Hi,
I saw somewhere that it was possible to set a limit for how long time a call
could be, for an extension in extension.conf. But I can't find it anymore.
Can someone please help.
Calls to '411' an operator may max. be 5 min.
I have this in extension.conf.
[shortcuts]
exten =
I am pitching an Asterisk solution to a local company when a few
interesting questions came up.
What is the max # of callers that can participate in a conference call? A
meetme style call, if it makes any difference. I have googled and looked
around and cannot find any limit.
Also, is there any
Did you compile the zap and lipri and installed ?
/HHA
app_dial.c:533 dial_exec: Unable to create channel of type
'ZAP'
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To UNSUBSCRIBE or
Hi,
I have 3 friends trying to connect to my Asterisk using x-lite, all of them
are using 3 dif. adsl-provider.
For each of them I got this in sip.conf:
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=g729
allow=g723.1
[seholm]
type=friend
secret=**
auth=md5
nat=yes
host=dynamic
I have no problem transfer from one GS adaptor to another GS adaptor.
/Hans-Henrik Andresen
Can anyone confirm that this problem exists?
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To
For what its worth, I just got started with NuFone.net ... and they
had me up and running in less than 3 hours (including installing
Asterisk on my box for me) with 2 Michigan DID numbers and
one toll free number.
So far, so good.
I'd appreciate (email me directly) any feedback on them, so
I know
exten = 1,AbsoluteTimeout ($SECONDS)
Ta
SJ
Hi,
I saw somewhere that it was possible to set a limit for how long time
a call could be, for an extension in extension.conf. But I can't find
it anymore.
Can someone please help.
Calls to '411' an operator may max. be 5 min.
I have
Hans-Henrik Andresen wrote:
Hi,
I have 3 friends trying to connect to my Asterisk using x-lite, all
of them are using 3 dif. adsl-provider.
For each of them I got this in sip.conf:
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=g729
allow=g723.1
[seholm]
type=friend
Senad Jordanovic wrote:
Hans-Henrik Andresen wrote:
Hi,
I have 3 friends trying to connect to my Asterisk using x-lite, all
of them are using 3 dif. adsl-provider.
For each of them I got this in sip.conf:
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=g729
allow=g723.1
[seholm]
Hi
The error message is generated by PHP... If your PHP version is higher
that v4.3.0
line 5: $stdout = fopen('php://stdout', 'w');
should be
$stdout = fopen(STDOUT, 'w');
Wrong... STDOUT is already open within php cli (4.3.0 and above)
so just do
fwrite(STDOUT,blah);
and you're
Hi
I tried to raise it to 5000, but still unreachable.
But as I wrote earlyer, for the same config, sjphone and a Grandstream 286
works.
/HHA
qualify=1000
If the client turns UNREACHABLE, you might want to change the qualify=
setting to qualify=yes,
that defaults to two seconds, instead of
I use telappliant's voiptalk service. I subscribed to their voiptalk
service, purchased ?10 of iaxtalk credit and was up and running in under 10
minutes! Very impressed with the quality of their termination so far. Their
support is also exceptional. Thinking of trying their telcentrex service
I saw in the current CVS version that a new parameter has been added to
app_dial.c...
The option string may contain zero or more of the following characters:\n
't' -- allow the called user transfer the calling user\n
'T' -- to allow the calling user to transfer the call.\n
'r'
Hi Jens,
MFE for TEI=76 means that a layer 2 p2p connection has been
established between your * and your telco's switch. This is a
good thing! :)
It looks like your telco is pulling down layer 2 when the line
is idle (probably also layer 1 for power saving). Do you see
a lot of card X span Y
Hi Rich
You never did tell us what the problem is that you're trying to solve,
or what you've done to help identify whatever the problem happens to be.
Thanks for responding.
What I am doing is I have 2 x Mediatrix 1104 boxes which I will be plugging
analog phones into.
I found the initial
Hello ,
i´ve been trying to connect with x-lite to our new installed asterisk
box ( version CVS-03/02/04-20:21:53 ) as i can see with etheral does
x-lite send sucessfully the REGISTER command to the server. On the
server i get during the same time this error message:
Mar 7 10:45:06
Been using VoicePulse connect service for bout 2
months now.
Got outbound calling working via IAX2 very quickly.
Call Quality-Great.
Flexibility with Connect Service - Great
Ordered DID. Was not working initially but they fixed
it. Total order time, about 6 Days.
Reliability - Lets just say its
[EMAIL PROTECTED] wrote:
I have no problem transfer from one GS adaptor to another GS
adaptor.
/Hans-Henrik Andresen
Can anyone confirm that this problem exists?
The problem I'm experiencing with many GS adapters, regardless of
firmware version is this. Call from one phone to another
I have made no recent changes to the IAX2 config on my system. Today I
tried a 1800 call and got the below error. Not sure when this started
since only use 800 once in a while. Does anyone know if IAXTEL is
experiencing problems connecting to the 8xx gateway?
7 16:14:54 WARNING[147466]:
Hi,
Thank you, but this I cant get to work.
/HHA
so that should enable you to do the following:
Call timeout = 20 sec
Max Call Duration = 300 sec = 5 min.
exten =
411,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],20,S(300))
however, I have not tried it yet so someone
Thank you This works, but. It just cut the line, I had hoped for some
bip bip bip to remind that now your about to be disconected, is this
possible as well ?
/Hans-Henrik
Senad Jordanovic [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
exten = 1,AbsoluteTimeout ($SECONDS)
Hi there,
Klaus-Peter Junghanns wrote:
It looks like your telco is pulling down layer 2 when the line
is idle (probably also layer 1 for power saving). Do you see
a lot of card X span Y state FZ (A_ST_RD_STA = 0x1Z) messages
in dmesg? Those indicate state changes on layer 1.
Negative. I do
You must change the setwhentohangup function, see channel.c for that.
Someone wrote a patch to do this (see http://bugs.digium.com/).
Regards,
Gus
- Original Message -
From: Hans-Henrik Andresen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, March 07, 2004 12:31 PM
Subject:
Yes, it is. (If I remember correctly :)
It is T that you need to include in that context.
[$CONTEXT]
exten = 1,AbsoluteTimeout($SECONDS)
exten = 2,Dial($SOMETHING)
exten = T,Playback($YOURMESSAGE)
Save $YOURMASSAGE in /var/lib/asterisk/sounds
If above does not work, please let me know.
Ta
SJ
Hello All,
Does anyone know if it possible to crossconnect PSTN and VoIP system in
India?
I am getting input from local people in India that is not possible due
Laws and found on Internet what is allowed since April 2002.
--
Regards,
Vasyl
smime.p7s
Description: S/MIME Cryptographic
Hello All,
Can anyone give some ideas what could be the problems?
-- B-channel 14 successfully restarted on span 1
Mar 7 11:27:15 WARNING[65541]: chan_zap.c:5978 zt_pri_error: PRI: Read
on 36 failed: Unknown error 500
Mar 7 11:27:15 NOTICE[65541]: chan_zap.c:6693 pri_dchannel: PRI got
Hello all
I have recently stumbled accross voip and asterisk.
We have a small network of vpns running in the uk. I have managed to get
the sip phones dialing each other through asterisk and it is working
great. (we are having long free conversations and that is something to
get excited
Hi All
However now I get a new problem :-)
Just to let you know it was my incompetance that caused the problem and I now
have it all working as expected :-)
Regards
Mark
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[EMAIL PROTECTED]
Simon,
Caller ID does not work in the UK, well not on my BT or Telewest line's.
Have a look at my sample configs http://www.codepipe.com/id25.htm , I am
also in the UK and these work for me.
Give me a call if ya want to chat about it.
Regards
Dave
-Original Message-
From: [EMAIL
Ok, it actually works fine here..
Asterisk CVS-03/06/04-14:35:21, Copyright (C) 1999-2004 Digium.
From extensions.conf:
[pstn-out-nat]
;
ignorepat = 0
; NOT USED
exten = _0XX0X,1,Congestion
; Local eight-digit dialing accessed through trunk interface
exten =
Hello,
From: Vasyl Rublyov [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Crossconnect VoIP and PSTN in India. Is it
allowed?
Date: Sun, 07 Mar 2004 11:22:53 -0500
Hello All,
Does anyone know if it possible to crossconnect PSTN and VoIP system in
HMM - This wont work :(
exten = 10,1,Dial(SIP/hha1,20,S(10))
exten = 10,2,VoiceMail,u10
exten = 10,102,VoiceMail,b10
Soren Rathje [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Ok, it actually works fine here..
Asterisk CVS-03/06/04-14:35:21, Copyright (C) 1999-2004 Digium.
On Sun, Mar 07, 2004 at 07:04:55PM +0100, Hans-Henrik Andresen wrote:
HMM - This wont work :(
exten = 10,1,Dial(SIP/hha1,20,S(10))
exten = 10,2,VoiceMail,u10
exten = 10,102,VoiceMail,b10
Maybe itneeds SIP/hha1|20,S(10) ?
___
Asterisk-Users
Sorry, but that IS NOT implemented into asterisk... (as far I know of).
Hi,
This isn't quit good :( The caller have the message played, but the
called person are cut off without any warning..
I hoped to be warned, like In 1 minnute the line will be
disconected, or just som beep beep
Yeah ..
Here's an update. It turns out I need to have a zaptel.conf
file !?? Ok, so I found that and I now have the followign
setup:
-
cat zaptel.conf
loadzone=us
defaultzone=us
# Load FXS device (a TDM400P) as Channel #, and use
kewlstart FXO signalling
fxoks=1
# Load FXO
CW_ASN wrote:
People:
1) Some guy wrote a app_dial modification to start to count time when answer
arrives. Interested? (thanks to Luciano!)
AbsoluteTimeOut counts time since this statement is executed... If you have
a long ring time (without answer), it is counted!
With the proper patch, the
Thanks for your help David
Your configs are a little to complicated for this complete asterisk
newbie though.
All i am actually after is how to get a sip phone to ring when the X100P
is dialed on out landline, and how to get a sipphone to dial out through
the X100P.
I have saved all your
This is wrongs. It's me who wrote the patch, it's available in CVS
Are you Klaus? If you're not Klaus, you wrote another patch. If you're
Klaus, as you see, works in that way.
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I wonder if those who have the nerve to piddle with Grandstream's
somewhat chaotic firmware release methodology could give me any idea as
to which version of the firmware I want to be running for a maximally
functional experience?
I'm still running 1.0.3.81 because I read that once you move up
Upgrade to the latest CVS and ast_rtp_read/write warnings will
disappear. GS .50 is buggy. Voice is very thin. Sipura G.726 is works
great.
Master
Greg Boehnlein wrote:
Hello all,
I'm trying to get the g726 codec patch contained in:
http://bugs.digium.com/bug_view_page.php?bug_id=0001104 to
On Sunday 07 March 2004 20:08, Simon Chappell wrote:
Thanks for your help David
Your configs are a little to complicated for this complete asterisk
newbie though.
All i am actually after is how to get a sip phone to ring when the X100P
is dialed on out landline, and how to get a sipphone to
On Sunday 07 March 2004 20:08, Simon Chappell wrote:
Thanks for your help David
Your configs are a little to complicated for this complete asterisk
newbie though.
All i am actually after is how to get a sip phone to ring when the X100P
is dialed on out landline, and how to get a sipphone to
Hello,
But what about if we have Lucent Definity already in place and I would
like to connect to Definity using PRI Asterisk box and route all USA
and Ukrainian extensions thru VoIP? The same for people in US/UA - they
will dial Indian extensions and go thru, does it possible/permitted?
The
That is really appreciated :-)
I look forward to receiving them and getting straight to work..
Many thanks
Simon
Jon Lawrence wrote:
On Sunday 07 March 2004 20:08, Simon Chappell wrote:
Thanks for your help David
Your configs are a little to complicated for this complete asterisk
newbie
CW_ASN wrote:
This is wrongs. It's me who wrote the patch, it's available in CVS
Are you Klaus? If you're not Klaus, you wrote another patch. If you're
Klaus, as you see, works in that way.
Nopez i'm not
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Asterisk-Users mailing list
Simon,
Try the following configs:
/etc/asterisk/zaptel.conf
fxsks=1
loadzone=uk
defaultzone=uk
/etc/asterisk/zapata.conf
languages=en
context=inbound-analog
signalling=fxs_ks
; I always create dial groups for making outbound calls, you can use the
specific channels as well
group = 1
Chris Clifton wrote:
I would buy this, but my 7960 is using g729a on both lines that I'm dialing
out on (to conference), my * installation is licensed for 3 g729 channels.
What codecs are you using ? Is there a conference config in the 7960 that
I'm missing ?
I can make inbound and outbound
thanks so much..
I have dialed from my mobile and nearly fell off my chair when the Sip
phoone rang ,,!! then was sad enough to answer it and have a chat with
myself!!
Is there any provision for dialing out in these configs ? and if so is
it dial 9 ?
Thanks again as this has been a four day
On Sunday 07 March 2004 21:28, Simon Chappell wrote:
thanks so much..
I have dialed from my mobile and nearly fell off my chair when the Sip
phoone rang ,,!! then was sad enough to answer it and have a chat with
myself!!
Is there any provision for dialing out in these configs ? and if so is
Thanks again..
I was beginning to think I had a faulty card now i see it was just me.. :-)
I am going to make a backup of my config and spend some time reading
about all the extra nice features that i can implement now the basic
stuff is working..
##Thought##
The last company i worked for I
On Mon, 8 Mar 2004, Master Abi wrote:
Upgrade to the latest CVS and ast_rtp_read/write warnings will
disappear. GS .50 is buggy. Voice is very thin. Sipura G.726 is works
great.
Hmm.. when was this fixed? I'm running a CVS version that was pulled and
built this morning, however I believe
Hi!
I'm still running 1.0.3.81 because I read that once you move up to
1.0.4.x you can't go back again, and my experience isn't *that* crappy.
You probably want to start with 1.0.4.26 although also 1.0.4.17 seems to
have been relatively stable. Then there is also 1.0.4.39 but it seems to
be
Vijay,
What is the rate of NuFone.net, not listed on their
web though.
I know voicepulse is $7.99 / month .
--John J.Wang
Vijay Vaidyanathan wrote:
For what its worth, I just got started with
NuFone.net ... and they
had me up and running in less than 3 hours
(including installing
Asterisk
Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
HMM - This wont work :(
exten = 10,1,Dial(SIP/hha1,20,S(10))
exten = 10,2,VoiceMail,u10
exten = 10,102,VoiceMail,b10
When did you checkout your version of Asterisk from CVS ??
This feature was put into CVS on
Soren Rathje wrote:
Hans-Henrik Andresen [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
HMM - This wont work :(
exten = 10,1,Dial(SIP/hha1,20,S(10))
exten = 10,2,VoiceMail,u10
exten = 10,102,VoiceMail,b10
When did you checkout your version of Asterisk from CVS ??
This feature
On Sun, 7 Mar 2004, Greg Boehnlein wrote:
On Mon, 8 Mar 2004, Master Abi wrote:
Upgrade to the latest CVS and ast_rtp_read/write warnings will
disappear. GS .50 is buggy. Voice is very thin. Sipura G.726 is works
great.
Hmm.. when was this fixed? I'm running a CVS version that was
I am not running the V1-0stable. Use the development version. My version
is 2 days old. G726 added to development CVS about 10 days ago.
Greg Boehnlein wrote:
On Mon, 8 Mar 2004, Master Abi wrote:
Upgrade to the latest CVS and ast_rtp_read/write warnings will
disappear. GS .50 is buggy.
we use .50 but yet YMMV
On Sunday 07 of March 2004 23:10, Philipp von Klitzing wrote:
Hi!
I'm still running 1.0.3.81 because I read that once you move up to
1.0.4.x you can't go back again, and my experience isn't *that* crappy.
You probably want to start with 1.0.4.26 although also
Are you Klaus? If you're not Klaus, you wrote another patch. If you're
Klaus, as you see, works in that way.
Nopez i'm not
In that case, exists another patch from a guy called Klaus. I'm using this
patch since Dec2003.
Maybe helps, I don't know, but this is other alternative.
Its merged
Caller ID does not work in the UK, well not on my BT or Telewest line's.
What I didn't understand yet about * + X100P with caller id not working
in some countries is, it's a hardware or software limitation?
Isamar
CW_ASN wrote:
Are you Klaus? If you're not Klaus, you wrote another patch. If you're
Klaus, as you see, works in that way.
Nopez i'm not
In that case, exists another patch from a guy called Klaus. I'm using this
patch since Dec2003.
Maybe helps, I don't know, but this is other
Never could find that NAT-capable TFTP server...
But I run my own standard TFTP server... I used to run the free TFTP
server from solarwinds.net but now I run tftpd on my asterisk box and it
works fine.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hello all,
I'm trying to get the g726 codec patch contained in:
http://bugs.digium.com/bug_view_page.php?bug_id=0001104 to work with the
latest GrandStream beta firmware and I am a lot closer than I was a couple
of weeks ago with the 1.0.4.46 firmware. I am now hearing Audio that is
Hi...I'm trying to figure out the famous 3 tests that a STUN client uses for
determining the kind of NAT that it is behind.
Is there a command line client available to send binding requests to a known
STUN server?
I'm aware of the SourceForge ones.
Either Linux or Windows
On Sun, 7 Mar 2004, James Golovich wrote:
On Sun, 7 Mar 2004, Greg Boehnlein wrote:
On Mon, 8 Mar 2004, Master Abi wrote:
Upgrade to the latest CVS and ast_rtp_read/write warnings will
disappear. GS .50 is buggy. Voice is very thin. Sipura G.726 is works
great.
Hmm..
On Sun, 7 Mar 2004, Greg Boehnlein wrote:
On Sun, 7 Mar 2004, James Golovich wrote:
On Sun, 7 Mar 2004, Greg Boehnlein wrote:
On Mon, 8 Mar 2004, Master Abi wrote:
Upgrade to the latest CVS and ast_rtp_read/write warnings will
disappear. GS .50 is buggy. Voice is
Dear all,
I have 2 questions:
1)
I've downloaded the zaptel CVS today (8 April 2004)
I've made sym link from /usr/src/linux-2.6.3 to /usr/src/linux-2.6
make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules
make[1]: Entering directory `/usr/src/linux-2.6.3-patch'
*** Warning: Overriding
My tests with the 7960 prove that ulaw allows perfect conferencing ...
Thanks, John !
- Chris
- Original Message -
From: John Fraizer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, March 06, 2004 9:07 AM
Subject: Re: [Asterisk-Users] 7960 conference ?
Chris Clifton wrote:
Looking at the # dialed, it looks like you need to strip the 9 off of your
${EXTEN} like this ${EXTEN:1}
Scott
- Original Message -
From: oliver vermeulen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 04, 2004 12:53 PM
Subject: [Asterisk-Users] Voicepulse error
Hi
quote who=John Wang
Vijay,
What is the rate of NuFone.net, not listed on their
web though.
I know voicepulse is $7.99 / month .
--John J.Wang
Nufone.net has no monthly fee. Think of it as a prepaid calling card.
You put money into an account, then use it by the minute.
US calls are 2.9
arhh - I did a checkout the 4th of marts - I will do a new checkout
/HHA
When did you checkout your version of Asterisk from CVS ??
This feature was put into CVS on the 6'th as a fix for bug #1107 but I
have not seen it in v1-0_stable.
___
What checkout name should I do ?
Just asterisk ?
# cd /usr/src# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot#
cvs login - the password is anoncvs.# cvs checkout asterisk
/HHA
This is a new feature, that's why it is NOT in 1.0-stable.
Only bugfixes go into
The G726 codec is not in the 1.0 stable branch, only in the HEAD branch of
CVS
Yes, unless you apply the patch from bugs.digium.com! ;) Which I did,
about 10 minutes after it was posted. ;)
The only patch that was posted was for the file format G726, not the codec
g726.
So you
Hello-
I asked this question a LONG time ago (when I
first got started with *), but seem to have lost
the answer in between my multiple Windows XP
repairs.
Has anyone experimented with or achieved PLAR
(private line auto ringdown) capability with
asterisk?
thx,
cedrick
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