Greetings List
Im interested to know how long the setup time is for a particular call on
asterisk. Is there any defined parameter that i can use to real this
behavior?
SETUP TIME = TIME BEFORE THE B-PART START RINGING
Thank you in advance
Sam
___
On Thursday 07 May 2009 18:21:55 arturo arturo wrote:
I have an asterisk 1.6.2 installation. I'm trying to configure func_odbc to
read some mysql tables... but every time I tried I got this message:
ERROR[24968] func_odbc.c: Unable to execute query [SELECT bloqueada FROM
funciones WHERE
It should be as simple as editing voicemail.conf :
; Voicemail Configuration
;
[general]
; Formats for writing Voicemail. Note that when using IMAP storage for
; voicemail, only the first format specified will be used.
format=wav49|wav|gsm
; Who the e-mail notification should appear to come from
Hi,
This may be completely wrong, but I have a feeling it may be related.
Have you enabled overlapdialling in zapata.conf for the channels
that are on the channelbank? If not, the 1st digit will be sent in,
not match the dialplan, and be hungup.
*7xxx is probably working because that matches a
Oh, and have you enabled Sangoma's DTMF detection in their config
file? That is probably also necessary.
Cheers,
Steve
2009/5/8 Steve Davies davies...@gmail.com:
Hi,
This may be completely wrong, but I have a feeling it may be related.
Have you enabled overlapdialling in zapata.conf for the
Been playing with G.722 in Asterisk 1.4.24.1 - using the back-ported
patches from http://carlton.oriley.net/drupal/node/12
Works just fine as far as I can tell - Grandstream phones anyway - playing
the G722 sound files, and calls between them.
Transcoding seems fine too - calling non G722
I have DTMF detection enabled. I will check if overlap makes a difference.
Right now I have it set to no.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Steve Davies davies...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
I am using TE110P card, I configured dahdi (dahdi_scan results below).
However, when I issue pro show spans, it does not give any outputs.
Am I missing some configuration?
active=yes
alarms=OK
description=Digium Wildcard TE110P T1/E1 Card 0
name=WCT1/0
manufacturer=Digium
devicetype=Digium
On Fri, May 08, 2009 at 05:25:08PM +0530, Jim Boykin wrote:
Hi,
I am using TE110P card, I configured dahdi (dahdi_scan results below).
However, when I issue pro show spans, it does not give any outputs.
Am I missing some configuration?
/etc/asterisk/chan_dahdi.conf
--
Hello,
I'm not understanding how to use GOSUB_RESULT in U() option from Dial app
(I'm using 1.6.1)
My extensions.ael is :
context mylocal {
2 = {
Dial(SIP/7530,,U(mynotify));
HangUp();
};
3 = {
Dial(SIP/7531);
Hi,
I wonder what is the difference between Transfer and Dial applications?
Could somebody give me an example of Transfer usage? (documentation and
voip-info looks poor a bit).
I'm using Asterisk 1.2.5 if it matters.
--
Mvh,
Aurimas Skirgaila
___
--
Despite the VPN overhead, running VOIP through VPN is good idea because VPN
reorders encapsulated UDP packets in correct order. Security matters as
well.
I'd suggest to route VNC packets rather over internet than VPN (so do I), as
VPN usually has the highest priority.
On Thu, May 7, 2009 at
On Fri, 8 May 2009, Aurimas Skirgaila wrote:
Despite the VPN overhead, running VOIP through VPN is good idea because VPN
reorders encapsulated UDP packets in correct order. Security matters as
well.
Reorders? How so? I think it will maintain the order, only if they have
arrived in the
(SIP/2022-083c53f0,
0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [...@macro-record-enable:4] AGI(SIP/2022-083c53f0,
recordingcheck|20090508-171018|1241782818.40) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090508
I am sending SIP or H323 calls to a carrier, and I need to store in the CDR
why the calls are rejected or why they hang up. In SIP, it can be code 503,
500, 488, etc. How do I get the information in my dialplan? I don't mean
$(DIALSTATUS}, but the real numeric code
F.Alves
Thanks. Solved.
On Fri, May 8, 2009 at 5:47 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Fri, May 08, 2009 at 05:25:08PM +0530, Jim Boykin wrote:
Hi,
I am using TE110P card, I configured dahdi (dahdi_scan results below).
However, when I issue pro show spans, it does not give any
Access-list 100 permit ip host asterisk server any
Class-map match-any voip
Match access-group 100
Policy-map voip
Class voip
Priority 256
Class class-default
Fair-queue
Interface fastethernet 0
Service-policy output voip
Above is what I do to prioritize 256kbit of outbound bandwidth
We have setup a system with TE110P in E1 mode. Everything works fine
except the DNID (number that was dialed.
) which is truncated to 7 digits. Any idea.
Thanks
Jim
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
)
-- Executing [...@macro-record-enable:4] AGI(SIP/2022-083c53f0,
recordingcheck|20090508-171018|1241782818.40) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090508-171018|1241782818.40: Outbound recording not
enabled
-- AGI Script
On Fri, May 8, 2009 at 3:45 PM, Jeff LaCoursiere j...@jeff.net wrote:
On Fri, 8 May 2009, Aurimas Skirgaila wrote:
Despite the VPN overhead, running VOIP through VPN is good idea because
VPN
reorders encapsulated UDP packets in correct order. Security matters as
well.
Reorders? How
Hi all...
Does anyone know if it is possible to override sip.conf settings in
extensions.conf
(for example: session-minse=90) without needing to create an overarching peer
in sip.conf
and selecting it specifically in the dial plan?
I'm on the 1.4 stable code base and looking to implement
On Thu, May 7, 2009 at 3:54 PM, Brent Davidson
br...@texascountrytitle.com wrote:
I've got multiple satellite office all linked back to the main office
via VPN. Each office has their own asterisk server which registers back
to the main office's Asterisk server. Each office also has a 1Mb
Josh Fuller wrote:
Hi all...
Does anyone know if it is possible to override sip.conf settings in
extensions.conf
(for example: session-minse=90) without needing to create an overarching peer
in sip.conf
and selecting it specifically in the dial plan?
You can do this to some extent
I would think that VoIP over VPN is a bad idea as UDP packets need to be
in realtime not corrected by the TCP of the VPN.
Garth van Sittert
Technical Director
BitCo
08600 24826
www.bitco.co.za
Aurimas Skirgaila wrote:
Despite the VPN overhead, running VOIP through VPN is good idea
because
Ah... ok thanks for that. In the end it was an SElinux problem. But I
was curious as to if I was missing some config somewhere. This clears
that up.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
On Friday 08 May 2009 10:07:43 Garth van Sittert wrote:
I would think that VoIP over VPN is a bad idea as UDP packets need to be
in realtime not corrected by the TCP of the VPN.
Not all VPNs use TCP. OpenVPN, in particular, uses UDP for the backbone.
--
Tilghman
Hi All,
Looking to gauge some opinions on redirect/proxy software.
I've two existing A*k servers out on the 'net. I need to redirect the
traffic going to those two servers, over to a new 3rd one.
Unfortunately, when the servers and clients were built, they used
hardcoded IPs, rather
Anyone who was at AMOOCON and who would deign to join us (ahem, Zoa,
alors?) to hash out what happened and make fun of the presenters,
please join us Friday at 6PM Paris time (5 PM UK) or 12 Noon EDT.
I myself was really pleased to be there and meet so many interesting
and amusing people.
Some
On Thu, May 7, 2009 at 11:44 AM, Steve Edwards
asterisk@sedwards.com wrote:
The meetmeadmin() dialplan function lets you specify a user to mute,
un-mute or kick. But how do you get a list of users in your dialplan?
You need a way to keep state. I use a database and AGI for that purpose.
Does anyone know if it is possible to override sip.conf settings in
extensions.conf
(for example: session-minse=90) without needing to create an overarching
peer in sip.conf
and selecting it specifically in the dial plan?
You can do this to some extent starting with Asterisk 1.6.1.
On Fri, May 08, 2009 at 12:11:40PM -0400, Mike van der Stoop wrote:
I have a Sangoma b600de analog card using dahdi 2.1.0.4 and I get the
following results (same dialplan, config etc):
Asterisk 1.6.0.1 = works fine
Asterisk 1.6.0.9 = can't dial out unless I dial in once or apply patch
I have a Sangoma b600de analog card using dahdi 2.1.0.4 and I get the
following results (same dialplan, config etc):
Asterisk 1.6.0.1 = works fine
Asterisk 1.6.0.9 = can't dial out unless I dial in once or apply patch
== http://bugs.digium.com/print_bug_page.php?bug_id=14577
Asterisk
Hello,
Here (http://downloads.digium.com/pub/telephony/codec_g729/README) are
instructions to install G729 software.
(I think I followed instructions step by step but g729 license doesn't seem
to show up).
My question is :
Is the command bellow still up to date ?
g729 show
Regards
PS:
Here
On Wed, May 6, 2009 at 8:17 AM, Johann Steinwendtner
steinwendt...@gmx.net wrote:
But it seems the Wait(60) lasts longer than 60 seconds:
-- Executing [...@from_meridian:1] NoOp(DAHDI/29-1, Test
wait) in new stack
-- Executing [...@from_meridian:2] Answer(DAHDI/29-1,
On Mon, May 4, 2009 at 10:52 PM, sean darcy seandar...@gmail.com wrote:
Receiving a fax with 1.6.1:
== Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on
'DAHDI/4-1'
-- Executing [...@incoming-pstn-line:1] NoOp(DAHDI/4-1, Fax
Detected) in new stack
-- Executing
-Original Message-
From: tiagodura...@gmail.com
Sent: Fri, 1 May 2009 11:02:58 -0400
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] AGI - Ways to create a call
On Fri, May 1, 2009 at 11:18 AM, Jimmy Godbout s...@inbox.com wrote:
Well, my question is: do you guys have
Thank you very much My problem was the res_odbc.conf file was not
configured... as soon as I enable the DSN it worked
2009/5/8 Tilghman Lesher tilgh...@mail.jeffandtilghman.com
On Thursday 07 May 2009 18:21:55 arturo arturo wrote:
I have an asterisk 1.6.2 installation. I'm trying to
Hello,
Page http://www.voip-info.org/wiki/view/Asterisk+config+meetme.conf seems to
include some old content.
Which is the simplest way to add MeetMe to a pure SIP 1.6.1 install on a
recent kernel ?
Is dahdi_dummy still required ?
Regards
___
--
I would think that VoIP over VPN is a bad idea as UDP packets need to be
in realtime not corrected by the TCP of the VPN.
That depends very much on the VPN in use.
OpenVPN doesn't suffer from this problem. Although it's SSL-based
(and one might think it does everything through SSL-over-TCP),
On Fri, 8 May 2009, David Backeberg wrote:
On Fri, May 1, 2009 at 11:18 AM, Jimmy Godbout s...@inbox.com wrote:
Well, my question is: do you guys have any tip in different ways to
create a call in Asterisk using AGI + PHP?
You may prefer AMI and the originate command as opposed to making
2009/5/8 Olivier oza-4...@myamail.com
Hello,
Here (http://downloads.digium.com/pub/telephony/codec_g729/README) are
instructions to install G729 software.
(I think I followed instructions step by step but g729 license doesn't seem
to show up).
My question is :
Is the command bellow still
Dave Platt wrote:
OpenVPN doesn't suffer from this problem. Although it's SSL-based
(and one might think it does everything through SSL-over-TCP),
it actually sends the VPN traffic via UDP... it uses TCP only
for the negotiation and administrative aspects of setting up
the VPN connection.
Hello!
It's me again. I began a fairly large modification to my CDR proposal
some weeks ago, and finally yesterday and this morning got enough
accomplished to allow a commit and some peer review.
Check the docs out via svn co
http://svn.digium.com/svn/asterisk/team/murf/RFCs
This is a
http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.confmailcmd
Mailcmd allows the administrator to override the default mailer
command with a defined command. Mailcmd takes a string value set to the
desired command line to execute when a user needs to be notified of a
voice mail message.
On Fri, 8 May 2009, David Backeberg wrote:
On Thu, May 7, 2009 at 11:44 AM, Steve Edwards
asterisk@sedwards.com wrote:
The meetmeadmin() dialplan function lets you specify a user to mute,
un-mute or kick. But how do you get a list of users in your dialplan?
You need a way to keep
quot;, "1?continue") in new stack -- Goto (macro-user-callerid,s,23) -- Executing [...@macro-user-callerid:23] NoOp("SIP/2022-083c53f0", "Using CallerID "EXTs1" 2022") in new stack -- Executing [90012127773...@from-internal:2] Set("SIP/2022-083c53f0&
I'd like to setup a single extension for which all INBOUND and OUTBOUND
calls are recorded to a wav file. I took a look at the wiki:
http://www.voip-info.org/wiki/view/Asterisk+record+calls
but it's not too helpful. Can someone show some sample code in out
recording?
Thanks,
MD
I want to record calls in wav format. Can someone tell me how many MB of
storage per minute each recording requires (assuming SIP / uLaw codec / full
duplex recording)
Thanks,
MD
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
On Fri, May 8, 2009 at 5:39 PM, Michelle Dupuis supp...@ocg.ca wrote:
I want to record calls in wav format. Can someone tell me how many MB of
storage per minute each recording requires (assuming SIP / uLaw codec / full
duplex recording)
Thanks,
MD
1 meg/min is a good rule of thumb.
It's been a few years ago, but Network Computing had tests results showing
that VoIP over a VPN was measurably better than outside a VPN. Why?
Because the latency was low enough that lost UDP packets (within the VPN
tunnel) could be re-transmitted before the jitter buffer had expired. Since
most
David Backeberg wrote:
On Thu, May 7, 2009 at 3:54 PM, Brent Davidson
br...@texascountrytitle.com wrote:
I've got multiple satellite office all linked back to the main office
via VPN. Each office has their own asterisk server which registers back
to the main office's Asterisk server. Each
Jeremy Mann wrote:
Access-list 100 permit ip host asterisk server any
Class-map match-any voip
Match access-group 100
Policy-map voip
Class voip
Priority 256
Class class-default
Fair-queue
Interface fastethernet 0
Service-policy output voip
Above is what I do to prioritize
Hi all,
This is my first post to the list.
I have searched the net far and wide but can't find an answer to this
problem.
When I have call forward working or use the voicemail from a SIP phone,
the first part of the message is always cut off. So instead of hearing
call forward cancelled I
The problem turned out to be that I had a typo in my include = line so I
was not including the extensions that I intended to in extensions.conf.
Unlike chan_sip, chan_dahdi does not print the nice messages on CLI that
says you have dialed an unknown extension. As I was use to these messages
for
Hello!
I tested the last revision of chan_mobile.c but I am still getting errors
during the call. The last time the error starts when the call was
established but with this new revision it starts sooner during the call
attempting.
The following error is reported:
[May 8 19:40:48] ERROR[6971]:
sox original-file.gsm appended-file.gsm pad 1.5
works with wav too.
Dan
-Original Message-
Sent: Friday, May 08, 2009 6:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Possible to add Voice delay?
Hi all,
This is my first post to the list.
George Farris wrote:
When I have call forward working or use the voicemail from a SIP phone,
the first part of the message is always cut off. So instead of hearing
call forward cancelled I hear l forward cancelled.
Or in voicemail I hear edian mail instead of comedian mail.
How can I add
On Fri, May 8, 2009 at 5:31 PM, Michelle Dupuis supp...@ocg.ca wrote:
I'd like to setup a single extension for which all INBOUND and OUTBOUND
calls are recorded to a wav file. I took a look at the wiki:
http://www.voip-info.org/wiki/view/Asterisk+record+calls
but it's not too helpful. Can
On Fri, May 8, 2009 at 4:54 PM, Steve Edwards asterisk@sedwards.com wrote:
On Fri, 8 May 2009, David Backeberg wrote:
You need a way to keep state. I use a database and AGI for that purpose.
I thought about keeping state in the db, but then I'd have to run the AGI
when they leave the
I am interested in setting up asterisk to record all calls it
processes. there are however some legal quirks to doing this that I
have run across such as one party notification vs 2 party notification
requirements which depend upon the physical endpoints of a call. If I
wish to discretely record
On Fri, 8 May 2009, David Backeberg wrote:
On Fri, May 8, 2009 at 4:54 PM, Steve Edwards asterisk@sedwards.com
wrote:
On Fri, 8 May 2009, David Backeberg wrote:
You need a way to keep state. I use a database and AGI for that purpose.
I thought about keeping state in the db, but then
On Fri, 8 May 2009, Eric Fort wrote:
I am interested in setting up asterisk to record all calls it
processes. there are however some legal quirks to doing this that I
have run across such as one party notification vs 2 party notification
requirements which depend upon the physical endpoints
On Fri, 8 May 2009, Michelle Dupuis wrote:
I want to record calls in wav format. Can someone tell me how many MB
of storage per minute each recording requires (assuming SIP / uLaw codec
/ full duplex recording)
Your choice of technology and codec are irrelevant to the size of wav file
On Fri, 2009-05-08 at 18:31 -0700, Trevor Peirce wrote:
George Farris wrote:
When I have call forward working or use the voicemail from a SIP phone,
the first part of the message is always cut off. So instead of hearing
call forward cancelled I hear l forward cancelled.
Or in
On Tue, 2009-04-28 at 00:06 +0200, Hans Witvliet wrote:
Sometime ago i got this status from _the_ guru...
Russell replied, referencing 1.6.2...
(but other code might get in-the-way)
Thanks!
From: Russell Bryant russ...@digium.com
There has been progress. It is not yet merged into the main
Could those on the list who have used or tried to use VoIP over a
satellite internet connection comment on how well it works or if it
even works at all in a reliable way. What is the effect of latency on
the VoIP path and how much is generally tolerable? routing via
satellite adds about a
Voip over satellite can be done if enough bandwidth is reserved properly for
it.
Use the g729 codec and ask for 24 kilobits of upstream cir and you should be
fine.
Also you'll want to mark your packets with the EF tos bit in sip.conf.
If done right the delay isn't too bad. Yes you can tell it is
Hi All,
I am trying to run the asterisk CLI commands from php.
Some thing like asterisk -rx reload.
But it is not working, where as when I try to run
linux ls command it works fine.
below is sample php code which I am trying to run.
$command = sudo asterisk -rx reload;
$value1 =
If people don't mind taking turns talking, it will work. It's just going
to be like talking on a CB. Reminds me of talking to my grandparents in the
Europe as a child in the early 80's.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Sat, May 09, 2009 at 10:17:27AM +0530, Sam Hawkin wrote:
Hi All,
I am trying to run the asterisk CLI commands from php.
Some thing like asterisk -rx reload.
But it is not working, where as when I try to run
linux ls command it works fine.
below is sample php code which I am trying to
I work on the salmon river in Idaho as a computer/radio tech.
All of the satellite isp's do not have the upstream capability.
Skype barely works. (you have to try upwards of 20 times for it to work)
If I have to make phone calls when I am there I always use the SSB
Radiophone or satellite phone
Greetings,I have a question for those who have done a few professional installs of Asterisk. Is it taboo to use something like AsteriskNow/FreePBX/Trixbox to get a base installation of Asterisk installed and functional for a small office? If not then do you always compile from scratch or use
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