Hello,
On Nov/11/2007, Tzafrir Cohen wrote:
On Sun, Nov 11, 2007 at 08:51:40PM +0100, Carles Pina i Estany wrote:
I also tried using bristuff 0.3y, 0.3s, etc. (is it 0.3 bristuff when
Asterisk is 1.2.X?). Always without any result :-(
Latest bristuff for 1.2 is y-k . See
On Nov 9, 2007 1:11 AM, Philip Prindeville
[EMAIL PROTECTED] wrote:
For someone that's network-aware, but hasn't sat down and plowed through
umpteen SIP-related RFC's and memorized the standards, is there a good
primer on troubleshooting SIP issues?
It's true that using Ethereal (is that what
On Mon, Nov 12, 2007 at 10:20:16AM +0100, randulo wrote:
It's true that using Ethereal (is that what Wireshark is nowadays?)
Ethereal was the original name of Wireshark.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406
I thought Wireshark was the cute Mac OS X name.
On Nov 12, 2007 10:32 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Nov 12, 2007 at 10:20:16AM +0100, randulo wrote:
It's true that using Ethereal (is that what Wireshark is nowadays?)
Ethereal was the original name of Wireshark.
--
And by the way, other than a dump, I'm surprised no one suggested
studying the source code. However, I don't think either would be as
useful as a good paper about these SIP transactions, especially an
asterisk-centric one that could even mention differences in newer
versions of asterisk.
In article [EMAIL PROTECTED],
Vincent [EMAIL PROTECTED] wrote:
Hello
About Record(), ATFT 2nd Edition says that if the filename
contains %d, these characters will be replaced with a number
incremented by one each time the file is recorded.
Problem is, the documentation doesn't
Hi,
I`m using several GXP2020 phones with newest Firmware 1.1.4.18.
Asterisk Version: 1.4.11.
It happens several times that users complain that the caller cannot hear the
transmitted voice from the phones
Also now it happens quite often that callers on hold beeing dropped.
Environment:
Use tcpdump to investigate that
Giedrius Augys wrote:
Hello,
My situation is that , I can't make calls with asterisk, but with
x-lite works fine. Asterisk shows , that successfully registers with
another SIP server, asterisk sends invite, gets trying, and after 30
secs asterisk gets 408
Hello!
I would like to store ISDNCAUSE on automatic call-out campaign
(possibly gives more detail on failed call). How is it possible?
I have tried 'failed' and 'h' extension. No luck. Extension 'failed'
does not know anything about ISDNCAUSE and 'h' extension is not called
at all. Any idea?
I
On Nov 7, 2007 11:43 PM, Gordon Henderson [EMAIL PROTECTED] wrote:
On Wed, 7 Nov 2007, Marek B wrote:
On Nov 3, 2007 9:03 PM, Bert Haverkamp [EMAIL PROTECTED] wrote:
This is generally not possible. The 3G phones (GPRS will be a strech
wrt bandwidth) that do video telephony, do not
Hey Guys,
I have something that just started happening. When my users call each
other on their 5 digit extensions their CallerID is showing as
[EMAIL PROTECTED] (X would be their Ext. and 10.25.2.50 is my
server) Calls in an out to the outside world are fine.
I have scoured my
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
Hello!
I would like to store ISDNCAUSE on automatic call-out campaign
(possibly gives more detail on failed call). How is it possible?
I have tried 'failed' and 'h' extension. No luck. Extension 'failed'
does not know
Hi,
I have the following situation.
I have one account created in my VoIP provider.
Asterisk registers this account with the usage of
'register = ' command in the sip.conf file.
I have a number of aliases assigned to my user which
correspond to different public/PSTN numbers through which I am
Greg Cockburn wrote:
Hi all,
the company I work for has an aging Digital PBX attached to an E1.
This PBX has a few analogue lines, one of which we use a 'traditional' fax
machine on.
I want to upgrade our PBX and Asterisk is almost a perfect fit.
The only problem I can't seem to find
Dave Fullerton wrote:
snip
From what I've heard, I think your best bet is to buy a multi-port
T1/E1 card for asterisk, put your E1 in one port and a channel bank in
the other port, then plug your fax extension into an FXS port on the
This is what we do for our fax machines along
On 12/11/2007, randulo [EMAIL PROTECTED] wrote:
I thought Wireshark was the cute Mac OS X name.
The author changed the name of the codebase last year due to employment changes:
http://en.wikipedia.org/wiki/Wireshark#History
Andrew
--
Linux supports the notion of a command line or a shell for
Artifex Maximus wrote:
Hello!
I would like to store ISDNCAUSE on automatic call-out campaign
(possibly gives more detail on failed call). How is it possible?
I have tried 'failed' and 'h' extension. No luck. Extension 'failed'
does not know anything about ISDNCAUSE and 'h' extension is
In article [EMAIL PROTECTED],
Tomasz Zieleniewski [EMAIL PROTECTED] wrote:
I have the following situation.
I have one account created in my VoIP provider.
Asterisk registers this account with the usage of
'register = ' command in the sip.conf file.
I have a number of aliases assigned to my
Brilliant program, whatever it's called this week.
http://www.wireshark.org/faq.html#q1.2
http://en.wikipedia.org/wiki/Wireshark#History
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To
Debian as well as everyone else 2.6.18-5
Zaptel is branch/1.4 latest.
The issue is not with Zaptel though...IMHO. If you look at /proc/driver/rtc, I find:
rtc_time : 14:34:27
rtc_date : 2007-11-12
rtc_epoch : 1900
alarm: 16:30:31
DST_enable : no
BCD: yes
24hr: yes
square_wave :
Hello all,
We're using a lot of the linksys phones, and while user feedback is
generally positive, the speakerphone leaves a bit to be desired.
For those of you using the polycom desk phones, how do you find the built-in
speakerphone?
Thanks,
Eric
Hello,
I have a strange situation:
I can talk to other SIP phones and via ISDN to the outside, but I don't hear
playbacks or the voicemail messages.
Asterisk show in the cli, that the corresponding files are played, but I hear
nothing at all.
Here is as simple example:
[monkeys]
exten =
For general SIP understanding, there's also Sip Scenario from IPtel (
http://www.iptel.org/~sipsc/ ). It will generate sort of human-readable
web stuff from captures, allowing you to click on the graphical portions
of the call and see the actual SIP packets that correspond to that.
N.
On Monday November 12 2007 9:38 am, Eric Jacksch wrote:
Hello all,
We're using a lot of the linksys phones, and while user feedback is
generally positive, the speakerphone leaves a bit to be desired.
For those of you using the polycom desk phones, how do you find the
built-in speakerphone?
Eric Jacksch wrote:
For those of you using the polycom desk phones, how do you find the built-in
speakerphone?
Excellent!
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
On Mon, 12 Nov 2007 09:38:57 -0500, Eric Jacksch wrote:
Hello all,
We're using a lot of the linksys phones, and while user feedback is
generally positive, the speakerphone leaves a bit to be desired.
For those of you using the polycom desk phones, how do you find the built-in
speakerphone?
On Nov 12, 2007 3:22 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Artifex Maximus wrote:
Hello!
I would like to store ISDNCAUSE on automatic call-out campaign
(possibly gives more detail on failed call). How is it possible?
I have tried 'failed' and 'h' extension. No luck.
On Nov 12, 2007 2:04 PM, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
Hello!
I would like to store ISDNCAUSE on automatic call-out campaign
(possibly gives more detail on failed call). How is it possible?
I have
I`m using several GXP2020 phones with newest Firmware 1.1.4.18.
I had issues with phone locking up using 1.1.4.18. I've now gone to 1.1.4.22
and have eliminated that.
Asterisk Version: 1.4.11.
Me too. Still testing 1.4.13 on a non-production system.
I use on every phone the 1 as local
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
On Nov 12, 2007 2:04 PM, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
Hello!
I would like to store ISDNCAUSE on automatic call-out campaign
On Mon, 2007-11-12 at 15:46 +0100, Stefan Guenther wrote:
Hello,
I have a strange situation:
I can talk to other SIP phones and via ISDN to the outside, but I don't hear
playbacks or the voicemail messages.
Asterisk show in the cli, that the corresponding files are played, but I hear
Michael Graves wrote:
On Mon, 12 Nov 2007 09:38:57 -0500, Eric Jacksch wrote:
Hello all,
We're using a lot of the linksys phones, and while user feedback is
generally positive, the speakerphone leaves a bit to be desired.
For those of you using the polycom desk phones, how do you find the
Stefan Guenther wrote:
Hello,
I have a strange situation:
I can talk to other SIP phones and via ISDN to the outside, but I don't hear
playbacks or the voicemail messages.
Asterisk show in the cli, that the corresponding files are played, but I hear
nothing at all.
Here is as simple
On Monday 12 November 2007 07:54:42 Dave Fullerton wrote:
From what I've heard, I think your best bet is to buy a multi-port
T1/E1 card for asterisk, put your E1 in one port and a channel bank in
the other port, then plug your fax extension into an FXS port on the
channel bank. Since both
This will also happen if there is a zap card installed and
unconfigured in zaptel.conf zapata.conf.
Forrest Beck
[EMAIL PROTECTED]
www.shift8.biz
dCAP
On Nov 12, 2007, at 9:46 AM, Stefan Guenther wrote:
Hello,
I have a strange situation:
I can talk to other SIP phones and via ISDN to
On Mon, Nov 12, 2007 at 07:14:31PM +0200, Atis Lezdins wrote:
Stefan Guenther wrote:
Hello,
I have a strange situation:
I can talk to other SIP phones and via ISDN to the outside, but I don't hear
playbacks or the voicemail messages.
Asterisk show in the cli, that the
On Mon, 2007-11-12 at 15:46 +0100, Stefan Guenther wrote:
Hello,
I have a strange situation:
I can talk to other SIP phones and via ISDN to the outside, but I
don't hear
playbacks or the voicemail messages.
Asterisk show in the cli, that the corresponding files are played,
At 08:38 11/12/2007, Eric Jacksch wrote:
Hello all,
We're using a lot of the linksys phones, and while user feedback is
generally positive, the speakerphone leaves a bit to be desired.
For those of you using the polycom desk phones, how do you find the built-in
speakerphone?
Thanks,
Hello,
I can talk to other SIP phones and via ISDN to the outside, but I
don't hear playbacks or the voicemail messages.
Asterisk show in the cli, that the corresponding files are played,
but I hear nothing at all.
Here is as simple example:
[monkeys]
exten = 99,1,ANSWER()
On Monday November 12 2007 1:50 pm, Doug wrote:
At 08:38 11/12/2007, Eric Jacksch wrote:
Hello all,
We're using a lot of the linksys phones, and while user feedback is
generally positive, the speakerphone leaves a bit to be desired.
For those of you using the polycom desk phones,
Doug wrote:
At 08:38 11/12/2007, Eric Jacksch wrote:
Hello all,
We're using a lot of the linksys phones, and while user feedback is
generally positive, the speakerphone leaves a bit to be desired.
For those of you using the polycom desk phones, how do you find the built-in
Hi,
with some messages the voicemailmain after give me the information
about the call (Days, hours and minutes) it finish.
Whant can I check for solve this problem?
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asterisk-users
Hi
additional information if I am going to wait at least 3 seconds after
the voicemail starts to give me the instruction I am able to listen my
messages.
But why I need to wait?
On Nov 12, 2007 2:28 PM, Il Neofita [EMAIL PROTECTED] wrote:
Hi,
with some messages the voicemailmain after give me
At 13:05 11/12/2007, John Millican, wrote:
Excellent speakerphone. Extremely cumbersome to
configure.
I do not understand how you can say that the Polycoms are Extremely
cumbersome to configure. I find them rather nice. Once you have one
working config it is very easy to copy that
Hi,
Excellent ! For me, Polycom have the best audio.
Just behind, I like also Aastra.
Best Regards,
Francois
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] la part de Eric Jacksch
Envoyé : lundi 12 novembre 2007 15:39
A : Asterisk Users Mailing List -
Am Montag, den 12.11.2007, 15:14 -0500 schrieb Il Neofita:
Hi
additional information if I am going to wait at least 3 seconds after
the voicemail starts to give me the instruction I am able to listen my
messages.
But why I need to wait?
On Nov 12, 2007 2:28 PM, Il Neofita [EMAIL PROTECTED]
Thank you for your answer.
The problem is quite different
for example,
I am leaving a message of 5 seconds
when I call to listen the message , asterisk answer and pass the call
to voicemailmain and it plays the welcome message
now if I press 1 before 3 or 4 seconds the voicemailmain gives me then
In my queue log I see that on the RINGNOANSWER Event I get different
content. Some events soe the ring timeout (15000). Other events show
0. Other yet show 1000 Doens anyone know what 0 means? Did it try to
ring the phone, but it was busy?
Thanks
Doug
Yes. That's supposed to to be the timeout value. In the case where it's
0 are you seeing a call reject or something else?
asterisk wrote:
In my queue log I see that on the RINGNOANSWER Event I get different
content. Some events soe the ring timeout (15000). Other events show
0. Other
Hello,
On Nov 12, 2007 5:52 PM, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
On Nov 12, 2007 2:04 PM, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
Hello,
On Nov/02/2007, Atis Lezdins wrote:
On 11/2/07, Tony Plack [EMAIL PROTECTED] wrote:
We are going to implement MeetMe, but this should still work right?
I had similar issues with 1.4.12 just one time (also topmost zaptel at
are you using Dell PowerEdge servers?
--
Carles Pina i
Hello,
On Nov/12/2007, Tony Plack wrote:
Debian as well as everyone else 2.6.18-5
Zaptel is branch/1.4 latest.
The issue is not with Zaptel though...IMHO. If you look at
/proc/driver/rtc, I find:
periodic_IRQ: no
If you notice, there is no
Hi,
I wish to integrate a Microsoft SQL server with Asterisk for CDRs and
for dialplan routing based on database values, and have this application
scale to a large number of simultaneous calls: The Asterisk: The Future
of Telephony 2nd edition book states that:
‡ The pooling and limit options
In article [EMAIL PROTECTED],
Artifex Maximus [EMAIL PROTECTED] wrote:
If the call is unsuccessful, the Dial command will return, and you can
then check ${ISDNCAUSE} on the next line of your dialplan.
There is no dial command because I only wrote a call file to
On Mon, 12 Nov 2007 09:58:50 + (UTC), [EMAIL PROTECTED]
(Tony Mountifield) wrote:
I'm a little surprised at the variety of band-aid suggestions that have
been posted. All you need to do is refer to show application record,
and you uwill see that the generated filename is available by using
On Sun, 11 Nov 2007 11:18:30 -0600, Eric \ManxPower\ Wieling
[EMAIL PROTECTED] wrote:
You need to look at the files in /path/to/src/asterisk/doc (or /docs, I
don't recall) there is much information there, including a file named
README.variables (1.2) or channelvariables.txt (1.4).
Will do.
On Sun, 11 Nov 2007 13:16:35 -0400, Baji Panchumarti
[EMAIL PROTECTED] wrote:
you can generate your own name using a combo of
STRFTIME() CALLERID() CDR() ( and RAND() if you like )
Thanks for the tip. That's what I'll end up doing, as the filename is
more descriptive than just using a
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, November 11, 2007 8:21 PM
Subject: Re: [asterisk-users] sangoma zaptel patches
On Sunday 11 November 2007 11:07:04
- Original Message -
From: Jonn R Taylor [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, November 10, 2007 5:45 PM
Subject: Re: [asterisk-users] 'Traditional' Faxing
Greg Cockburn wrote:
Hi all,
the
I am also very interested in these scripts.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: Tuesday, 13 November 2007 10:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 'Traditional' Faxing
- Original Message -
From: Bincy K. Philip [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, November 08, 2007 2:13 PM
Subject: [asterisk-users] asterisk and installing chan_h323.so rpm
Hello,
When I tried to install chan_h323-1.0.1-module.i386 RPM i got these
- Original Message -
From: Gopal krishnan [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Saturday, November 10, 2007 12:46 PM
Subject: [asterisk-users] Asterisk direct dialing
Hi,
I am using Asterisk 1.2.24, I have written my dialplan to land
with an IVR with the
You need to provide more information that just that. Maybe a CLI output ? Have
you tested with any other providers ? SIP debug ? Ran a trace ? We aren't mind
readers here.
- Original Message -
From: Jarga Jallow
To: Asterisk Users Mailing List - Non-Commercial Discussion
I wrote this for a client a while back:
[del-all-vm]
exten = s,1,Set(TIMEOUT(digit)=3)
exten = s,2,Set(TIMEOUT(response)=6)
exten = s,3,Background(enter-exten-for-vm-to-delete)
exten = _XX,1,Set(THIER_EXTEN=${EXTEN})
exten = _XX,2,Goto(del-all-vm-confirm,s,1)
exten = i,1,Playback(invalid)
exten =
- Original Message -
From: Bincy K. Philip [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, November 08, 2007 2:13 PM
Subject: [asterisk-users] asterisk and installing chan_h323.so rpm
Hello,
When I tried to install chan_h323-1.0.1-module.i386 RPM i got
No, but if you have a hint, I would love it. This is still plaguing me.
Hello,
On Nov/02/2007, Atis Lezdins wrote:
On 11/2/07, Tony Plack [EMAIL PROTECTED] wrote:
We are going to implement MeetMe, but this should still work
right?
I had similar issues with 1.4.12 just one time (also
Afternoon All,
Today rolled a pre-production box from Trunk back to 1.4.7 (In an attempt to
get a working SCCP channel). During the process Music On Hold appears to
have died (Not, just when calling from a SCCP device, but coming in on SIP
also).
CLI is showing
-- Executing [EMAIL
Hi Vivek,
Thanks for the link. I had a look through and couldn't find anything that
worked. There are no NAT problems as this is all taking place on my internal
network. The rtp.conf is used to configure the ports. There are no firewalls or
gateways in between these devices.
Asterisk is
What format is your music on hold in?
PaulH
On Tue, 2007-11-13 at 15:04 +1100, Nick Brown wrote:
Afternoon All,
Today rolled a pre-production box from Trunk back to 1.4.7 (In an
attempt to get a working SCCP channel). During the process Music On
Hold appears to have died (Not, just when
Does anyone know anything about the Chatterbug product? I can't tell
if it's an ATA with a modem or some sort of LCR proxy or somesuch.
Anyone?
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asterisk-users mailing list
To
It was using the 3 wav's from Freeplay. I have just recompiled and told it
to pull down the ULAW versions, then removed the Wav's however it has made
no difference.
Cheers
Nick
On 13/11/07 3:56 PM, Paul Hales wrote:
What format is your music on hold in?
PaulH
On Tue, 2007-11-13 at
http://www.oldskoolphreak.com/tfiles/voip/chatter_bug.pdf
PaulH
On Mon, 2007-11-12 at 21:07 -0800, Robert Goodyear wrote:
Does anyone know anything about the Chatterbug product? I can't tell
if it's an ATA with a modem or some sort of LCR proxy or somesuch.
Anyone?
Is it possibly a funny zaptel issue?
Paul Hales
AsteriskIT
On Tue, 2007-11-13 at 16:20 +1100, Nick Brown wrote:
It was using the 3 wav's from Freeplay. I have just recompiled and told it
to pull down the ULAW versions, then removed the Wav's however it has made
no difference.
Cheers
Hi All,
I was wondering, what tools are readily available out
there in Asteriskland for me to use in stress/load
testing asterisk box I have in the lab. I want to
observe how my box holds out under heavy/light/medium
load.
Thanks,
Jeng
Jeng Yu wrote:
Hi All,
I was wondering, what tools are readily available out
there in Asteriskland for me to use in stress/load
testing asterisk box I have in the lab. I want to
observe how my box holds out under heavy/light/medium
load.
Try SIPp from HP
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