problem FIX!!!
Thanks
-Tim Schacher
On Tue, 2004-11-30 at 23:16, Mike Benoit wrote:
Do you by chance have another Linux box on the same network? One that
could be running the LISa daemon (network neighborhood browsers),
which often gets installed with Mandrake or KDE.
If you do, disable
The only clue that I can spot on this is the output from zttool where it
says:
Span 1: 1 total channels, 1 configured F1=Details
F10=Quit
I might be very wrong, but that message implies the Clone card is (for
whatever reason) not recognized as a x100p but as something
Hello all,
I am looking for a solution where people can use Asterisk as
an answering machine where with a password they can post voice message to a
server in mp3 format. Its for journalists that they are far from the office
like correspondents can put their job for the final
diffusion
Daniel Eboa wrote:
I use asterisk-oh323-0.6.3b, pwlib-v1_6_6 and openh323-v1_13_5.
This is the complete error: H245:818c6c0 PWLIB Assertion Fail: file transports.cxx, line 1637
Go up to v0.6.4 version of asterisk-oh323 (I guess that you use
Asterisk CVS stable).
Thanks.
Daniel
Michael
On Wed, 2004-12-01 at 10:24 -0500, Gregory Junker wrote:
How about following the very easy-to-understand UNSUBSCRIBE procedure
outlined at the bottom of every message from this list? (Oh gawd, I
sound like Critchfield now :p )
Ohh noo, now you know that it doesn't take someone being mean or
On Wed, 1 Dec 2004, Steve Underwood wrote:
Peter Svensson wrote:
Maybe he has NFAS (Non Facility Associated Signalling) where the D channel
on one of the BRI lines handles the signalling for the B channles on all 4
BRIs.
I think NFAS would be a pretty unusual thing for BRI. However, he
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