For the record, I was just referring to the 0${EXTEN}portion.
The rest was the OP's reference.
But, I have needed to use the r option when interfacing to a PBX that didn't
supply dialtone, nor ringing tones.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope
Is there some way * can trim the trailing silence in a voicemail
message? There's the maxsilence setting for silence detection which
is related to what I'm asking but not the same. Let's say I set the
maxsilence to 8 seconds. During the recording of a voicemail, if
someone doesn't say
We have ran out of stock in our office in UK. All GSM Gateway are now being
send from HK therefore the shipping will be more expensive than usual.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bails
Sent: Friday, January 06, 2006 12:18 AM
To: Asterisk
I have some DID numbers that come into Asterisk via PRI, then connect to a
Panasonic DBS PBX via PRI.
Outbound calls from the PBX go out via PRI to Asterisk, then out to the Telco's
PRI.
The sync LEDs on my PBX show that it is synced to Asterisk via the PRI.
I have users complaining about
I have some DID numbers that come into Asterisk via PRI, then connect to a
Panasonic DBS PBX via PRI.
Outbound calls from the PBX go out via PRI to Asterisk, then out to the Telco's
PRI.
The sync LEDs on my PBX show that it is synced to Asterisk via the PRI.
I have users complaining about
Corey S. McFadden wrote:
PHP/MySQL based content manager for the Cisco 79XX series IP Phones
Any mailing list available for this project? I have some
questions/updates about this project...
___
--Bandwidth and Colocation provided by Easynews.com --
In article [EMAIL PROTECTED],
amaury BOSSE [EMAIL PROTECTED] wrote:
I have tried to use UserEvent() command to send data to Asterisk Manager from
my dialplan.
It works fine if the body only contains 1 line but I don't know how to send
multiple
arguments in the body.
I have never found
John covici wrote:
I did get the latest zaptel from cvs, but maybe this isn't up to date
-- sorry for the confusion.
There are multiple branches of Zaptel in CVS (and Subversion). This is
not specific enough for us to be able to help you.
How do y9ou determine the zaptel version for future
Kerry Garrison wrote:
The SC430 will experience unusable call quality with a TDM400P due to
IRQ Sharing problems. If you have some magic to get around this,
please share because everyone I know that has tried using an SC430
has given up and switched to other platforms. -Kerry
I experience
SICPE has a new product called the GSM Call Director that may be of interest
to GSM enthusiasts.
http://www.sipcpe.com/fx300GSM.html
Cory Andrews
Purchasing Manager
++
VOIPSupply.com
A Division of b2 Technologies
454 Sonwil Drive
Buffalo, NY 14225
direct -
Use channel of your agentChannel:
SIP/dov.bigioMaxRetries: 3RetryTime: 40WaitTime:
25Context: 01.telecomApplication: ChanSpyData: SIP/234-ssnfPriority: 1Cheers,Giovanni Miano2006/1/5, Dov Bigio
[EMAIL PROTECTED]:
Hi,
I have developped an application that monitors the
status of my queues
www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+outCheers,Giovanni Miano2006/1/5, Hill, John
[EMAIL PROTECTED]:
I was looking for a way to catch the zap busy return and do a redial.I would dial out on a zap channel. If the call is busy it would then hangupthe zap channel and ask if I
Is Exist InternalExtension context ? and 2093 exten ?2006/1/5, Aisling
[EMAIL PROTECTED]:
Hi all,
I am having difficulty getting incoming PSTN calls working.
I have set up an account with a third party provider. In my system, the user
register with SER and use Asterisk for
I gather from the information that this is not usable in the US market,.
It is NOT a world band GSM .
John Novack
Sam Tam wrote:
We have ran out of stock in our office in UK. All GSM Gateway are now being
send from HK therefore the shipping will be more expensive than usual.
-Original
$300 seems pretty expensive for such a device, especially since someone using it in conjunction with Asterisk would most likely not need its built-in routing features. It's a nice looking device though! Thanks,
RustyOn 1/5/06, Cory Andrews [EMAIL PROTECTED] wrote:
SICPE has a new product called
What is the price and availability?
robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Cory Andrews
Sent: Thursday, January 05, 2006 3:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GSM
I looked at this.
Iguess I willuse dialstatus busy and create a
.call file And see what happens.
Thanks
--john
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giovanni
MianoSent: Thursday, January 05, 2006 4:05 PMTo:
Asterisk Users Mailing List -
On Thu, Jan 05, 2006 at 10:19:23AM -0700, Douglas Garstang wrote:
I'd like to have Asterisk log useful messages during operation.
Is there any way in extensions.conf that I can manually log messages
to a file, say via syslog()? The console output is ugly, with all the
extra Executing
Scott DesBles wrote:
I am working on adding three older Cisco phones to *, two 12SPs and
one 30VIP. One of the 12SPs (griffin) and the 30VIP (scott) is
booting correctly and I have dial tone. The other 12sp starts up,
then I get a message on the display stating Requesting
Try chan_sccp
When I try to boot my 501, it runs through the usual stuff, then stops with
Config file error
Error is 0x4020
and then reboots.
The log on the FTP server shows:
0105164151|app1 |3|00|Bootline: ircaIP
0105164155|cfg |3|00|Image bootrom.ld has not changed.
0105164159|cfg |3|00|0004f202f803.cfg
The polycoms have a similar page, when set up just right. (from memory)
PaulH
- Original Message -
From: Aaron Daniel [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, January 06, 2006 4:24 AM
Subject: Re:
Problem resolved.
This makes it nice and simple to 'flash' an incoming POTS line (ZAP channel) as
opposed to the dialplan scripts that I have seen that require tranferring the
call, hanging up, and waiting for a call back. That was too confusing for my
wife. Now all she has to do is pres *3
Hi guys,
I've been installing and configuring a TE110p card. The compile and
install went very well. I'm using this on FC4 and I compile with
linux26 as well checked I on the udev configs.
zttool and ztcfg both indicate that the card is ready.
But when I try to load chan_zap.so then I get
I have been looking for analog phones for my * system that work with our plantronic amplifiers and headsets. The problem I am having with the Aastra phones that I have purchased (PT-390, 9116, 9120, 8009 ), is that they don't seem to stay hung up unless you physically hang up the handset
It's not working for me. If I make interdigit pause more than 1 sec, I
get hangup (busy) if number is not complete.
I don't know if it's possible, but I use a workaround to simulate the
external dialtone:
I use '0' to access external lines
exten - _0,1,ChanIsAvail(Zap/g1)
exten -
Hi everyone.
I'm trying to load test a FastAGI that sets CallerID data with
information from NANPA. Please feel free to connect to my test server
with Asterisk... you should get back the city/state of the caller.
I'd consider adding CallerID name support if I can find an inexpensive,
Hi,
I think you didn't install the libpri.
On Thu, 2006-01-05 at 22:48 +0100, Arnar Gestsson wrote:
Hi guys,
I've been installing and configuring a TE110p card. The compile and
install went very well. I'm using this on FC4 and I compile with
linux26 as well checked I on the udev configs.
Hi there,
well supposedly I've compiled and installed libpri and it actually
resides in /usr/lib. I've installed /usr/lib in the /etc/ld.so.conf
explicitly but without a luck. But I'll check this path anyhow more
thoroughly.
Thanks for the advice.
BR. Arnar
On Thu, 2006-01-05 at 23:01
Is the mac-address.cfg file name in lower case?
On Jan 5, 2006, at 1:37 PM, Ken D'Ambrosio wrote:
When I try to boot my 501, it runs through the usual stuff, then stops
with
Config file error
Error is 0x4020
and then reboots.
The log on the FTP server shows:
0105164151|app1
To be honest (and not answer your question
correctly):
We have used Polycom and Snom IP phones with
plantronics headsets, and they work very well.
regards,
PaulH
- Original Message -
From:
Richard Reina
To: asterisk-users@lists.digium.com
Sent: Friday, January
Hi,
did you compile the asterisk after the libpri install?
check it!
[EMAIL PROTECTED] ~]# ldd /usr/lib/asterisk/modules/chan_zap.so
libpri.so.1 = /usr/lib/libpri.so.1 (0x2ac3)
libtonezone.so.1 = /usr/lib/libtonezone.so.1
(0x2ad58000)
libc.so.6 =
I had this problem and while not 100% sure. I think the solution was
whitespace at the end of the lines in the conf file. Opened the file with
vi, went to each line and hit end to make sure there were no extra spaces,
there were, removed them, rebooted and issue went away.
Hi guys,
I've
On Thu, 2006-01-05 at 15:57 -0500, Cory Andrews wrote:
SICPE has a new product called the GSM Call Director that may be of interest
to GSM enthusiasts.
http://www.sipcpe.com/fx300GSM.html
Looks nice, doing triple band and so on.
I presume it works like an mobile-phone.
Does the counter
Good guess,
I recompiled asterisk and now things are falling in place. Thanks for
the pointers.
BR. Arnar
On Thu, 2006-01-05 at 23:12 +0100, Domjan Attila wrote:
Hi,
did you compile the asterisk after the libpri install?
check it!
[EMAIL PROTECTED] ~]# ldd
Anthony Rodgers wrote:
Is the mac-address.cfg file name in lower case?
Yeah, it is. Hell -- I've cut-and-pasted the filename from the below
logfile, and been able to FTP it just fine. I've run an ethereal dump,
and it never even -asks- the server for the file, so I'm kind of
confused there.
trying to make phpagi stream a .gsm file from sounds dir. Asterisk is
answering the call and runnng the script. However no file to be heard
over the phone. Has anyone ever used the Asterisk phpagi and was there
any configurations that i should be aware of to stream the files? Oh all
I suspect that there might be more to this question than has been answered
so far. Most firewalls will allow you to open and forward a port range;
thus they are SIP compliant.
However, if you want more than one SIP client behind your firewall, you will
want a firewall with a SIP application
I have seen issues where people had edited the files in windows.
The hint is that Asterisk was complaining about having ^M in the config
files
PaulH
- Original Message -
From: stotaro [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
OK, I will download the trunk from subversion and see if that works.
on Thursday 01/05/2006 Kevin P. Fleming([EMAIL PROTECTED]) wrote
John covici wrote:
I did get the latest zaptel from cvs, but maybe this isn't up to date
-- sorry for the confusion.
There are multiple branches of
I found solution:
Just set overlapdial=yes in zapata.conf, and then in extensions.conf
dial(zap/1/) or if you using groups dial(zap/g1/), and you will get
dialtone from local exchange (telekom).
Cheers.
It's not working for me. If I make interdigit pause more than 1 sec, I
get hangup (busy)
Can you make calls between SIP phones? It could be a phone config issue...
PaulH
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, January 06, 2006 1:50 AM
Subject: [Asterisk-Users] Bizarre Answering Behavior
Ok, I've been trying to figure
We've done a direct swap of an old Amanda voicemail system with a shiney
new Asterisk system (Asterisk 1.0.9). The system consists of 4 FXO
ports on the * box (TDM400P), and three old Wildcards we aren't using
(too buggy we found).
CO lines- Toshiba - FXO ports on *
We want to branch out
Technical Support wrote:
So if I can rephrase your question for the group, are there any (linux?)
firewalls with SIP RTP application filters?
Pretty much any recent one, just load the ip_conntrack_sip module:
http://www.iptel.org/sipalg/
Tony
___
If you're wanting to scroll through output from a CLI command, use:
asterisk -rx command | less
Tomislav Parcina wrote:
What is command when I wona to list something page by page in * CLI?
Something that works like |less or |more.
Have a nice day!
Philip,
Your problem with dialing an extension on the Toshiba and only getting a
second of music on hold has to deal with the fact that you are using an
analog trunk. Asterisk will always say that the analog channel has
answered as soon as it is done sending dtmf on the line. You could help
Hi all,
I am having problems detecting fax on a client site using the TE406P
(card with echo cancellation module) under 1.2 and was wondering if
anybody was having or has had similar problems?
Currently I can not get the system to detect fax for love nor money,
in zapata.conf the relevant
make[2]: *** [obj_linux_x86_r/simph323] Error 1
make[2]: Leaving directory `/usr/src/openh323/samples/simple'
make[1]: *** [opt] Error 2
make[1]: Leaving directory `/usr/src/openh323'
make: *** [optshared] Error 2
any idea ?
Thx in Advance
make[2]: *** [obj_linux_x86_r/simph323] Error 1
make[2]: Leaving directory `/usr/src/openh323/samples/simple'
make[1]: *** [opt] Error 2
make[1]: Leaving directory `/usr/src/openh323'
make: *** [optshared] Error 2
any idea ?
Thx in Advance
Dear All,
Can someone give me some sample script to limit outgoing call based on
their Ext to outside using pstn line ?
Let Say :
Ext 101, he just can make a call to interlocal with duration just 5
minute and from Mon-Fri at 8AM-18PM.
Ext 102-105, they just can make a call to local only
The PBX I'm getting ready to replace has a really nifty feature -- one
that I'm not even sure Asterisk -could- do -- though I'm hoping to be
proven wrong. When a call goes to voicemail, the end-user can listen to
the VM as it's being recorded, and can interrupt and answer the call if
it's someone
Cameron Grant wrote:
I am having problems detecting fax on a client site using the TE406P
(card with echo cancellation module) under 1.2 and was wondering if
anybody was having or has had similar problems?
Yes, we are aware that the VPM currently breaks FAX tone detection. For
a temporary
Mr. James W. Laferriere wrote:
Is the in developement functionality in the svn ?
No. It will be once I create a 'netsock2' tree in my development area,
but that likely won't happen for a week or two.
___
--Bandwidth and Colocation provided by
I have ISDN BRI line (Point-to-multipoint), with HFC card and
bristuff/zaphfc driver.
I also have TA attached to NT, with analog phones and modem on it.
When I'm dialling through Asterisk/bristuff, and in the same time TA
have some conversation (or maybe modem link) on channel 1, I can hear
that
Here is the issue:
[EMAIL PROTECTED] ~]# modprobe zaptel
[EMAIL PROTECTED] ~]# lsmod | grep zaptel
zaptel206724 0
crc_ccitt 2113 1 zaptel
[EMAIL PROTECTED] ~]#
[EMAIL PROTECTED] ~]#
[EMAIL PROTECTED] ~]# modprobe ztdummy
Notice: Configuration file is
The PBX I'm getting ready to replace has a really nifty feature -- one
that I'm not even sure Asterisk -can- do -- though I'm hoping to be proven
wrong. When a call goes to voicemail, the end-user can listen to the VM
as it's being recorded, and can interrupt and answer the call if it's
someone
Do bristuff/zaphfc support CD (Call Deflection)?
How to deflect call (transfer before answering) with bristuff?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
On Thu, 2006-01-05 at 22:29 -0500, Ken D'Ambrosio wrote:
The PBX I'm getting ready to replace has a really nifty feature -- one
that I'm not even sure Asterisk -can- do -- though I'm hoping to be proven
wrong. When a call goes to voicemail, the end-user can listen to the VM
as it's being
For your first problem, try using callprogres=yes in
zapata.conf. may or may not work.Its easier to integrate with Toshiba
Strata a TE110.CCF-Original
Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]On
Behalf Of PhilipEdelbrockSent: Friday, January 06, 2006 08:01To:
Olle E Johansson wrote:
Mikael Magnusson wrote:
Olle E Johansson wrote / skrev:
Andreas Koch wrote:
Hello,
how is it possible to connect (register) more the one Phone to One
Sip-Acoount.
With, for example sipgate.de this is not a special feature, it is
common.
We have users, what like
BJ Weschke wrote:
On 1/3/06, Kerry Garrison [EMAIL PROTECTED] wrote:
The magic setting is callprogress=yes, however, we have this working
properly in the lab but not at this particular client location right now.
Strange, but true.
-Kerry
You're going to have very unpredictable results with
Frank Liu wrote:
Within asterisk, is it possible to detect that an incoming call is a
direct dialing, or forwarded via another place? When a call is being
forwarded via a 3rd party (say, SBC), will it have some indication in
the call packet?
You mean something like as is documented in
Code Lover wrote:
Hi friends,
How i can enable and disable RBT in asterisk for SIP users.
We have linksys IP Phones but its give ring to the caller before
ringing the called phone.
Don't use the r option to Dial
___
--Bandwidth and Colocation
I've got an EM wink T1 into a Sangoma card, and it's dropping my DNIS
digits. I'm supposed to get four digits from the CO, and I reliably get
the first digit -- it's a crapshoot as to how many after the first I
get. I have overlapdial=yes and immediate=no.
Any other suggestions of things to
The dispatcher module in OpenSER can load balance calls based on a hash of the
SIP call-id. Supposedly the latest version even supports failover. O, fancy.
Doug.
-Original Message-
From: tijmen van den brink [mailto:[EMAIL PROTECTED]
Sent: Wed 1/4/2006
I've asked this question several times before and was always told it wasn't
possible. However, after reading a thread posted to the list today, I'm not so
sure my question was understood.
So, here I go again...
Is it possible to have multiple Asterisk systems share a common realtime
101 - 165 of 165 matches
Mail list logo
