If you change anything in your mysql sip table you do not need to reload
the modue, what you need to do is
sip prune realtime peername
from the CLI
Without reload prune does not take effect in 1.4.x
And after reload all registrations are lost.
So basically Asterisk Realtime is big mess from
I have a CentOS release 4.7 box running asterisk-1.4.26.1 with
dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0
I regularly get these messages, is this something i should be worried
about?
[Aug 21 01:05:07] VERBOSE[4343] logger.c: codec_g726.so = (ITU
G.726-32kbps G726 Transcoder)
[Aug 21 01:05:07]
I have to disagree with you there, we use 1.4.17 and sip prune realtime
works fine
Mindaugas Kezys wrote:
If you change anything in your mysql sip table you do not need to reload
the modue, what you need to do is
sip prune realtime peername
from the CLI
Without reload prune does not take
Howz about creating the directory manually if it dont exists and making a
new file with name transcode in it. like (/dev/dahdi/transcode)
I dont think so you need to worry about this
Best Regards
Shakeel Abbas
On Fri, Aug 21, 2009 at 2:27 PM, Remco Barendse aster...@barendse.towrote:
I have
On Thu, Aug 20, 2009 at 05:36:47PM -0700, Steve Edwards wrote:
On Fri, 21 Aug 2009, Lee, John (Sydney) wrote:
How can I check what format my channels are using?
Format? Format is the envelope. Codec is the algorithm used to encode
and decode.
Try sip show peer peer-name to see which
On Fri, Aug 21, 2009 at 11:27:01AM +0200, Remco Barendse wrote:
I have a CentOS release 4.7 box running asterisk-1.4.26.1 with
dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0
I regularly get these messages, is this something i should be worried
about?
[Aug 21 01:05:07] VERBOSE[4343] logger.c:
Hi All
I have questions relation to the playback/background , record commands .
1. These commands takes parameters for a sound file(ie
playback(/home/sounds/hehe)). Is it possible that I have sound files saved
in Database (Postgres) and play sound files from there.
2. When we use Record command
Hi everybody,
I've found where the problem was. The A400E card had no FXS/FXO module,
well, in fact it had 2 FXS (I asked 2 FXO but the vendor sent me the card
with 2 FXS and I didn't realize) but I didn't supply external power to the
A400E so for the system it had no module.
So if you ever find
We use 1.4.18.1 and 1.4.26.1 and it does not work - settings are not changed
after prune, asterisk must be reloaded, sip reload or iax2 reload makes
changes.
But after that all devices loose registration.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
Hello,
How can I do t-38 passthrough with asterisk 1.6 ?
I know how to do with 1.4 but not with 1.6
Thank you
Cordialement,
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
___
-- Bandwidth and Colocation
Hi
I had a working system, until recently - its asterisk 1.6.1 from debian
- not the lastest as the last doesn't seem to work.
but somebody who rang me said my voice mail announcement was all
stuttery. so i dialed my voicemail box and its really stuttery...
so I have done a reboot and its just
On Fri, Aug 21, 2009 at 8:39 AM, Alex Samad a...@samad.com.au wrote:
Hi
I had a working system, until recently - its asterisk 1.6.1 from debian
- not the lastest as the last doesn't seem to work.
but somebody who rang me said my voice mail announcement was all
stuttery. so i dialed my
Hi,
I have an app that makes a call via originate or a call file the dumps
into an IVR context in extensions.conf. The call works fine, except that
the cdr never gets set ss ANSWERED. I tried a work around where the call
dumps to a context which then Dials(local/) to a second context which is
the
BERGANZ François wrote:
How can I do t-38 passthrough with asterisk 1.6 ?
I know how to do with 1.4 but not with 1.6…
There is no difference, the identical configuration should work. I would
recommend using the 1.6.0.14 or 1.6.1.5 release candidates (or any later
releases) as they contain a
When I receive a fax it is in g711
After pickup, the fax invite again with T38 in the SDP.
Have I something to insert in the dialplan or other to let the T38 passthrough ?
Cordialement,
Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-Message d'origine-
De :
On Fri, 21 Aug 2009, ABBAS SHAKEEL wrote:
I have questions relation to the playback/background , record commands .
1. These commands takes parameters for a sound file(ie
playback(/home/sounds/hehe)). Is it possible that I have sound files saved
in Database (Postgres) and play sound files
BERGANZ François wrote:
When I receive a fax it is in g711
After pickup, the fax invite again with T38 in the SDP.
Have I something to insert in the dialplan or other to let the T38
passthrough ?
No.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive
While this is not possible, it wouldn't be a big step to do a fastagi to
push and pop the sound to/from the database to /tmp/filetouse.xxx or
/home/sounds/filetouse.xxx. Even using a regular AGI and sox the process
would only take 1-2 seconds per file depending on size.
-Original Message-
I have that problem:
[Aug 21 15:57:35] WARNING[5198]: chan_sip.c:6737 get_ip_and_port_from_sdp:
Failed to read an alternate host or port in SDP. Expect audio problems
[Aug 21 15:57:35] WARNING[5198]: chan_sip.c:17425 handle_request_invite: Failed
to set an alternate media source on glared
2009/8/21 Ishfaq Malik i...@pack-net.co.uk
I have to disagree with you there, we use 1.4.17 and sip prune realtime
works fine
After a few test, I notice these events when I use
asterisk+mysql+realtime+sip
1) after a sip prune realtime peername, peername will not be reachable
by another
Hi,
I'm calling asterisk with a swedish PSTN-phone line with caller presentation
(DTMF) activated.
I'm using asterisk 1.4.20.1 and cannot upgrade unfortunately, so I have to stay
with this release.
I use a TDM800P 8 channel PSTN card working as answering phones (I connect a
phoneline with
On 21 Aug 2009, at 15:16, Johan Sandgren wrote:
Maybe an option in some conf-file could be wrong?
What is in your config file?
___
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
On Friday 21 August 2009 08:22:41 Steve Edwards wrote:
On Fri, 21 Aug 2009, ABBAS SHAKEEL wrote:
I have questions relation to the playback/background , record commands .
1. These commands takes parameters for a sound file(ie
playback(/home/sounds/hehe)). Is it possible that I have sound
Have you set the qualify column in the sip table?
harry R wrote:
2009/8/21 Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk
I have to disagree with you there, we use 1.4.17 and sip prune
realtime
works fine
After a few test, I notice these events when I use
Have you set the qualify column in the sip table?
yes and default set to yes
___
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AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
Tilghman,
In your opinion, how much pain is usually involved with bringing in
out-of-date branches? I suppose that would depend on the complexity of
the function?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
i got three replies because i asked three times :)
Thanks alot Steve , Dhanny , Tilghman.
Steve and Tilghman ... may be this happen for future generations :)
Dhanny i was also thinking same but wondering there may be a way to do it
direclty. thanks alot.
Cheers
Shakeel Abbas
On Fri, Aug
2009/8/21 Tzafrir Cohen tzafrir.co...@xorcom.com
On Fri, Aug 21, 2009 at 11:27:01AM +0200, Remco Barendse wrote:
I have a CentOS release 4.7 box running asterisk-1.4.26.1 with
dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0
I regularly get these messages, is this something i should be worried
I solved it myself.
It's so typical, I just need to send a question to the list, and then I can
finally solve it myself.
I guess I just need to know you are there to help me or something.
The error was (found it on digiums site, how to set up callerid, and what to do
if problems occurred)
BERGANZ François wrote:
I have that problem:
[Aug 21 15:57:35] WARNING[5198]: chan_sip.c:6737 get_ip_and_port_from_sdp:
Failed to read an alternate host or port in SDP. Expect audio problems
[Aug 21 15:57:35] WARNING[5198]: chan_sip.c:17425 handle_request_invite:
Failed to set an
On Friday 21 August 2009 09:42:13 Danny Nicholas wrote:
In your opinion, how much pain is usually involved with bringing
in out-of-date branches? I suppose that would depend on the complexity
of the function?
Pretty much, yes. Also depends upon the complexity of the merged patches
On Fri, 21 Aug 2009, ABBAS SHAKEEL wrote:
i got three replies because i asked three times :) Thanks alot Steve ,
Dhanny , Tilghman.
Steve and Tilghman ... may be this happen for future generations :)
Dhanny i was also thinking same but wondering there may be a way to do
it direclty.
Ahan Steve how can i achieve this ?
Best regards
On Fri, Aug 21, 2009 at 9:26 PM, Steve Edwards asterisk@sedwards.comwrote:
On Fri, 21 Aug 2009, ABBAS SHAKEEL wrote:
i got three replies because i asked three times :) Thanks alot Steve ,
Dhanny , Tilghman.
Steve and Tilghman ...
On Fri, Aug 21, 2009 at 9:26 PM, Steve Edwards
If you told Asterisk to record to a file on a RAM disk and then
inserted the file's data in the database, you could achieve your goal
without a major performance hit.
Inelegant, but functional.
On Fri, 21 Aug 2009, ABBAS SHAKEEL wrote:
Did a lot of looking/digging on this and realized I might not have asked
clearly.
I'm generating the calls using call files, but can't seem to get the info to
show in HUD right. Is there something I could do in the call file to get
HUD to show the right thing?
On Thu, Aug 20, 2009 at 11:02 AM,
I don't think the HUD reads the CDR. IMO the HUD would read the callerID
from the line or from asterisk. When you say you set CID, did you set
CID(number), CID(name), CID(all) or something else? Have you tried just
making a simple call file (test.call) and tweaking that and copying to
On Fri, 21 Aug 2009, J. G. wrote:
Did a lot of looking/digging on this and realized I might not have asked
clearly.
I'm generating the calls using call files, but can't seem to get the
info to show in HUD right. Is there something I could do in the call
file to get HUD to show the right
Thanks for kind help steve:)
I will try on monday and will provide you feed back
Cheers
On Fri, Aug 21, 2009 at 10:10 PM, Steve Edwards
asterisk@sedwards.comwrote:
On Fri, Aug 21, 2009 at 9:26 PM, Steve Edwards
If you told Asterisk to record to a file on a RAM disk and then
inserted
First off this is not my work for extensions.conf it is modified from
http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbackl
ogin-to-standard-dialplan-methods-part-1/
So credit to Leif Madsen http://www.leifmadsen.com
But as to my question
[AgentLogin]
;A replaced
On Thu, Aug 20, 2009 at 08:45:58PM +0200, Stefan Tichy wrote:
Recording a call without mixing fails.
User hit '*1' to record call. filename: wav,auto-1250793354-24-22,m
Not even the Monitor application or the Monitor manager command do work.
ast_monitor_change_fname seems to return -1 all
hello friends,
i have to configures asterisk n my hardware details are
O.S - Ubuntu 8.04 Lts
Memory - 1 GB
Proccessor- core 2 duo
is any
one having a good link or how to related asterisk.
any help,support will
be higly appreciated
thx___
Start here, http://www.asterisk.org/support
~T
On Aug 21, 2009, at 1:22 PM, aster...@opensourcesolution.in
aster...@opensourcesolution.in
wrote:
hello friends,
i have to configures asterisk n my hardware details are
O.S - Ubuntu 8.04 Lts
Memory - 1 GB
Proccessor- core 2 duo
is
Here is my dialplan for my support queue:
exten = 201,1,Verbose(2,Doing support call)
exten = 201,n,Answer()
exten = 201,n,Wait(0.5)
exten = 201,n,Set(qac=${QUEUE_MEMBER(support,free)})
exten = 201,n,GotoIf($[${qac} 0]?HAVEAGNT)
exten = 201,n,Verbose(2,No agents free in support queue)
exten =
We have Valet Park working well with 1.4.25. We have programmed the Polycom
softkeys to include a park button that does a blind transfer to the park
extension.
Has anyone gotten the a button to activate when a particular park orbit is
in use? It would be great if you could press the button to
http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Ubuntu
On Fri, Aug 21, 2009 at 5:21 PM, trebaum treb...@telepaths.org wrote:
Start here, http://www.asterisk.org/support
~T
On Aug 21, 2009, at 1:22 PM, aster...@opensourcesolution.in
aster...@opensourcesolution.in
wrote:
When I reload chan_sip.so, it seems that connected terminals are no longer
detected by Asterisk because when I tape CLI command sip show peers,
there is no results displayed. Any reflexions about that ?
They won't be found in the CLI command until Asterisk receives another packet
from that
Hi,
We've just completed an interview with John Todd - the Open Source
Community Director for Asterisk.
Some interesting comments in it - hope you enjoy it:
http://www.venturevoip.com/news.php?rssid=2200
--
Cheers,
Matt Riddell
Director
___
On Friday 21 August 2009 20:20:14 Jonathan Thurman wrote:
When I reload chan_sip.so, it seems that connected terminals are no
longer detected by Asterisk because when I tape CLI command sip show
peers, there is no results displayed. Any reflexions about that ?
They won't be found in the
On Fri, Aug 21, 2009 at 08:53:23AM -0400, Steve Totaro wrote:
On Fri, Aug 21, 2009 at 8:39 AM, Alex Samad a...@samad.com.au wrote:
Hi
I had a working system, until recently - its asterisk 1.6.1 from debian
- not the lastest as the last doesn't seem to work.
but somebody who rang me
Ideally, the way realtime works, it shouldn't matter at all whether the record
exists in memory or in the database. In reality, there's a few cases where
the data needs to exist in memory for a particular event to occur correctly
(such as device state notifications). I think a better goal
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