Thank you for you reply?
is that mean STDERR couldn't show under Asterisk CLI mode?
it's only saved to some file?
2010/1/7 Steve Edwards asterisk@sedwards.com:
On Thu, 7 Jan 2010, Zhang Shukun wrote:
i use agi to send message back to Asterisk by STDERR, but why i could't
see the message
hello, all of users:
there are header files missed when you compile dahdi with kernel-2.6.29 or
2.6.33. i believe
that few files are affected: wctdm.c dahdi-base.c wcb4xxp/base.c, opvxa1200.c...
the errors look like these:
from
Hello,
STDERR goes to the original Asterisk process only, not any asterisk -r
connections that you may use. If you launch Asterisk in a screen like we
do, then you can see it and log it in context with when the output is
happening. We find it very useful to do it this way.
MATT---
On 1/7/10,
On Thu, Jan 07, 2010 at 04:19:21PM +0800, james.zhu wrote:
hello, all of users:
there are header files missed when you compile dahdi with kernel-2.6.29 or
2.6.33. i believe
that few files are affected: wctdm.c dahdi-base.c wcb4xxp/base.c,
opvxa1200.c...
the errors look like these:
Hi,
Am Dienstag, den 05.01.2010, 15:38 +0100 schrieb Christian Theune:
Hi,
I tried again getting DTMF detection on my ISDN devices with dahdi going
again. I used the channel debug to see whether asterisk sees the frames
and detects them as DTMF.
Interestingly here's what works:
1.
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
Thanks for any help
John
2010/1/4 John Taylor j...@vetsurgeon.org.uk:
I have recently moved our asterisk server from our LAN
Hi,
I have occasionally experienced the same problem too, and I suspect it was
caused by some spikes in network traffic (e.g. for an intensive file transfer)
that delayed too much SIP OPTION response, so that Asterisk marked these
devices as UNREACHABLE; I was able to use the devices too: in
Hello there!
If your box has a live Internet connection, then all you need is a sip
provider.
Back to when I lived in the UK, there was this voipuser.org which gave me
a fixed british number for free, and some outbound call minutes too.
I'm sure that if you search around for SIP Providers, you
2010/1/7 David Backeberg dbackeb...@gmail.com
On Wed, Jan 6, 2010 at 6:23 PM, Olivier oza-4...@myamail.com wrote:
The second time I'm dialing an internal extension attached to the same
ReceiveFAX application :
2. sendfax/hylafax/iaxmodem asterisk spandsp
In the 2nd case,
Olivier schrieb:
2010/1/7 David Backeberg dbackeb...@gmail.com
mailto:dbackeb...@gmail.com
On Wed, Jan 6, 2010 at 6:23 PM, Olivier oza-4...@myamail.com
mailto:oza-4...@myamail.com wrote:
The second time I'm dialing an internal extension attached to the
same
Hello,
I've read in Mantis that asterisk.conf's internal timing option could
positively impact Asterisk behaviour during faxing (
http://issues.asterisk.org/view.php?id=16374).
Before using it, I would be very pleased to read a line or two about its
use.
I've read
Hello,
I've upgraded asterisk to 1.6.0.20 version and found , if I want change
queue strategy to linear, I must restart Asterisk:
[Jan 7 08:16:10] WARNING[9578]: app_queue.c:1304 queue_set_param: Changing
to the linear strategy currently requires asterisk to be restarted.
[Jan 7 08:16:10]
Can anyone shed any light on this error?
Will
Jan 7 11:03:34 asterisk pppd[9168]: Plugin zaptel.so loaded.
Jan 7 11:03:34 asterisk pppd[9168]: Zaptel Plugin Initialized
Jan 7 11:03:34 asterisk pppd[9168]: Using zaptel device 'stdin'
Jan 7 11:03:34 asterisk pppd[9168]: pppd 2.4.4 started by
2010/1/7 Johann Steinwendtner steinwendt...@gmx.net
Olivier schrieb:
2010/1/7 David Backeberg dbackeb...@gmail.com
mailto:dbackeb...@gmail.com
On Wed, Jan 6, 2010 at 6:23 PM, Olivier oza-4...@myamail.com
mailto:oza-4...@myamail.com wrote:
The second time I'm dialing
Hello Tiago, I think that this is the route I will be trying to go as
its a proof of concept sort of project. After that - we'll see.
Thank you!
On Thu, Jan 7, 2010 at 4:43 AM, Tiago Geada tiago.ge...@gmail.com wrote:
Hello there!
If your box has a live Internet connection, then all you need
PS: If you compile Asterisk from source after installing spandsp, SendFAX
and ReceiveFAX would automatically be included.
I opened another thread about that but I doubt that both SendFAX and
ReceiveFAX behave exactly the same (no side effect), no matter the installed
spandsp version.
I would be
7 jan 2010 kl. 10.21 skrev Aggio Alberto:
Hi,
I have occasionally experienced the same problem too, and I suspect it was
caused by some spikes in network traffic (e.g. for an intensive file
transfer) that delayed too much SIP OPTION response, so that Asterisk marked
these devices as
7 jan 2010 kl. 12.00 skrev Olivier:
Hello,
I've read in Mantis that asterisk.conf's internal timing option could
positively impact Asterisk behaviour during faxing
(http://issues.asterisk.org/view.php?id=16374).
Before using it, I would be very pleased to read a line or two about its
Nobody can help me on this??
--
Hi all,
I want to compile zaptel in data mode but i got this errors:
/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c: In function âzt_xmitâ:
/usr/src/zaptel-1.4.12.1/kernel/zaptel-base.c:1618:
hi,
i want to dial a number to let two phone ring at the same time or
alternative ring,
how should i configure in asterisk? or how to right the Dialplan code?
Thanks very much!
--
Best regards,
Sucan
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On Thu, Jan 7, 2010 at 2:38 PM, Zhang Shukun bit...@gmail.com wrote:
hi,
i want to dial a number to let two phone ring at the same time or
alternative ring,
how should i configure in asterisk? or how to right the Dialplan code?
exten = 12345,1,Dial(${PHONE1}${PHONE2})
each phone variable
On 01/07/2010 07:21 PM, Olivier wrote:
PS: If you compile Asterisk from source after installing spandsp,
SendFAX and ReceiveFAX would automatically be included.
I opened another thread about that but I doubt that both SendFAX and
ReceiveFAX behave exactly the same (no side effect), no matter
Hello users,
i am working on directly calling the numbers from the sip provider of my
choice from asterisk using Dial command as follows.
extensions.conf
[dial-out]
exten = _XX,1,NoOp(Dialing out)
exten =
_XX,n,Dial(SIP/1{EXTEN}:password:md5secret:authname:tarnsp...@host:port
,
Thank you!
but how can i determine whether ring at the same time or
alternative ring?
BTW, the uri
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.con
can't open.
Could you paste it again?
2010/1/7 Randy R randulo2...@gmail.com:
On Thu, Jan 7, 2010 at 2:38 PM,
On Thu, Jan 07, 2010 at 07:27:00AM -0600, mos...@infolog.mr wrote:
Nobody can help me on this??
--
Hi all,
I want to compile zaptel in data mode but i got this errors:
On Thu, Jan 7, 2010 at 3:07 PM, Zhang Shukun bit...@gmail.com wrote:
Thank you!
but how can i determine whether ring at the same time or
alternative ring?
BTW, the uri
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.con
It got mistyped or cut, it's
Can I be taken off the mailing list please.
Thanks.
rick
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To UNSUBSCRIBE or update options visit:
read your posting and it will tell you haw to remove yourself.
On Thu, Jan 7, 2010 at 10:49 AM, Rick Dean ric.d...@gmail.com wrote:
Can I be taken off the mailing list please.
Thanks.
rick
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Go to this address for information on how to remove yourself:-
http://lists.digium.com/mailman/listinfo/asterisk-users
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rick Dean
Sent: 07 January 2010 15:50
To:
I have several sip stations that on a that are on a nat'd network behind a
nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc.
However, I can't get any of my phones to Transfer or Blind Transfer..
I search and search, and well, just about gone nuts on this one.
Here is
On Thu, Jan 7, 2010 at 11:15 AM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
I have several sip stations that on a that are on a nat'd network behind a
nice friend firewall.. no audio path issues, 2 way audio works,
etc,etc,etc.
However, I can't get any of my
7 jan 2010 kl. 17.15 skrev William Stillwell (Lists):
I have several sip stations that on a that are on a nat'd network behind a
nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc.
However, I can't get any of my phones to Transfer or Blind Transfer..
I search
Ok, im gonna go craw back under a rock..
Third line of my sip.conf
allowtransfer=no
Thanks for those who responded (Steve Ollie)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olle E.
Johansson
Sent:
Ever since upgrading to 1.6 I get messages like these. I want
everything else that shows up, but is there a way to make all the dns
messages go away?
Ira
doing dnsmgr_lookup for 'gw5.telasip.com'
doing dnsmgr_lookup for 'sipconnect.ipcomms.net'
doing dnsmgr_lookup
Steve Totaro wrote:
read your posting and it will tell you haw to remove yourself.
On Thu, Jan 7, 2010 at 10:49 AM, Rick Dean ric.d...@gmail.com
mailto:ric.d...@gmail.com wrote:
Can I be taken off the mailing list please.
Thanks.
rick
I've never seen that in Outlook. What client do you use?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Francesco
Peeters
Sent: 07 January 2010 18:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Gmail DOES process those headers...
And a proper mail client will also parse the headers and provide unsubscribe
information/buttons based on that
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Dan Journo wrote:
I've never seen that in Outlook. What client do you use?
Lately I have been using Thunderbird with an RFC2369 header plugin.
--FP
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I use gmail but don't see any buttons for unsubscribe or anything like that?
Also, gmail defaults to top posting...which seems to upset some people
'round these parts. I have yet to find a way to make gmail not top-post by
default...
On Thu, Jan 7, 2010 at 1:16 PM, Francesco Peeters
At 2:01 PM on 07 Jan 2010, Dan Journo wrote:
I've never seen that in Outlook. What client do you use?
Claws Mail provides a Mailing-List sub-menu under the Message menu,
which includes Post, Subscribe and Unsubscribe options, among others.
It's amazing what paying attention to standards can do
On Thu, Jan 7, 2010 at 1:27 PM, Warren Selby wcse...@selbytech.com wrote:
I use gmail but don't see any buttons for unsubscribe or anything like that?
Click on 'show details' at the top of the message and it will expand
to show those options.
I just found them over Christmas as I was trying to
On 7 Jan 2010, at 19:01, Dan Journo wrote:
I've never seen that in Outlook. What client do you use?
He said 'proper' mail client ;)
*holy war*
Sorry...
S
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asterisk-users
I haven't had a good mailing list war in a while.
Yes, gmail DOES default to top posting, because bottom posting is silly (in
general, but especially for a client that hides quoted text (like gmail)). Top
posting is modern. And better. And doesn't make me scroll through 10 thousand
messages
http://www.washington.edu/computing/mailman/faqs/mailman.email.html
Em 07/01/2010, às 15:29, C. Chad Wallace
cwall...@lodgingcompany.com escreveu:
At 2:01 PM on 07 Jan 2010, Dan Journo wrote:
I've never seen that in Outlook. What client do you use?
Claws Mail provides a Mailing-List
My friends,
I'm having some problems in my Asterisk, the thing is that Asterisk seem to
be crashed (or dead) sometimes (2 times in 3 weeks)
I noticed this today, when i could not make any internall call, tha calls to
the voicemail (*1) did not work it just don't say nothing, nothing appears
in
On Thu, 7 Jan 2010 15:58:43 -0430
Danny Dias ing.diasda...@gmail.com wrote:
My friends,
I'm having some problems in my Asterisk, the thing is that Asterisk
seem to be crashed (or dead) sometimes (2 times in 3 weeks)
[Jan 5 16:51:19] WARNING[6787] channel.c: Channel allocation failed:
On Thu, Jan 07, 2010 at 12:50:03AM -0600, Doug wrote:
At 00:22 1/7/2010, Tzafrir Cohen wrote:
On Wed, Jan 06, 2010 at 11:41:54PM -0600, Doug wrote:
At 16:49 1/5/2010, Tzafrir Cohen wrote:
On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote:
Hi,
Having problems with
Has there been any improvement with app_fax ?
I stopped using it as I had a high failure rate with inbound faxes (10%+)
1000 faxes a week ,with over a 100 failures can get quite annoying from
people complaining.. I could get it to fail everytime I tried sending a
solid black fax page.
(ie, take
Hi,
I'm having a bit of a problem with storing voicemail messages in an
odbc database. I *think* I've got everything configured correctly but
messages are stored on the asterisk server instread of in the database.
System info
64 bit redhat RHEL 5.1
Asterisk 1.4.26
unixODBC installed
On Thu, Jan 07, 2010 at 05:05:09PM -0500, William Stillwell (Lists) wrote:
Has there been any improvement with app_fax ?
Builds with spandsp 0.0.6 (as opposed to older versions that required
older versions of spandsp).
It is also a more well-behaving Asterisk app in its logging (does not
keep
Hi All,
I'm running an AGI, calling a perl script the does number lookups to a
remote server. I would like to put a timeout in the script. The
problem I'm running into is if the DNS server is not responding, the
script hangs and waits for 30 seconds before returning to the Asterisk
dialplan. I
On Thu, 7 Jan 2010, JR Richardson wrote:
I'm running an AGI, calling a perl script the does number lookups to a
remote server. I would like to put a timeout in the script. The
problem I'm running into is if the DNS server is not responding, the
script hangs and waits for 30 seconds
What are the LDAP searches like?
On 05/01/2010, Jorge Salamero Sanz ben...@cauterized.net wrote:
Hi all,
I've updated Asterisk trunk LDAP schema [0] [1] to include queues and other
attributes needed for a working LDAP backend (I'll open a bug to include
these
changes on svn).
SIP users
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson jmr.richard...@gmail.com wrote:
problem I'm running into is if the DNS server is not responding, the
script hangs and waits for 30 seconds before returning to the Asterisk
dialplan. I would like a timeout of 1 second, then return.
A few things...
On Thursday 07 January 2010 16:20:38 Alex Sharaz wrote:
I'm having a bit of a problem with storing voicemail messages in an
odbc database. I *think* I've got everything configured correctly but
messages are stored on the asterisk server instread of in the database.
System info
64 bit
On 01/08/2010 06:05 AM, William Stillwell (Lists) wrote:
Has there been any improvement with app_fax ?
I stopped using it as I had a high failure rate with inbound faxes (10%+)
1000 faxes a week ,with over a 100 failures can get quite annoying from
people complaining.. I could get it to fail
On Thursday 07 January 2010 18:59:24 David Backeberg wrote:
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson jmr.richard...@gmail.com
wrote:
problem I'm running into is if the DNS server is not responding, the
script hangs and waits for 30 seconds before returning to the Asterisk
dialplan. I
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson jmr.richard...@gmail.com
wrote:
problem I'm running into is if the DNS server is not responding, the
script hangs and waits for 30 seconds before returning to the Asterisk
dialplan. ?I would like a timeout of 1 second, then return.
On Thursday
Careful, or Steve will un top post YOU!
David Gibbons wrote:
I haven’t had a good mailing list war in a while.
Yes, gmail DOES default to top posting, because bottom posting is
silly (in general, but especially for a client that hides quoted text
(like gmail)). Top posting is modern. And
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson
jmr.richard...@gmail.com
wrote:
problem I'm running into is if the DNS server is not responding, the
script hangs and waits for 30 seconds before returning to the Asterisk
dialplan. ?I would like a timeout of 1 second, then return.
On
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson
jmr.richard...@gmail.com
wrote:
problem I'm running into is if the DNS server is not responding, the
script hangs and waits for 30 seconds before returning to the
Asterisk dialplan. ?I would like a timeout of 1 second, then return.
On Thu, 7
On Thursday 07 January 2010 21:17:52 JR Richardson wrote:
On Thu, 7 Jan 2010, Tilghman Lesher wrote:
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson wrote:
problem I'm running into is if the DNS server is not responding, the
script hangs and waits for 30 seconds before returning to the
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