On Thu, Aug 19, 2010 at 5:56 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Thu, 19 Aug 2010, Carlos Chavez wrote:
I am making a web interface so users can manage their voicemail.
The only problem I have is that since the Web server and Asterisk run as
different users I need to
On Thu, Aug 19, 2010 at 9:44 PM, Tim Nelson tnel...@rockbochs.com wrote:
- Zhang Shukun bit...@gmail.com wrote:
== Using SIP RTP CoS mark 5
[Aug 20 18:23:21] NOTICE[14543]: chan_sip.c:8454 process_sdp: No
compatible codecs, not accepting this offer!
Could you tell me what 's wrong?
On Thu, Aug 19, 2010 at 7:46 PM, Nasir Iqbal na...@ictinnovations.com wrote:
Hi,
there's still no conceivable reason
What can be? except performance! (as asterisk has to create one additional
leg and bridge it) Which is very conceivable to those who are dealing with
high load traffic.
And
...@gmail.com wrote:
This is what I ended up doing, working fine now. Cheers
On Thu, 19 Aug 2010, Sherwood McGowan wrote:
LOL, I hate to say this but writing an AGI script just adds yet another
application layer to your total solution.
Yes, another layer, but a layer where you will have full access
On Thu, Aug 19, 2010 at 11:01 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Thu, 19 Aug 2010, Tino wrote:
But when i call my DID number following dialplans are being executed.
What i need is to set a variable with one value for one DID number and set
the same variable with another
I'm going to go ahead and say that while I'm not one of the
developers, I think it's safe to say that you cannot record to a file
and play it back at the same time. Probably something like file
locking (for the record, locks it from access by other processes,
etc)...
On Thu, Jul 29, 2010 at
On Thu, Jul 29, 2010 at 10:37 AM, Danny Nicholas da...@debsinc.com wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nasir Javaid
Subject: [asterisk-users] How to extract channel-id of a user or peer
my question is how can i get
On 7/29/2010 6:51 PM, Travis Langhals wrote:
Thanks Sherwood for all the info.
The devices are using ulaw and rfc2833. There is no transcoding on my
server, but not sure what my trunk providers are doing.
I was thinking about the frequency detection issue as it seems to be
primarily
+sip+lockup
Cheers,
Sherwood McGowan
...I've been working with VoIP for almost 10 years now!?!?!AUUGH!
On Wed, Jul 28, 2010 at 4:54 PM, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
We are running asteriskNow 1.4.18 and after a few days it becomes
unresponsive
that is speaking at the
time of the DTMF event on your various captures will have a frequency
range in common...a very close range...maybe look up DTMF tone
definition and get the freqs(did itmore detail than even I
feel like doing right now :D)
Cheers,
Sherwood McGowan
On Wed, Jul 28, 2010 at 6:43
There's an app_record, and I believe app_dictate
On 7/27/2010 7:39 PM, Michelle Dupuis wrote:
Is there a prebuild module/dialplan which gives me a nice interface to
recording messages? Assuming I can't use the voicemail command, I need to
offer users a way to record, playback, erase,
Record does not continue until the end of the call, it records until the
# is pressed or the max duration is reached:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Record
Enjoy
On 7/27/2010 9:00 PM, Michelle Dupuis wrote:
That's along the lines of what I was thinking, but how do you trap the
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Please send the relevant portion of your extensions.conf, as that is
where the problem is
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Yes, I know I could accomplish this using the dialplan and MySQL
--
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...then open the pcap file in Wireshark and look in the
SIP header for the callerid information
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The Linksys is taking *81 as a local spree code, causing it to be
stripped. The problem you're then getting is probably that Asterisk is
using _it's_ caller id information for your peer
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Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL
to add a pattern that matches your extension's setup (3 digit
extensions get _XXX, etc...) and then call a macro or subroutine that
will perform the standard extension dialing:
EXAMPLE:
exten = _XXX,1,Macro(std-exten,${EXTEN})
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sounds like the sample rate is wrong somehow
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VoIP / Telecom Solutions
[EMAIL PROTECTED
this practice illegal:
http://www.engadget.com/2007/06/29/congress-looking-to-make-caller-id-spoofing-illegal/
http://www.govtrack.us/congress/bill.xpd?tab=summarybill=s110-704
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Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED
Sherwood McGowan wrote:
Gentlemen,
I'm using 1.4.21 SVN Tag, and have the queues set up to use Realtime.
This system works fine with 1.2.28, and everything loads fine with no
errors, but when I log an agent in I see the extra message (not in
use) by their listing and they are not rang
anything I find. Unfortunately, I've not
seen anyone else with this issue, as I've googled like crazy
--
Sherwood McGowan
VoIP / Telecom Solutions
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Mark Michelson wrote:
Sherwood McGowan wrote:
Gentlemen,
I'm using 1.4.21 SVN Tag, and have the queues set up to use Realtime.
This system works fine with 1.2.28, and everything loads fine with no
errors, but when I log an agent in I see the extra message (not in
use) by their listing
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Quick note, almost every cell phone carrier has an email to SMS gateway
that you can use to send their customers SMS via email
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VoIP / Telecom Solutions
is called.
Any ideas?
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call a shell
script that transcodes the file before offering it up.
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. :-)
LOL, good question...I wouldn't mind havin' one...i can haz tweaker? ROFL
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post-call by speexenc. We also run
PostgreSQL and Apache on the same system to serve CDRs with links to
recordings.
Anything else you'd like to know?
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VoIP / Telecom Solutions
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Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: Tuesday, June 17, 2008 5:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Center
broadband Voice wrote:
Is anyone using Asterisk
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: Tuesday, June 17, 2008 5:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Center
broadband Voice wrote:
Is anyone using Asterisk
on the entire new CDR/CEL branch :)
ROFLno seriouslyI want one ;-)
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extension, use monitor + soxmix to mix the
recordings) will work just fine, I use it on a medium-large installation
that does about 10K calls a day, with no issues in regards to recordings
or ability to access calls/recordings.
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Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED
:)
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, as the code that you
had at fail is taken care of within the if statement's execution.
Let me know if there's any issue, if there is it's probably in the
implementation of the conditional
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Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED
Sherwood McGowan wrote:
snip
I'll be more than glad to help :)
Here's the code:
context default {
_X. = {
Set(DID=${EXTEN:6});
continue:
Noop(${DID});
Set(GROUP(IAX)=incoming);
if(${MATH(${GROUP_COUNT([EMAIL PROTECTED])}+${GROUP_COUNT([EMAIL
PROTECTED
to do database cleanup.. i'd appreciate if you'd shed more light
just in case am missing something..
Rgds
On Fri, Jun 13, 2008 at 3:05 PM, Sherwood McGowan
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Robor Oghene wrote:
hello All,
How do I start and run an agi
Robor Oghene wrote:
Thanks Sherwood, I haven't tried what the line you sent but I think it
would solve my problem I just hope it would run from asterisk
realtime...
On Fri, Jun 13, 2008 at 3:05 PM, Sherwood McGowan
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Robor Oghene
Robor Oghene wrote:
Thanks Sherwood, I haven't tried what the line you sent but I think it
would solve my problem I just hope it would run from asterisk
realtime...
On Fri, Jun 13, 2008 at 3:05 PM, Sherwood McGowan
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Robor Oghene
Fork The CDR into 2 separate entities.
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executions
(if AGI is used, yes you need DeadAGI).
Just thought I'd clear it up
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Atis Lezdins wrote:
On Thu, Jun 12, 2008 at 9:14 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote:
Hi all,
I have setup an asterisk system which:
recieves incoming sip calls
ask
(call_to ${exten_copy} ${call_to});
}
}
Ah I love to see AEL in a suggestion post :)
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Sherwood McGowan wrote:
Gentlemen,
I have a particularly strange problem, just started happening. One of
my clients is running Asterisk 1.2.28 and has mysql realtime queues.
We log in a member, and then place a test call to the 0 queue but
since joinempty is set to no, and Asterisk thinks
Jared Smith wrote:
On Mon, 2008-06-09 at 00:26 -0500, Sherwood McGowan wrote:
Members:
9001 (Invalid) has taken no calls yet
It appears that there are no valid members of the queue, which at first
glance would seem to me to be your problem.
Thank you to all who
.
-- Playing 'ivr/new/nwi-welcome' (language 'en')
-- Hungup 'IAX2/carp-1957'
== Spawn extension (pri-inbound, s, 3) exited non-zero on
'SIP/pri-006ba540'
-- User disconnected
--
Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED
/listinfo/asterisk-users
He's right, you should get congestion in less than a second (unless your
provider is slow anyway in which case you should switch providers anyway).
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VoIP / Telecom Solutions
[EMAIL PROTECTED]
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indications.conf is the file you want to edit :) It defines what
ringtones and other indication signals to use.
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VoIP / Telecom Solutions
[EMAIL PROTECTED]
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. The actual tones
; used by BT include some volume differences so sound slightly different
; from Asterisk-generated ones.
dial = 350+440
Any idea why?
Thanks
Adrian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: 05
and
causing havoc.
From: Sherwood McGowan [EMAIL PROTECTED]
Date: Thu, May 22, 2008 7:13 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
snip
True True...it's only been a minor annoyance for me, but in the interest
of improving Asterisk I
options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
I've wondered the same thing, but it hasn't caused me any issues. I've
searched google quite a few times, haven't found anything useful.
Sherwood McGowan
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Sherwood,
I've done the backtrace. Maybe you can submit yours too.
http://bugs.digium.com/view.php?id=12709
Thanks,
Mark.
Original Message
Subject: Re: [asterisk-users] reload stopping EVERYTHING on CLI and
causing havoc.
From: Sherwood McGowan [EMAIL PROTECTED]
Date: Thu
Adrian Marsh wrote:
Hi All,
In my old telco days (SS7), if I was wanting to hand back a call to
the network for transfer to a different PSTN number, there was a
specific SS7 action I could take, which send the call back to the
network, which in turn then routed the call appropriately. It
Brian J. Murrell wrote:
On Thu, 2008-05-22 at 14:05 +0200, nik600 wrote:
No, i'm just wondering because there is creating a greater difference
between my installation and the actual Asterisk.
If it ain't broke, don't fix it. You are already so far behind that any
upgrade is going
Thought I'd post this, see if anyone has experienced this...I am using
Asterisk 1.6 from the SVN branch and Asterisk-addons 1.6 from the SVN
branch. Mysql is running and I've connected using the information that
is in res_mysql.conf but when I try to check the realtime status I get
the
.
Hope this helps.
JR
Thanks, the change to 127.0.0.1 did the trick.
Sherwood McGowan
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continued to allow calls to take place.
I've found that sometimes exiting and reconnecting to the CLI helps, but
there have been a couple occasions where NOTHING would allow the server
to restart save for a reboot. Even killall asterisk didn't kill the
process
Sherwood McGowan
James Sneeringer wrote:
On Thu, May 15, 2008 at 10:47 AM, James Sneeringer [EMAIL PROTECTED] wrote:
When I try to use ChanSpy, the following message is sent repeatedly to
the console (wrapped for readability):
WARNING[32125]: chan_sip.c:3709 sip_write: Asked to
transmit frame type
Steve Totaro wrote:
On Thu, May 22, 2008 at 2:02 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Mark Hamilton wrote:
Hi,
Yesterday I made a change in queues.conf and so tried doing a reload
app_queue.so in the CLI. (Using 1.4.18). It didn't seem to do
anything, infact all action
Steve Totaro wrote:
On Thu, May 22, 2008 at 2:02 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Mark Hamilton wrote:
Hi,
Yesterday I made a change in queues.conf and so tried doing a reload
app_queue.so in the CLI. (Using 1.4.18). It didn't seem to do
anything, infact all action
Steve Totaro wrote:
On Thu, May 22, 2008 at 3:56 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Steve Totaro wrote:
On Thu, May 22, 2008 at 2:02 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Mark Hamilton wrote:
Hi,
Yesterday I made a change in queues.conf
: [asterisk-users] reload stopping EVERYTHING on CLI and
causing
havoc.
On Thu, May 22, 2008 at 2:02 PM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Mark Hamilton wrote:
Hi,
Yesterday I made a change in queues.conf and so tried doing a reload
Pezhman Lali wrote:
Dear,
after a lot of searching and testing I can not find a
total solution for nat, with ser -- asterisk.
now I have 3 selections:
1)using iax-phones instead of sip phones with asterisk
2)using sip phones registered in asterisk,
3)using sip phones with ser/openser and,
Don Pobanz wrote:
Joe Pukepail wrote:
Is there a way to see error counts on the T1 of a PRI?
Hooked up to asterisk via a digium TE122. Looking for
something to make sure I'm not getting any CRC, framing or
other errors on the T1.
Many moons ago I use to used a
that if you're considering using the cdr_mysql addon, I would highly
suggest it as I've used it with MUCH success on high load servers.
Sherwood McGowan
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Douglas Garstang wrote:
I personally can tell you I've never had a problem with either the
PostgreSQL or MySQL cdr apps themselves losing records. However, I can't
say personally how well the ODBC method works. I'll just stick to saying
that if you're considering using the cdr_mysql addon, I
6 months since I worked with a low load server.
Sherwood McGowan
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Steve Prior wrote:
Tilghman Lesher wrote:
Correct; it's actually a workaround for a bug in the MySQL drivers. It was
discovered long after 1.2 was end-of-lifed.
I got bit by MySQL reconnects on some other software I wrote I think when I
jumped from MySQL 4.* to 5.*. If memory
for the input side, I always get them confused
Just be sure not to use the m option, that would mix the two channels
together into a single sound file.
Hope this helps,
Sherwood McGowan
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Steve Totaro wrote:
On Tue, May 20, 2008 at 7:42 AM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Arjan Kroon | Mobillion wrote:
Hi,
Is it possible top use a form of Karaoke Functionality?
When a caller calls a number, he hears a voicefile.
During this voicefile he sings along
Lee, John (Sydney) wrote:
As a result, I just go back to put all users in [default] in
voicemail.conf.
Am I missing anything?
What do those contexts mean in your setup (beside being arbitrary
groups)?
I just want to group the mailboxes by say department rather than putting
Thank you all for your input. Currently, nothing has improved the
dropped call rate by more than .2%, leaving me at 1.8% dropped calls
still..Luckily, our switch back to PRI is due anytime in the next day or
so..
Sherwood McGowan
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Mike wrote:
I am having trouble with Polycom forwards and Asterisk. Basically, I
have no clue on how to force callerid or even custom variables (set
using SetVar in the sip.conf file) on the transfered call.
For example, I set a variable called var_a to foo. When the call
comes in, the
Welcome!
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Alexander Olekhnovich wrote:
Thanks very much for your examples
On Fri, May 16, 2008 at 8:59 PM, Sherwood McGowan
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Alexander Olekhnovich wrote:
Hi Asterisk Users,
I'm interested in how many concurrent calls Asterisk can
Brent Davidson wrote:
Another solution that works for me is to add Playback(silence/1)
just before whatever you are about to do. Something about the
playback command opens the channel up.
-Brent
Sherwood McGowan wrote:
Alan Lord wrote:
Sherwood McGowan wrote:
snip
Joseph L. Casale wrote:
Hi,
What is the method (preferred) way Asterisk handles the incoming
sip lines? I am currently trying to setup two lines, one has
unlimited in/out channels and the other phone number has only two.
The provider has given a macro that manages dialing out on the two
Joseph L. Casale wrote:
Yes, in your dialplan you should have one extension set up for the first
number and where to send it, and a second for the other.
So, if the sip.conf config sends the did into the [incoming] context
and its phone number is 555-1212, would this be the right way:
Alan Lord wrote:
Sherwood McGowan wrote:
snip /
Hrm...I have encountered this before and sometimes doing an explicit
Answer() then a Wait(2), then calling the service can help.
Hope this is helpful
Sherwood McGowan
Bingo!
Thanks a bunch. That sorted it.
Al
Eric Wieling wrote:
Make SURE you are not using callprogress=yes or busydetect=yes (they
default to no). These options are commonly known in the Asterisk world
as randomlydisconnectmycalls=yes.
Sherwood McGowan wrote:
Steve Totaro wrote:
On Thu, May 15, 2008 at 12:59 PM, Don
an explicit
Answer() then a Wait(2), then calling the service can help.
Hope this is helpful
Sherwood McGowan
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David Backeberg wrote:
Has anybody ever tried to roll their own VoIP or Zaptel load
simulator? How did they do it?
SIPP can help with benchmarking SIP calls and you can loop back T1
calls if you have two machines with T1 cards or even one machine with
multiple T1 ports.
SIPp
better)
I hope these two examples help show you how two similar machines can
vary drastically in performance with similar hardware. Differences in
implementation make a BIG difference.
Slainte,
Sherwood McGowan
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with dropped
calls (being worked on with LD provider) and occasional static, due to
using EM Wink signalling (provider screwed up and did not provision PRI)..
Stay with PRI T's and you should be fine.
Sherwood McGowan
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Matt Watson wrote:
You'd probably want to run something else to handle your registrations like
OpenSER with that many phones.
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-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bhrugu Mehta
Sent: Thursday, May 15, 2008 8:31 AM
To: Asterisk
as well, from
file, cannot find ANYTHING...
Thanks for any help,
Sherwood McGowan
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that would seem to cause the
problem after it rained, and the other case was bad carrier equipment
at their shelf, once they moved it to another port on another shelf
the problem disappeared.
Good luck,
MATT---
On 5/15/08, Sherwood McGowan [EMAIL PROTECTED] wrote:
Alright guys and gals
that would seem to cause the
problem after it rained, and the other case was bad carrier equipment
at their shelf, once they moved it to another port on another shelf
the problem disappeared.
Good luck,
MATT---
On 5/15/08, Sherwood McGowan [EMAIL PROTECTED] wrote:
Alright guys and gals
Steve Totaro wrote:
On Thu, May 15, 2008 at 12:59 PM, Don Pobanz
[EMAIL PROTECTED] wrote:
On Thursday, May 15, 2008 11:11 AM - Sherwood McGowan said
...
we've been temporarily stuck with a pair of EM Wink T's. Ever since
then, we've been dropping 1-2% of all calls (in or out
Wai Wu wrote:
Hi all,
What is maximum number of three party conferences can a quadcore 3GHz
system can handle? All the parties a setup with G.711 codec.
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Sherwood McGowan wrote:
Just a quick question, wanted to see if anyone knew where the
menuselect app stored your choices.
I think it's menuselect.makeopts but I'm not sure...just thought
someone might know.
Sherwood McGowan
P.S. I'll post here if I figure it out before there's
Just a quick question, wanted to see if anyone knew where the menuselect
app stored your choices.
I think it's menuselect.makeopts but I'm not sure...just thought someone
might know.
Sherwood McGowan
P.S. I'll post here if I figure it out before there's a response
solution for integration in these sort of specialized systems. I know
they've saved me many headaches.
On Mon, May 12, 2008 at 11:29 AM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Gentlemen,
First let me say it's great to be back on the Asterisk mailing lists.
Those of you who have been
out there
have a 405 out there that they have installed in a 3U?
Thanks in advance for any help that can be offered,
Sherwood McGowan
VoIP / Telecom Solutions Consultant
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk
with the manufacturer of the chassis.
--
Matt
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: Monday, May 12, 2008 11:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] 3U server chassis
David Van Ginneken wrote:
Al Baker wrote:
I would love to be able to issues the necessary Mysql commands to have
true TRANSACTIONS
Such as - Begin Transaction
Select @var=agent.id, agent.exstension where
agent.status='free'
Update agent.status='BUSY'
Also, there's a WinAmp plugin to handle GSM files
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-BJ Weschke
-Sent: Friday, November 04, 2005 4:08 PM
-To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
-Non-Commercial Discussion
-Subject: Re:
We'd really need your extensions.conf to troubleshoot, as well as the error
messag you get when you attempt to dial
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Wilson Pickett
-Sent: Thursday, November 03, 2005 1:51 PM
-To: Asterisk Users
I agree, I would definitely love to find out more about a lot of the
features, new apps (MixMonitor?), etc... I did full text searches against
the tree and couldn't find a single reference to mixmonitor...
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On
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