Hello Faris,
Only I've sidetracked and am currently trying to use capi4hylafax instead of
iaxmodem which seems to work wonderfully but I'm having some issues with
root verses uucp permissions which is spoiling my fun.
Make sure not to run faxgetty together with capi4hylafax.
--
Best regards
I have been trying for some time now to make the hook flash work on the
FXS port.
I am using Asterisk 1.4.10.1 with zaptel 1.4.4.
When I manually flash the hook I can manage to find the duration to put
a call on hold. However when pushing the flash button it never works. The
phone's flashtime
Hi,
I am trying to configure for MFC/R2 for asterisk. With the help of one
of the asterisk users group
member patrick I am able to install libunicall library. Now, when trying to
install libmfr2-0.0.3 it is giving error.
On running running command $./configure
It is giving error -
On Thu, 16 Aug 2007, Bill Andersen wrote:
OK, I understand that. But if I gotta learn how to support
myself to do advanced features, why pay them at all? I'll
just become my own expert :()
That's how I started...
Sit-down and work out what features you want - and do you want them
On Fri, Aug 17, 2007 at 01:23:11PM +0530, [EMAIL PROTECTED] wrote:
Hi,
I am trying to configure for MFC/R2 for asterisk. With the help of one
of the asterisk users group
member patrick I am able to install libunicall library. Now, when trying to
install libmfr2-0.0.3 it is giving error.
100% repeatable (for me).
Sip phone A calls Sip phone B.
Either Sip phone A or B does #700. The party that keyed #700 gets the
parked announcement (eg 701) and the other party get MOH. There is
still an audio channel between the two SIP phones at this point.
When the party that typed #700
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all
I am working in a new set up with Grandstream GXP-2000 handsets. I
like those phone, but they lack a feature I need: the phone cannot be
locked by the user.
What I actually want is a user to be able to avoid someone else making
calls from his
Haven't tried with new 1.4 branch of Asterisk/Zaptel, but in 1.2, if
correctly configured, there was no need to Wait(x) to let zaptel to get
the CID on analog lines: it was zaptel itself to not let the call go through
the dialplan until the second ring. I think it shoud be something like:
Steven wrote:
I am curious.
Why would one need to do this?
If a phone connect to 5060 from another port number, asterisk happily works,
so why use multiple port on asterisk?
I cannot see the thread history but from the context, I would say
because many ISPs block 5060, 25, and others.
What i actually do is make asterisk listen on some other port like 5097 and
redirect port 5060 to it with iptables like this
/sbin/iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5060 -j DNAT --to
YOURIPHERE:5097
This works very well . If i make asterisk listen on 5060 and redirect say
5097
Summary:
Can't hear incoming/outgoing calls to/from ISDN over Sangoma A102 card.
Debian stable + Asterisk 1:1.2.13.
Thanks for the response, Andres. We've changed the timing source but
still no joy. This is very odd; calling between internal AIX/SIP
extensions works fine.
If anyone else can
On Thu, 2007-08-16 at 19:38 -0600, Steve Murphy wrote:
On Thu, 2007-08-16 at 07:56 -0400, Russell Bryant wrote:
Gordon Henderson wrote:
; *99:
; 99 bottles of beer on the wall.
exten = *99,1,Noop(99 Bottles of beer on the wall)
exten = *99,n,Answer()
exten =
On Fri, 17 Aug 2007, Andres Jimenez wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all
I am working in a new set up with Grandstream GXP-2000 handsets. I
like those phone, but they lack a feature I need: the phone cannot be
locked by the user.
What I actually want is a user to
Ahh, I see.
Good point.
--
--
Steven
http://www.glimasoutheast.org
Steve Totaro [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Steven wrote:
I am curious.
Why would one need to do this?
If a phone connect to 5060 from another port number, asterisk happily works,
so why
Hello,
I have a Sangoma A101 connected to an ISDN30 (E1 in the UK) with some
Snom 300 and Idefisk softphones.
I can do SIP and IAX2 calls just fine, however I cant get any audio in
either direction on the Zap channels. When I call in or dial out over
the ISDN30 (UK E1) I can see the call
Russell Brown wrote:
100% repeatable (for me).
Sip phone A calls Sip phone B.
Either Sip phone A or B does #700. The party that keyed #700 gets the
parked announcement (eg 701) and the other party get MOH. There is
still an audio channel between the two SIP phones at this point.
Gordon Henderson wrote:
On Fri, 17 Aug 2007, Andres Jimenez wrote:
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,VMAuthenticate(${me})
exten =
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
2007/8/17, Gordon Henderson :
S (all untested!)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten =
I am trying to get a GSM gateway (Alpha Tech GSM Gateway Blue Gate Dual
Band Analoog FXO) working with Asterisk.
I had a working FXO configuration to a analog port of a small home 1/4
ISDN pbx.
I used this same configuration to connect a GSM Gateway that is supposed to
be connected to the
I dialed it, but I am still thirsty. ;-)
On Thu, 2007-08-16 at 19:38 -0600, Steve Murphy wrote:
On Thu, 2007-08-16 at 07:56 -0400, Russell Bryant wrote:
Gordon Henderson wrote:
; *99:
; 99 bottles of beer on the wall.
exten = *99,1,Noop(99 Bottles of beer on the wall)
On Fri, 17 Aug 2007, Doug Lytle wrote:
Gordon Henderson wrote:
On Fri, 17 Aug 2007, Andres Jimenez wrote:
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten =
Doug wrote:
I've looked at the following pages, and they are
just so garbled. I keep going around in circles:
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
Hi, folks.
I have a problem using Asterisk 1.2. I create conferences using
app_meetme and Zap channels, and for every participant I run the script
defined by AGI_BACKGROUND_SCRIPT to be able to listen and react to DTMF
tones. As the docs tell me, when using the AGI background script one
loses the
Gordon Henderson wrote:
On Fri, 17 Aug 2007, Doug Lytle wrote:
XOR in dialplan :)
9 lines of code vs. my 7 (which include a validation) though ;-)
Ooops! Missed that line.
One thing I noted recently is that phones sometimes do weird things with
*'s and #'s )-: The Siemens C460IP
Hi all,
Some of my asterisk users have used their maximum call limit for incoming
calls (peers). There incoming call limit should automatically reset to zero
after hangup but its not happening and they no longer can recieve any calls
as their allowed limit is already full. So is there any way to
On Fri, 17 Aug 2007, Hans Feringa wrote:
I am trying to get a GSM gateway (Alpha Tech GSM Gateway Blue Gate Dual
Band Analoog FXO) working with Asterisk.
I had a working FXO configuration to a analog port of a small home 1/4
ISDN pbx.
I used this same configuration to connect a GSM Gateway
In article [EMAIL PROTECTED],
Rory Campbell-Lange [EMAIL PROTECTED] wrote:
Summary:
Can't hear incoming/outgoing calls to/from ISDN over Sangoma A102 card.
Debian stable + Asterisk 1:1.2.13.
Thanks for the response, Andres. We've changed the timing source but
still no joy. This is very odd;
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Thanks to Gordon Doug I have now a very good locking system using
one only extension.
Extension informs you about the current lock situation and, if
authentacated, it changes it and explain the change done.
;Locking system
;LOCK
exten =
Get the providers support on the line and I will bet anything you are
not hitting their side of the line, this is most likely a signaling or
channel numbering issue.
Anthony
Veselin Kantsev wrote:
Hello,
I have a Sangoma A101 connected to an ISDN30 (E1 in the UK) with some
Snom 300 and
I think you did not installed properly the libraries. libmfcr2 check for:
checking tiffio.h usability... yes
checking tiffio.h presence... yes
checking for tiffio.h... yes
checking for TIFFOpen in -ltiff... yes
That is, you need to have BOTH, headers and shared library.
ldd in protocol_mfcr2.so
Have you looked in show channels/core show channels to see if they have any
dead/zombie channels, which you can remove with soft-hangup?
What version of * are you running?
What kind of phones?
What config options are you using in SIP (or other tech) to limit the calls?
On 8/17/07, Rizwan
Thanks for your response. My answers below.
On Fri, 17 Aug 2007, Hans Feringa wrote:
I am trying to get a GSM gateway (Alpha Tech GSM Gateway Blue Gate Dual
Band Analoog FXO) working with Asterisk.
I had a working FXO configuration to a analog port of a small home 1/4
ISDN pbx.
I used this
Hello fellows!!!
I'm having problems with Ericsson Dialog 3221 phone and DISA. I've
configured an extension to test DISA and it work properly with all other
phones, but freeze with the mentioned phone.
Here is my extension:
exten = 105,1,Answer
exten =
Hello,
I have the Jain-Sip-Applet-Phone installed on two machines in a small LAN
network. These machines are connected through an Asterisk Server (Using
Trixbox). I run the phone as an application on both machines through Eclipse
and I am able to log on as a user with one of the extensions
Thanx for ur reply.
Im running * 1.4.2. i dont think there is any problem in asterisk because
only one user is having this problem.
User is using Aastra 480i Cordless phone
Here is the sip config for that user. Im using call-limit=2 for every user
[saadfarr]
username=saadfarr
type=friend
I'm hoping people can suggest some ideas for debugging a problem that I'm
having with DTMF.
Unlike most of the DTMF problems reported here, it has nothing to do with
Asterisk interpreting DTMF. My problem is with the synthesis of DTMF tones
on outbound calls on a PRI connected to a TE412P card.
If I recall 1.4.2 has a deadlock problem on their SIP channels -- you should
upgrade to at least 1.4.5, which is when this was resolved. The problem was
present in 1.2 and 1.4 -- 1.4.5 earliest that I believe does not have this
issue.
Anthony
On 8/17/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
On Thu, Aug 16, 2007 at 11:00:05PM -0400, Zeeshan Zakaria wrote:
This is really ridiculous. So this means that now nobody can use
fax-to-email without paying to J2 first?
Horseshit. Prior art is trivial. How old is Hylafax?
Cheers,
-- jra
--
Jay R. Ashworth Baylink
Thanks, I'll take a look at it and see what new tricks I can learn. I did
use the astdb initally in my first version but at the time didn't see a good
way to add the state field that I wanted in there. At the moment I have a
number of external programs which use the postgres database - is it
Interesting. I have essentially the same settings but if I don't wait
a brief period then I don't get the callerid filled in. This was with
1.2 or 1.4
Haven't tried with new 1.4 branch of Asterisk/Zaptel, but in 1.2, if
correctly configured, there was no need to Wait(x) to let zaptel to get
I have the following dialplan.
Everything seems good except for one thing.
If the background message is playing and the user hangs up and does not
press a digit
how do I run an agi on that event.
I tried an exten = h,1,agi(smvoice,-digium_failed) but that was never
called.
I am using 1.4.10
On Fri, 17 Aug 2007, Hans Feringa wrote:
Thanks for your response. My answers below.
exten = _87.,n,Dial(Zap/1/${EXTEN:2},30,rtT)
My only other suggestion would be to remove the timeout...
Other than that there's no major difference between my setup and yours.
For outbound dialling, I use:
Hi,
I am using Asterisk 1.2.19 and have noticed a strange behaviour in my
system.
Sometimes (I could not reproduce it yet), when there is a call in place
between one extension and a PSTN number, the MOH suddenly starts to play
for one of them, while the other can still hear what is being said
On Fri, Aug 17, 2007 at 12:51:35PM -0400, Matthew Rubenstein wrote:
eBay is that marketplace, owns Skype, that telco, owns PayPal, that
bank. This outage should be screaming from the headlines. As those three
essential services become essential to more people around the world,
they need
ok this is a wired problem. when i use X-Lite - after i register with
asterisk X-lite sends a subscribe/notify request to asterisk to
determine if the account has any messages waiting.
if i create a sip.conf account using:
user 12345 with a voicemail box 12345 - MWI works
user jwolosuk with a
On Aug 17, 2007, at 5:55 AM, Rory Campbell-Lange wrote:
Summary:
Can't hear incoming/outgoing calls to/from ISDN over Sangoma A102
card.
Debian stable + Asterisk 1:1.2.13.
Thanks for the response, Andres. We've changed the timing source but
still no joy. This is very odd; calling
On Aug 17, 2007, at 6:52 AM, Doug Lytle wrote:
Gordon Henderson wrote:
On Fri, 17 Aug 2007, Andres Jimenez wrote:
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten = ,n,Set(DB(${me}/locked)=1)
exten = ,1,Answer()
exten = ,n,Set(me=${CALLERID(num)})
exten
At 06:37 AM 8/17/2007, you wrote:
Some of my asterisk users have used their maximum call limit for
incoming calls (peers). There incoming call limit should
automatically reset to zero after hangup but its not happening and
they no longer can recieve any calls as their allowed limit is
already
I think its an Asterisk bug, call-limits stopped working for me once I
upgraded from 1.2.16 to 1.2.18. There is a bug opened for it, but the
issue hasn't been resolved yet. Here is the link:
http://bugs.digium.com/view.php?id=9794
-Remi
Ira wrote:
At 06:37 AM 8/17/2007, you wrote:
Jay R. Ashworth wrote:
On Thu, Aug 16, 2007 at 11:00:05PM -0400, Zeeshan Zakaria wrote:
This is really ridiculous. So this means that now nobody can use
fax-to-email without paying to J2 first?
Horseshit. Prior art is trivial. How old is Hylafax?
Cheers,
-- jra
It's
It's a bug and you should probably upgrade, but yes, restarting will resolve
the problem temporarily.
On 8/17/07, Ira [EMAIL PROTECTED] wrote:
At 06:37 AM 8/17/2007, you wrote:
Some of my asterisk users have used their maximum call limit for
incoming calls (peers). There incoming call limit
Horseshit. Prior art is trivial. How old is Hylafax?
Cheers,
-- jra
It's never trivial if you're a small company. J2 has already won
settlements from several smaller companies, which gives it
precedence.
Once precedence is established, it's almost a done deal for future
At 08:19 8/17/2007, Dave Fullerton wrote:
Doug wrote:
I've looked at the following pages, and they are
just so garbled. I keep going around in circles:
http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
I have the sample problem. Just turned it off for now.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Remi Quezada
Sent: Friday, August 17, 2007 2:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call
Hi
iam using Asteriks 1.2.17
Server Side ( provider Side g729)
clients side gsm
when iam calling, iam getting lot of errors like below
and lot of voice breaks
Aug 16 21:23:14 WARNING[9521] dsp.c: Inband DTMF is not supported on codec
gsm. Use RFC2833
any suggestions
ram
On Fri, 17 Aug 2007, Andres Paglayan wrote:
Guys, very nice dialplan programming,
as a user's opinion, the two extension approach might be better.
so the user doesn't need to remember whether the phone is locked or not,
and accidentally lock it when the contrary was meant,
(unless you send
2007/8/17, Andres Paglayan [EMAIL PROTECTED]:
Guys, very nice dialplan programming,
as a user's opinion, the two extension approach might be better.
so the user doesn't need to remember whether the phone is locked or not,
and accidentally lock it when the contrary was meant,
(unless you send
Andres Jimenez wrote:
In the latest version (see below) I added some playback that will say
if the phone is lock or unlock, before and after locking/unlocking it.
Just a note,
You will want to make sure that (911/999) calls are handled properly
when the phone is locked down.
Doug
--
On Sat, 18 Aug 2007, ram wrote:
Hi
iam using Asteriks 1.2.17
Server Side ( provider Side g729)
clients side gsm
when iam calling, iam getting lot of errors like below
and lot of voice breaks
Aug 16 21:23:14 WARNING[9521] dsp.c: Inband DTMF is not supported on codec
gsm. Use RFC2833
On 8/18/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Sat, 18 Aug 2007, ram wrote:
Hi
iam using Asteriks 1.2.17
Server Side ( provider Side g729)
clients side gsm
when iam calling, iam getting lot of errors like below
and lot of voice breaks
Aug 16 21:23:14
Hello,
I have a Sangoma A101 connected to an ISDN30 (E1 in the UK) with some
Snom 300 and Idefisk softphones.
I can do SIP and IAX2 calls just fine, however I cant get any audio in
either direction on the Zap channels. When I call in or dial out over
the ISDN30 (UK E1) it rings the other end but
On 8/17/07, David Roden [EMAIL PROTECTED] wrote:
Hi, folks.
I have a problem using Asterisk 1.2. I create conferences using
app_meetme and Zap channels, and for every participant I run the script
defined by AGI_BACKGROUND_SCRIPT to be able to listen and react to DTMF
tones. As the docs tell
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
2007/8/17, Doug Lytle :
Just a note,
You will want to make sure that (911/999) calls are handled properly
when the phone is locked down.
Good point.
- --
Andres Jimenez
GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]
-BEGIN PGP
Hi all and thanks for every suggest about my problem, I found that my TDM400P
was sharing IRQ with onboard sound device using cat /proc/interrupts, lspci -v
and lspci -vb. When I disable all unnecessary hardware on my machine and test
it, clicking sounds continue on the line with the same
On Fri, 2007-08-17 at 18:22 +0200, Trixter aka Bret McDanel wrote:
On 8/17/07, Aleks Clark [EMAIL PROTECTED] wrote:
Actually, the crazy p2p connections actually reinforce their algorithm
story. If their p2p algorithms have flaked out, it could cause all sorts of
trouble. OTOH, I don't think
Questions:
1. Is the wiki DUNDi example and the dundi.conf file too difficult to
follow for new users?
2. Does the complexity of the DUNDi setup discourage you from using it
or even attempting to configure it?
3. If there was a simple tutorial, step by step guide with easy to
setup and test
1. Yes
2. Yes
3. Yes
Nice sales pitch, sounds like one of those late night get rich now! schemes.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of JR Richardson
Sent: Friday, August 17, 2007 4:35 PM
To: asterisk-users@lists.digium.com
Subject:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of JR Richardson
Sent: Friday, August 17, 2007 4:35 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk
Users,Please Give Feedback
Upgrade to what? I'm running the most recent 1.2 version and I can't
run 1.4 because every 3 calls I have to reboot the machine.
Ira
At 11:00 AM 8/17/2007, you wrote:
It's a bug and you should probably upgrade, but yes, restarting will
resolve the problem temporarily.
On 8/17/07, Ira
JR Richardson wrote:
Questions:
1. Is the wiki DUNDi example and the dundi.conf file too difficult to
follow for new users?
2. Does the complexity of the DUNDi setup discourage you from using it
or even attempting to configure it?
3. If there was a simple tutorial, step by step guide with
What exactly is patented by J2? Is it receiving fax over the Internet,
converting to PDF and sending as an email attachment using sendmail or
postfix etc? Or is it receiving it through PRI, or PSTN line over the
computer and converting and emailing? What exactly they don't want others to
do?
The Asterisk.org development team has announced the release of Zaptel
versions 1.2.20 and 1.4.5. These releases are maintenance releases that
fix various known issues. See the ChangeLog included in the releases for
a full list of changes. The ChangeLogs are also available separately on
the ftp
Bill Andersen wrote:
I'm a network admin that maintains 3 commercial Asterisk
servers for my employer.
I am wanting to move away from the pre-packaged commercial PBXs
to a more pure asterisk setup. The systems I have utilize a nice
web GUI to make changes, but it really limits what I can
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I have what sounds like a simple requirement to add a feature to
an existing dial plan context. Currently the user gets a normal .wav
file played to them via MusicOnHold while they go through a basic menu
prompt. However I now need to take
At 19:35 8/17/2007, Lee Jenkins wrote:
Bill Andersen wrote:
I'm a network admin that maintains 3 commercial Asterisk
servers for my employer.
I am wanting to move away from the pre-packaged commercial PBXs
to a more pure asterisk setup. The systems I have utilize a nice
web GUI to
On Friday 17 August 2007 04:34:33 pm JR Richardson wrote:
Questions:
1. Is the wiki DUNDi example and the dundi.conf file too difficult to
follow for new users?
it's a bit complicated, though it seems to make sense for large-scale ops.
2. Does the complexity of the DUNDi setup discourage
Hi,
With more recent version of v1.2 and with v1.4 are things like the
AstManProxy still recommended if you want to have a bunch of
applications talking directly to Asterisk?
Cheers!
--
Andrew Ruthven, Wellington, New Zealand
At work: [EMAIL PROTECTED]
At home: [EMAIL PROTECTED]
GPG fpr: 34CA
On Fri, Aug 17, 2007 at 06:15:28PM -0500, Asterisk Development Team wrote:
The Asterisk.org development team has announced the release of Zaptel
versions 1.2.20 and 1.4.5. These releases are maintenance releases that
fix various known issues. See the ChangeLog included in the releases for
a
On Fri, Aug 17, 2007 at 04:34:33PM -0500, JR Richardson wrote:
Questions:
1. Is the wiki DUNDi example and the dundi.conf file too difficult to
follow for new users?
You imply that both need fixing
2. Does the complexity of the DUNDi setup discourage you from using it
or even attempting
Zeeshan Zakaria wrote:
What exactly is patented by J2? Is it receiving fax over the Internet,
converting to PDF and sending as an email attachment using sendmail or
postfix etc? Or is it receiving it through PRI, or PSTN line over the
computer and converting and emailing? What exactly they
Use a MeetMe room
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Vincent Sweeney
Sent: Friday, August 17, 2007 9:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk Channel as
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