Hi all,
Today's jam-packed sessions include the security theme for the first
hour or so, then a debate about hosted vs local VoIP services.
Hour one guests are Sjur Usken, telecom consultant who has been
working with VoIP since 2002 and helping companies migrate to an all
IP world and Sandro
Hi all,
Here is the script i want to make
- Caller call to a number to record a message
- Asterisk answer and start recording message as following
+ User press * to start recording
+ Record is finished if:
+ User press #
+ OR message duration
OMG I overlooked that portion Please honor my apology.
On Thu, Mar 11, 2010 at 6:23 PM, Jeff LaCoursiere j...@jeff.net wrote:
On Fri, 12 Mar 2010, Angelito Manansala wrote:
If you are having trouble reading this email, read the online
version
Hello list,
I have nat=no and qualify=no in my sip peer definition and still my CLI
is flooded with :
[Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985
handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (30ms /
2000ms)
[Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985
Probably has anyone idea how dtmf payload type could be changed in Asterisk
say to 100?
On Wed, Mar 10, 2010 at 2:53 PM, Katerina Borin katerin.bo...@gmail.comwrote:
Hello,
I encountered the dtmf problem between my asterisk box (1.4.23) and
suppliers gateway (unknown vendor). I have dtmf mode
12 mar 2010 kl. 10.45 skrev Katerina Borin:
Probably has anyone idea how dtmf payload type could be changed in Asterisk
say to 100?
On Wed, Mar 10, 2010 at 2:53 PM, Katerina Borin katerin.bo...@gmail.com
wrote:
Hello,
I encountered the dtmf problem between my asterisk box (1.4.23) and
Edit logger.conf and set the desired log level.
To disable the messages below just remove the severity notice from console.
console = notice,warning,error,debug
Alex
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di jonas kellens
On 12 Mar 2010, at 10:05, Alexandru Oniciuc wrote:
Edit logger.conf and set the desired log level.
To disable the messages below just remove the severity notice from
console.
console = notice,warning,error,debug
Because if you can't see it it's not broken?
S
--
I figured that out, i can use monitor() function to record and using a
loop to count down 60s.
But I dont think it is best solution, any suggestion is appreciated.
And still, how can i capture '#'?
On Fri, 2010-03-12 at 15:03 +0700, Pham Quy wrote:
Hi all,
Here is the script i want to make
That is indeed an option, thank you.
It also went away by restarting Asterisk, but this is not desirable in
production environment.
On Fri, 2010-03-12 at 11:05 +0100, Alexandru Oniciuc wrote:
Edit logger.conf and set the desired log level.
To disable the messages below just remove the
On Fri, 12 Mar 2010, Pham Quy wrote:
I figured that out, i can use monitor() function to record and using a
loop to count down 60s.
But I dont think it is best solution, any suggestion is appreciated.
And still, how can i capture '#'?
Have you reied reading the manual, or the wiki, or even
If I have this is sip.conf :
[general]
disallow=all
allow=g729
allow=alaw
The prefered codecs set in my Grandstream phone is G729, alaw.
In the sip peer definition I have commented out 'disallow=' and
'allow='.
The prefered codecs set in the Zoiper softphone is alaw, gsm.
In the sip peer
Am 10.03.2010 17:33, schrieb Kevin P. Fleming:
Klaus Darilion wrote:
That's weird. AFAIK Asterisk does not allow multiple ranges. Maybe they
are having 2 ranges for RTP and UDPTL (T.38). Asterisk allow
configuration of different ranges for UDPTL and RTP (although it
shouldn't be a problem
Am 02.03.2010 13:29, schrieb Magnus Benngård:
Hi!
Did a setup of 2 peers as Klaus suggested, it worked thx!
Has anyone thought about the possibility to add multiple ip/hosts to
host=?
I my case: host=130.244.190.42,130.244.190.46 or
host=sip-corporate1.tele2.se,sip-corporate2.tele2.se
Quoting Jason Aarons (US) jason.aar...@us.didata.com:
I'm experiencing runaway ringing too, can we make this a class action
against someone?
Strangely enough, I have experienced this on what is a small domestic
system - when a call is answered, sometimes other SIP phones
(softphones only
12 mar 2010 kl. 12.01 skrev Klaus Darilion:
Am 02.03.2010 13:29, schrieb Magnus Benngård:
Hi!
Did a setup of 2 peers as Klaus suggested, it worked thx!
Has anyone thought about the possibility to add multiple ip/hosts to
host=?
I my case: host=130.244.190.42,130.244.190.46 or
2010/3/10 Tilghman Lesher tles...@digium.com
On Wednesday 10 March 2010 02:09:54 voipas wrote:
Does asterisk func odbc support multi query? I'm executing stored
procedure which returns two tables. With tsql command I can see both
tables. But asterisk only shows the first.
My database
Hi,
I am just moving from Asterisk 1.2+bristuff up to 1.6.2, a huge leap
:) I was wondering if someone could point me at 3 things that I appear
to have lost?
1) ZapEC(off) - Is there an equivalent dialplan command to request no
EC on a channel before dialling in DAHDI?
2) rxfax(file.tiff) - I
Hi,
I'm one of those people who still need to maintain * 1.2 systems and cannot
easily upgrade. :-(
My 1.2 systems were very stable until I upgraded from 1.2.37 to 1.2.40. I have
made some changes within my dialplan but nothing unusual.
Today I've had a crash:
I would also add the following :
sip.conf has :
[general]
disallow=all
allow=g729
allow=alaw
allow=gsm
And again the same in the sip peer definition :
disallow=all
allow=g729
allow=alaw
allow=gsm
sip debug shows :
[Mar 12 15:28:23] Found audio description format G729 for ID 18
[Mar 12
Hi All,
I got this figured out, when the privacy is ON at the other end of the
server and when we get the Invite message to the server connected to PRI's,
just take the details from the invite message in the Dial plan and send the
calls as anonymous:
exten =
Hi
I'm trying to get ExtenSpy to work but it wont, I'm dialling a number
from my mobile which comes into our server and answering the number on a
particular SIP extension which all works fine. I'm then dialling an
exten from my own SIP extension which executes the ExtenSpy for the
correct
Hi Zoa,
Nice work.
1. It would be nice to have the T30 libraries and include files being
distributed to the root filesystem or externally defined DESTDIR.
2. Do you have any plans to put the libt30 integration automated via the
configure script in Asterisk ?
In addition, just playing
Hello,
I need a hand in choosing a small ATA, even with one FXS port,
that should do only fax with T38.
I've tried Grandstream (ht286 model) but the faxes go out
without ECM, even if the Fax machine has ECM enabled.
Is there anyone that can
Klaus Darilion wrote:
Is Asterisk really that thumb and announces port befores testing if it
actually can open the socket?
No.
Usually you have other services running on the same server to (e.g. DNS
uses UDP ports), and just specifying port=1000-1999 in rtp.conf does not
prevent that any
On Friday 12 March 2010 05:55:33 voipas wrote:
2010/3/10 Tilghman Lesher tles...@digium.com
On Wednesday 10 March 2010 02:09:54 voipas wrote:
Does asterisk func odbc support multi query? I'm executing stored
procedure which returns two tables. With tsql command I can see both
On Fri, Mar 12, 2010 at 4:25 AM, jonas kellens jonas.kell...@telenet.bewrote:
Are you using SIP realtime?
--
Thanks,
--Warren Selby
http://www.selbytech.com
--
_
-- Bandwidth and Colocation Provided by
On 03/13/2010 12:00 AM, Alexandru Oniciuc wrote:
Hello,
I need a hand in choosing a small ATA, even with one FXS port, that
should do only fax with T38.
I’ve tried Grandstream (ht286 model) but the faxes go out without ECM,
even if the Fax machine has ECM enabled.
Is there anyone that
Hi Alex,
I'm having the same problem and there is an open problem report about
it. However, if you modify your /etc/asterisk/cdr_custom.conf file to
add the field it will show up in your Master.csv log file but still not
in the DB record. You could in essence use the log file to rebuilt your
Hi Dave,
Thought I'd give you an update - I completely rebuilt my astdb the other
night by renaming it, having * recreate it and then re-creating all my
custom entries in it.
Didn't have any effect, I had somebody report false MWI notifications again
earlier this morning.
--
Matt
On Tue, Mar
Steve-
On 03/13/2010 12:00 AM, Alexandru Oniciuc wrote:
Hello,
I need a hand in choosing a small ATA, even with one FXS port, that
should do only fax with T38.
Ive tried Grandstream (ht286 model) but the faxes go out without ECM,
even if the Fax machine has ECM enabled.
Is
Hi Steve,
the remote device is an Hylafax Server that does ECM. The sending fax device,
that's attached to the ATA, is a Philips fax machine with ECM enabled. If I
send with the same machine but attached to a Patton 4114 with T38 enabled my
faxes go to the other end with ECM enabled and with
Hi,
I have a strange problem with all of our Polycom 550 650 phones. I am
running a TFTP server on my Asterisk server and option 66 Boot Host pointing to
Asterisk on my DHCP server. The auto-provisioning is working because the
phones are registering correctly with their extension. If I
hi,
i'm looking for documentation on configuring asterisk to work with a modem that
should work with an analog line. i don't see the info in the handbook or
reference manual or o'reilly's. any references and/or links, much appreciated.
thanks.
g.
We have a client with Asterisk 1.6 installed via yum (onto Centos). It did
not included the chan_h323 driver apparently, so we installed add-ons by
yum. We then got ooh323.
Is it possible to install the H.323 drivers without compiling from source?
--
I'm running Asterisk 1.6.2.5 with chan_skype on a x64 linux platform.
When a user calls from skype (not skype-in) to asterisk, dtmf (basically menus
for a conference system) works just fine.
But when a user from the inside (soft or hardware sip phone) calls out via
skype-out dtmf doesn't work.
I
Joakim Eriksson wrote:
I'm running Asterisk 1.6.2.5 with chan_skype on a x64 linux platform.
When a user calls from skype (not skype-in) to asterisk, dtmf (basically
menus for a conference system) works just fine.
But when a user from the inside (soft or hardware sip phone) calls out via
Thank for the help :)
Then i can just hope it gets fixed soon.
(But now that i know about it, its not as critical anymore).
//Joakim
On Mar 12, 2010, at 8:24 PM, Kevin P. Fleming wrote:
Joakim Eriksson wrote:
I'm running Asterisk 1.6.2.5 with chan_skype on a x64 linux platform.
When a user
I do not use ConfBridge() in a large installation. I use MeetMe on
1.6.0.*
The timing is different for ConfBridge, as it does not require DAHDI.
If you have that good of an experience with 1.4, why change anything?
I like new things. ConfBridge eliminates the need for an external timing
The Asterisk Development Team has announced the release of Asterisk 1.6.1.18.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.1.18 resolves several issues reported by the
community, and would have not been
The Asterisk Development Team has announced the release of Asterisk 1.6.2.6.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.6 resolves several issues reported by the
community, and would have not been
The Asterisk Development Team has announced the release of Asterisk 1.4.30.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.30 resolves several issues reported by the
community, and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 1.6.0.26.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.0.26 resolves several issues reported by the
community, and would have not been
i gave up on ReceiveFAX and uses iaxmodem/hylafax instead.
Tommy Botten Jensen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
Edwin Lam skrev:
Klaus Darilion wrote:
The backtrace is not useable. Try to rebuild Asterisk with the Don't
Optimize Option (make menuconfig and the the
Yes I am for most of my SIP peers.
On Fri, 2010-03-12 at 10:51 -0600, Warren Selby wrote:
On Fri, Mar 12, 2010 at 4:25 AM, jonas kellens
jonas.kell...@telenet.be wrote:
Are you using SIP realtime?
--
Thanks,
--Warren Selby
http://www.selbytech.com
--
Hello,
http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly
The link above indicates that it is possible to setup RTP streams to
directly flow between endpoints and completely bypass Asterisk. I would
like to know if this configuration would work when,
a) both
Hi Gordon,
What i'm doing now is that something like karaoke. While music is
playing back, caller voice is being record by the way i mentioned
earlier. I should give you the whole picture of what i'm doing.
I did google for it, and Monitor() function seem to be the best choice
to do that.
I
Here again, the script should be described as
- Caller call to a number
- Asterisk answer, play back music and start MONITORING as following
+ User press * to start MONITORING
+ Record is finished if:
+ User press #
+ OR
On 03/13/2010 02:03 AM, Jeff Brower wrote:
Steve-
On 03/13/2010 12:00 AM, Alexandru Oniciuc wrote:
Hello,
I need a hand in choosing a small ATA, even with one FXS port, that
should do only fax with T38.
I’ve tried Grandstream (ht286 model) but the faxes go out without ECM,
Is there something strange about using regular expressions in the context
to which incoming Skype calls go?
If I set up accounts, foobar1, foobar2, etc, it doesn't seem to work to
have:
exten = _foobarX,1,...
should it?
--
On 03/09/2010 07:31 AM, Edwin Lam wrote:
hi folks.
i recently upgraded asterisk to 1.6.1.17(from 1.2) and i'm having
problems with fax. after receiving fax with the ReceiveFAX app.
everything seems ok. the .tiff file was there, phone line seems
to hang up. then asterisk will crash. any
Hi All,
Found an issue with DUNDILOOKUP function in Asterisk 1.6.0.5.
I was using DUNDIQUERY (Set(ID=${DUNDIQUERY(${MNUM},priv,b)})) for
dundilookup and it was working fine.
But when I tried to use DUNDILOOKUP function
(Set(DL=${DUNDILOOKUP(${MNUM},priv,b)})), it didn't retuen me a
result.
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