On Mon, 18 Jan 2010, ev...@disruptor.nl wrote:
Hey Ira,
It seems after a several testing, that the wait(1) seems to solve the issue.
Only now weirdly enough the phone keeps ringing if the caller hangs up
before i picked up the phone (pstn call)
Regards,
Evert
Hi Evert
I have been
Hey Remco,
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,Dial(SIP/phone)
A bit more simplified then what i have in my config, but exactly the same
order.
Regards,
Evert
On Tue, 19 Jan 2010 10:46:52 +0100 (CET), Remco Barendse
aster...@barendse.to wrote:
On Mon, 18 Jan 2010,
On Wed, 2010-01-13 at 11:13 -0500, Kristian Kielhofner wrote:
On Wed, Jan 13, 2010 at 10:39 AM, listu...@spamomania.co.uk
listu...@spamomania.co.uk wrote:
..snip..
I've not been able to get that out of them, but I don't *think* It's
Asterisk based because they say:
Unfortunately, our
listu...@spamomania.co.uk wrote:
I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to
recognize digits pressed on a keypad coming in from a Sipgate trunk.
There answer was to set this:
dtmfmode=rfc2833
in the general section of sip.conf
This has made no difference.
On Tue, 2010-01-19 at 13:15 +0100, joern wrote:
listu...@spamomania.co.uk wrote:
I raised an issue with Sipgate because my Asterisk 1.6 plain refuses to
recognize digits pressed on a keypad coming in from a Sipgate trunk.
There answer was to set this:
dtmfmode=rfc2833
in the
Hi,
I've now set dtmfmode=rfc2833 and that seems to have fixed it
John
2010/1/7 John Taylor j...@vetsurgeon.org.uk:
We're now getting this problem on outgoing calls. I've forced the port
to 100FD but still no joy. Anyone any ideas how to debug this- have
added verbose to logger.conf
Hi all,
I've now set dtmfmode=rfc2833 instead of inband and that seems to have fixed it
John
2010/1/4 John Taylor j...@vetsurgeon.org.uk:
I have recently moved our asterisk server from our LAN to a Debian
Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our
network. Our
Hello, list.
First of all i want to say sorry for my english.
Long story short, on my future work i'll deal with asterisk and now i
have a test case. But i'm very young to asterisk and don't have a lot
of time so any help is appreciated.
Test case:
We have e1 trunk and multi-channel sip line.
Just a WAG - Playback is freaking out because you have the wav and gsm file
there concurrently. Remove the wav file and try it again.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zhang Shukun
Sent: Tuesday,
I'm having a problem receiving incoming faxes and I'm hoping someone
here can help me out.
My system is a PBX in a Flash with one dahdi card for my incoming analog
lines and another dahdi card for my analog devices (fax and modem).
My dahdi-channels.conf file looks like:
; Autogenerated by
Hello,
excuse my unknowing question, but how can I open the dialplan in hex-mode?
What file is it? Normaly, my dialplan ist read in textmode from
/etc/asterisk/extensions.conf. Sure, I can insert a new line with a hex-editor
into the textfile, but then the line got wrapped:
WARNING[1450]:
Well, never mind on this (didn't get any responses anyways). I basically
removed the meetme announcement options and wrote the functionality from
scratch into my AGI framework along with an announcement queuing daemon that
runs continuously every second in the background that generates a call
This link might help
http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dominik
Sent: Tuesday, January 19, 2010 11:01 AM
To: Asterisk Users Mailing List -
I know in v1.6 its part of logger.c but I noticed this:
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=11625
However, it doesn't seem to ever been applied to any version of 1.4.x
branch..
Nor can I figure out what it was applied to?
This is over 3 years old, you
of figured it would have been applied to
1.4 at some point in time..
Any ideas?
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Hi all,
I've got 12 Polycom 430's behind a NAT that are working pretty well except for
one thing:
If one of my users is on the phone when another call comes in, there is about
a 10 second time during which the user can't hear the call they're on. Then
it returns to normal.
My research
Hello,
When a new mailbox is defined in voicemail.conf, it seems nothing is changed
in /var/spool/asterisk/voicemail
and the /var/spool/asterisk/voicemailcontext/vmid/ hierarchy is created
whenever a message is dropped.
Is this correct ?
If positive, is it safe (ie is this a feature) to create
I don't think my previous answer was very clear.
Please, apologize for that.
Some time ago, I also asked for the same feature as yours : to be able to
send empty strings to IP phone Thomson ST2030.
Tilghman Lesher from Digium was kind enough to code a patch for 1.4 (svn
187362).
As
2010/1/19 Danny Nicholas da...@debsinc.com
This link might help
http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText
Yes but the trouble is you can't send empty strings with SendText.
With this specific IP phone, you sometimes have to.
Cheers
-Original Message-
From:
On Tuesday 19 January 2010 13:03:19 Olivier wrote:
I don't think my previous answer was very clear.
Please, apologize for that.
Some time ago, I also asked for the same feature as yours : to be able to
send empty strings to IP phone Thomson ST2030.
Tilghman Lesher from Digium was kind enough
Hm,
(svn 187362) Permit zero-length text messages in SIP
Maybe a zero-length text also works to clear the display of Thomson 2030:
But with SendText it doesn't:
-- Executing [...@intern:1] Answer(SIP/4711-0016, ) in new stack
-- Executing [...@intern:2] SendText(SIP/4711-0016, )
I'm using realtime sip peers and I need to enable a range of IP
addresses for a peer.
I have:
deny = 0.0.0.0/0.0.0.0
permit= xxx.yyy.zzz.0/255.255.255.0
mask = 255.255.255.0
defaultIP = xxx.yyy.zzz.112
host = xxx.yyy.zzz.112
Addresses other than .112 are being denied. Can
okay. Never mind. The redirecting number information is indeed available via
${CALLERID(rdnis)} variable. I was testing with two different phones and it
happens that our ISDN provider is having problem sending RDNIS information
on one of the phone that got me confused.
Thanks.
On Tue, Jan 19,
You need to set: host=dynamic Otherwise only .112 is allowed.
-Jonathan
On Tue, Jan 19, 2010 at 1:17 PM, Bruce Ferrell bferr...@baywinds.org wrote:
I'm using realtime sip peers and I need to enable a range of IP
addresses for a peer.
I have:
deny = 0.0.0.0/0.0.0.0
permit =
Thank you!
Jonathan Thurman wrote:
You need to set: host=dynamic Otherwise only .112 is allowed.
-Jonathan
On Tue, Jan 19, 2010 at 1:17 PM, Bruce Ferrell bferr...@baywinds.org wrote:
I'm using realtime sip peers and I need to enable a range of IP
addresses for a peer.
I have:
deny
2010/1/19 Dominik d0m1...@geekmail.de
Hm,
(svn 187362) Permit zero-length text messages in SIP
Maybe a zero-length text also works to clear the display of Thomson 2030:
But with SendText it doesn't:
-- Executing [...@intern:1] Answer(SIP/4711-0016, ) in new
stack
-- Executing
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