Hi all,
How i can change the CallerId format in plan id?
for the example i can see the value of CALLERID variable like
lateef 110001
I want to let asterisk do in plain id like
lateef
any idea?
--
Thank You,
Code Lover
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Hello Robert,
I have a similar issue with the Aastra 9133i and recorded .wav voicemail
files. The recorded wav is too soft. I need to find a way to boost the
volume level. Does anyone have any solutions or ideas?
I know this is done by setting the handset tx gain: and handset sidetone
On Dec 21, 2005, at 10:26 AM, [EMAIL PROTECTED] wrote:
Regards to All,
I recently setup an Asterisk system ([EMAIL PROTECTED]) and it works like a
charm so
far. It is in a SOHO behind another Linux iptable NAT firewall with no
problems.
Hopefully this isn't too dumb a question, and its the
On Fri, Dec 23, 2005 at 01:28:14PM -0600, Kevin P. Fleming wrote:
C F wrote:
Kevin, are you saying that in 1.2 a peer can make calls to asterisk as
well, so there is a reason to set the context?
If so what is the difference between friend and peer?
Yes. All configuration options
2005/12/23, Rich Adamson [EMAIL PROTECTED]:
I'm really puzzled and I don't know why it is behaving this way. Any hints?Are you sure the phones and system at work are not electronic/digital sets?Yes, I'm sure phones and lines are both analogic (POTS).
2005/12/24, James B. MacLean [EMAIL PROTECTED]:
A problem I had, although different than this, was caused by having/etc/zapata.conf as koolstart when they should have been loopstart.Might be something else to try :).JESI also tried with loopstart in
zaptel.conf and zapata.conf, but without
2005/12/23, Kerry Garrison [EMAIL PROTECTED]:
Possible polarity problem with the
jack.
-KerrySo I should to check I have +48VDC on the phone line (red wire is +). Right?
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Asterisk-Users
Armin,
I changed the dial-string to include flags 'ob' as you mentioned (below)
and now I get the following when I dial a BT phone number
- dial number, get:
Proceeding (in 100) briefly
- after a second or so:
Ringng Destination (in 180)
- double ringing tone:
BT style
Hi,
I try to install asterisk on my laptop. I have two options
for the ISDN card: AVM B1 PCMCIA and Eicon Diva Mobile V90.
Has anyone any experiences using one of that cards?
Kind regards and merry christmas,
Sascha
--
[EMAIL PROTECTED]
___
On Sat, 24 Dec 2005, Michael J. Tubby G8TIC wrote:
I changed the dial-string to include flags 'ob' as you mentioned (below)
and now I get the following when I dial a BT phone number
- dial number, get:
Proceeding (in 100) briefly
- after a second or so:
Ringng
Hi,
I got the newest asterisk (SVN-trunk-r7413) that compiled
fine without any errors or warnings. I got chan_capi 0.4
PRE1 and modified the sources together with a friend so
ina way that no error or warning occurs.
When I try to load chan_capi the following error is printed
and asterisk quits:
On Sat, 24 Dec 2005, Sascha Andres wrote:
Hi,
I got the newest asterisk (SVN-trunk-r7413) that compiled
fine without any errors or warnings. I got chan_capi 0.4
PRE1 and modified the sources together with a friend so
ina way that no error or warning occurs.
When I try to load chan_capi
Hi list:
i have a bad experience with voip providers , Any body
knows a voip provider i can depend on and to trust
with good rates and quality?
__
Yahoo! for Good - Make a difference this year.
Hi,
When I call to my asterisk server, voice
prompts play ok but when it goes to music on hold, sound is all broken. Why is
that, is there some ports which Music on hold uses which are not configured
properly, or there is some other reason.
Zeeshan A Zakaria
On Fri, Dec 23, 2005 at 05:18:01PM -0500, Ken D'Ambrosio wrote:
Hi, all. I like AMP a lot -- I think it's a nifty program, and it makes
a lot of tasks very easy to do. However, as with any GUI, it's hard to
automate what it does.
So: are there any CLI equivalents for the stuff AMP does?
On Sat, 2005-12-24 at 04:31 -0800, jonny hashem wrote:
Hi list:
i have a bad experience with voip providers , Any body
knows a voip provider i can depend on and to trust
with good rates and quality?
best is relative.. There is no single 'best'.
The 'best' provider for you may be different
Hi all :
I need a little help please.
I have a clone X100P board. I have it all set up and working (just
testing so far) for incoming calls from PSTN.
For outgoing to PSTN I have a strange problem.
I dial out OK, the Zap channel answers the SIP channel ok, (But I do not
see a Call bridged
I had the same problem at first. Try adding a w or two before the
${EXTEN}. That makes it wait a little bit before sending the DTMF numbers.
Here is the dial() I'm using:
Dial(ZAP/1/ww${EXTEN})
Try it out and see. Let us know if it works.
Ryan
Hi all :
I need a little help please.
I have
hi:
Iam using voip providers to get international calls,I
provide Callshops with international calls ,my
prefered destinations are
Germany,Srilanka,Iraq,Egypt,Kuwait , i send calls 700
to 1000 minutes daily. My big problem is bad voice
quality that i have experience it with many voip
providers.
On Sat, 2005-12-24 at 09:21 -0800, chawki hammoud wrote:
hi:
Iam using voip providers to get international calls,I
provide Callshops with international calls ,my
prefered destinations are
Germany,Srilanka,Iraq,Egypt,Kuwait , i send calls 700
to 1000 minutes daily. My big problem is bad
Dov Bigio wrote:
Hi,
When I set monitor-format=wav49 on file queues.conf for a queue,
Asterisk records calls at /var/spool/asterisk/monitor. But the file
names it users are the call-ids of the calls.
Is there a way to change that, and use information such as date, time,
agent and queue to
Faris,
Is there a way to have * send save these in an off-server location? Or
have * e-mail them via smtp and then delete them from the server
automatically?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Faris
Raouf
Sent: Saturday, December 24, 2005
Way back I was still on Asterisk 1.0.7, I configured my systems to mix
the incoming and outgoing audio call recordings into one file per call
for both normal calls and queued calls using:
exten =
_9.,1,Monitor(wav49,${TIMESTAMP}-${CALLERIDNUM}-to-${EXTEN:1}-${UNIQUEID},m)
; m option merges
Tom Lynn wrote:
Faris,
Is there a way to have * send save these in an off-server location? Or
have * e-mail them via smtp and then delete them from the server
automatically?
I'm sure there is a very technical way of doing it. For example if I
remember correctly you can set your own script
For the last 2.5 years I've been using Asterisk with a Bell Canada PRI with no
issues. In the last month or so something has changed; I cannot send *any*
caller ID. Incoming works great, and if I place a call through a VOIP
provider the caller ID I'm sending shows up.
I have not changed any
Rsync could happen overnight, but I'm really looking for a solution that
removes the recording from the system so as not to kill my limited
storage. I'll be running astlinux from a 256mg Compact Flash card and
256meg of USB keydisk space for configs and recordings. I need to move
'em off fast.
sound is all broken? WTF is that meant to mean. Does it play or
doesn't it?
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Zeeshan wrote:
Hi,
When I call to my asterisk server, voice prompts play ok but when it
goes to music on hold, sound is all broken. Why is that, is there
Assuming its a SIP based device
[110001]
user=something
allow=whatever
callerid= lateef
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Code Lover wrote:
Hi all,
How i can change the CallerId format in plan id?
for the example i can see the value of CALLERID variable like
lateef
It means that Music on Hold works but listener listens it in bits and
pieces.
Zeeshan A Zakaria
-Original Message-
From: Mark Phillips [mailto:[EMAIL PROTECTED]
Sent: Saturday, December 24, 2005 2:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
WTF is that meant to mean.
Does it play or doesn't it?
Calm down. It probably means that it's breaking up while it is
playing. But let the OP explain... no need to discourage him like
that.
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Andrew Kohlsmith wrote:
For the last 2.5 years I've been using Asterisk with a Bell Canada PRI with no
issues. In the last month or so something has changed; I cannot send *any*
caller ID. Incoming works great, and if I place a call through a VOIP
provider the caller ID I'm sending shows up.
How to pass some parameters to shell script, invoked in CLI through
application system(...)?
I want to do some logging of incoming CID-s to file. Is there some other
method to do this?
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Not in CLI, Invoked in extensions.conf:
exten = s,1,system(/usr/bin/logscript) ;and how to pass some parameters
here?
if I do somenhing like:
exten = s,1,system(/usr/bin/logscript,${CALLERID},pstn)
then I get error.
- Original Message -
From: Pisac [EMAIL PROTECTED]
To: Asterisk Users
Greetings from Lima Peru
Carlos Rojas
On 12/23/05, Guillermo Salas M [EMAIL PROTECTED] wrote:
Xmas is tomorrow at my country.. Merry Xmast to all :)Greetings from Ecuador - South America ;)
On Fri, 2005-12-23 at 19:20 -0500, tracinet wrote: Nothing wrong at all - this is the Merry Christmas
On Sun, 2005-12-25 at 01:08 +0100, Pisac wrote:
How to pass some parameters to shell script, invoked in CLI through
application system(...)?
I want to do some logging of incoming CID-s to file. Is there some other
method to do this?
Other than through system? Or did you want information on
On Sun, 2005-12-25 at 01:22 +0100, Pisac wrote:
Not in CLI, Invoked in extensions.conf:
exten = s,1,system(/usr/bin/logscript) ;and how to pass some parameters
here?
if I do somenhing like:
exten = s,1,system(/usr/bin/logscript,${CALLERID},pstn)
then I get error.
The singlke argument to
Instal subversion package, in your linux to be abale to use svn.
Regards
Carlos Rojas
On 12/20/05, Douglas Garstang [EMAIL PROTECTED] wrote:
No idea on this. Can't find it on digium's web site. How do we download the latest source for Asterisk? Looks like they switched to SVN from CVS? Never
When someone calls me via BroadVoice, they get a busy signal. My * box
is behind a NAT firewall. I have enabled port forwarding of UDP 5060
and 1:2 to the * box. I added nat=yes externalip and localnet
to the sip.conf under [general]. It still doesn't work. I just want *
to be
On Sat, 2005-12-24 at 20:17 -0500, Robert La Ferla wrote:
When someone calls me via BroadVoice, they get a busy signal. My * box
is behind a NAT firewall. I have enabled port forwarding of UDP 5060
and 1:2 to the * box. I added nat=yes externalip and localnet
to the sip.conf
trixter aka Bret McDanel wrote:
On Sat, 2005-12-24 at 20:17 -0500, Robert La Ferla wrote:
When someone calls me via BroadVoice, they get a busy signal. My * box
is behind a NAT firewall. I have enabled port forwarding of UDP 5060
and 1:2 to the * box. I added nat=yes externalip
Need help Install asterisk-oh323hi everybodyi have just installed asterisk 1.2.1 and added asterisk-oh323-0.7.3installed also pwlib1.5.2
and openh323_1.12.2 (the Mimas patches 2)i did followed all instructions but when i it created this problem.
hope you could help me.thanks so much in
trixter aka Bret McDanel wrote:
On Sat, 2005-12-24 at 20:17 -0500, Robert La Ferla wrote:
When someone calls me via BroadVoice, they get a busy signal. My * box
is behind a NAT firewall. I have enabled port forwarding of UDP 5060
and 1:2 to the * box. I added nat=yes externalip
Hi,
I am using SIPS softphoe. and i tested with another
SIP Gatekeeper and i can see callerid in plain format.
But when i am trying using Asterisk it is apearing
callerid, username.
So i don't think this is from client side or
softphone.
Yours,
Abdul Lateef
Computer Programmer
HATIF
hello,
You can check this compnay.
http://www.hatif.com
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
Tel: +974 - 4883068
ICQ: 276994704
YM!: abdul_zu
Fax: +974 - 4883063
Doha Qatar
http://www.hatif.com
On Fri, Dec 23, 2005 at 06:59:50AM -0600, Rich Adamson wrote:
This is another thing: Linux tends to use the availble free memory for
IO buffers, disk cache and such. So in the output of 'free', look at the
second line.
I'm not the OP, but for those of us that are not considered strong
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