Is there a way to check if a peer is registered with the other box and
forward the call there if a call comes in?
Yes, you can (if nothing else, I'm fuzzy this morning) try forwarding
the call and it will fail if the device is not registered because
Asterisk will report it not found with a SIP
Douglas Garstang wrote:
I am trying to call the DUNDILOOKUP dialplan function from ael2, like this:
context route {
Set(PATH=${DUNDILOOKUP(${EXTEN},DUNDIRegistr)});
}
The DUNDILOOKUP function returns no data. However, when I call it exactly the
same way in a regular context, it
Is it just me or am I seeing more AEL2 code in people's examples? Could
it be that AEL2 is starting to finally catch on?
SKM
-AEL2 Fanatic, Potato Eater, and General Lurker
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asterisk-users
Gents,
First, let me apologize for cross-posting and for posting off topic.
Cross post was only to reach members of one list that may not be on the
other.
Those of you that know me, know that I don't post off topic very often,
let alone put out a list wide request for help. however, a client of
Gentlemen,
An update on my prior post. I have not confirmed a solution is in place,
but I do know that a gent has identified the virus, and symantec has
confirmed it's new. I don't know the prefix, but it'll be named after my
coworker.dcollins is the name it'll be under.
I'll update if we
Please _don't_ ! I'm sympathetic to your situation,
and we have all shouted for help in a crisis, but.
This VectorGraphics+Javascript+IE+windows
exploit has _no_ relevance to this group.
The only way that it is related to a discussion of asterisk
source code development would be if
Just wanted to apologize again for the OT post. Also, I'll not be
posting further on this subject. If you want fix information, contact me
offlist and I'll forward any information I'm given by DCollins
--
S McGowan
VoIP Consultant
[EMAIL PROTECTED]
-BEGIN PGP PUBLIC KEY BLOCK-
Version:
calvis wrote:
Have you check out http://www.f-secure.com/weblog/ to see if it is related
your problem? They offer a few solutions.
Charles Alvis
Internet Technology Group, Inc.
Redmond, WA
Personal Blog http://www.spamspotter.com
Thank you for the link! I've forwarded it to the admins
Ryan Burke wrote:
Thanks for the info. So it was really just one server that handled 2.5k
user registrations and up to 500 concurrent calls? Do you remember
anything about the codecs? Was there any transcoding done, music on
hold, queues, etc? Usually for a dual Xeon 3Ghz people say they get
good stuff mate.
a few clarifications:
you had static extensions.conf, realtime sipusers, etc, right?
Also, abt features like call fwding, etc, which one is better,
performance wise, using a mysql db, or use Asterisk's internal
DB(berkeley db, isnt it?using those DBput n DBget
I would like to know how you got Asterisk to function with 2500 SIP
registrations. Did you have qualify enabled?
Yes, qualify was enabled, using the standard length of qualification
period between checks. Very few accounts had custom qualify settings.
What about the 500 simultaneous
S McGowan,
I don't know if you missed my question (from the slew of questions you've
received and answered), but I was wondering about transcoding and PSTN
channels. What kind of codecs were used and was there any transcoding
happening? Was this box only responsible for VoIP-to-VoIP calls
Kristian Kielhofner wrote:
Rushowr wrote:
S McGowan,
I don't know if you missed my question (from the slew of questions
you've
received and answered), but I was wondering about transcoding and PSTN
channels. What kind of codecs were used and was there any transcoding
happening
adebayo omo-dare wrote:
Hi Sheerwood,
I unfortunately saw a bit of what I percieve to be an error in what you
said. BerkeleyDB does in fact support replication across nodes - see:
http://www.sleepycat.com/docs/ref/rep/intro.html - possibly you meant to
say the version implemented in * does
Kristian Kielhofner wrote:
[EMAIL PROTECTED] wrote:
Again, I'm amazed by this example since it
seems to be way over what anyone else normally reports as usable.
Exactly!
--
Kristian Kielhofner
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Natambu Obleton wrote:
Ok. First question is how to make it say my number back.
Like if you call extension 1000 from extension 1001, I want it to say
“Number is 1,0,0,1” like an ANI number? Help.
Also I want to setup a meetme conference so that it asks “Enter
conference
Marco Mouta wrote:
Hi all,
I'm planing to develop a solution based on Asterisk for about 300 users.
My question now is, do I really need to use openSER as the sip proxy and
Asterisk for the PBX functions?
Can i trust in a solution only with Asterisk to make all this install?
Please
Benjamin Jacob wrote:
Rushowr wrote:
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Ryan wrote:
Can you explain your design in a little more detail? What kind of hardware
did you use to get over 1k users on a single box and 500 concurrent calls?
Sounds like a very interesting medium-large scale implementation that
others could learn from.
thanks,
Ryan
I'll do the best
Hall, Eric M. wrote:
I got the config working. Not sure if someone has pre-recorded sounds
for this app or not. Looked all over for them and I'm unable to locate
them.If anyone has sound file they would like to share that would help
me greatly.
Thanks
*Sent:* Friday, September
[EMAIL PROTECTED] wrote:
Can you explain your design in a little more detail? What kind of hardware
did you use to get over 1k users on a single box and 500 concurrent calls?
Sounds like a very interesting medium-large scale implementation that
others could learn from.
thanks,
Ryan
bilal ghayyad wrote:
Hi list;
Does asterisk work with fedora because redhat
enterprise is licensed and costly.
Regards
Bilal
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http://mail.yahoo.com
Rich Adamson wrote:
Rushowr wrote:
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Christopher Corn wrote:
thanks for the reply. why are residential lines cheaper than businesses?
say for unlimited, it always costs more for residential.
*/Michael Graves [EMAIL PROTECTED]/* wrote:
I'd
Benjamin Jacob wrote:
Hello ppl,
Wanted to know your experiences, if you've worked with Asterisk Realtime
Architecture.
Which one do you prefer, static or realtime?
I personaly think, the static architecture is a better solution, cuz, in
the realtime config, to check the dialplan(n hence
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Benjamin Jacob wrote:
Rushowr wrote:
Benjamin Jacob wrote:
Hello ppl,
Wanted to know your experiences, if you've worked with Asterisk Realtime
Architecture.
Which one do you prefer, static or realtime?
I personaly think, the static
Hugo wrote:
Anyone could help to use Static RealTime with SIP.CONF. I use Dynamic
Realtime successfully. In fact, I want to know how to compos the correct
DB(postgres or mysql) fields (I think STATIC configuration is different
from DYNAMIC).
Regards,
Hugo
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Dijkstra, Roelof wrote:
Hello,
We currenty have an asterisk cluster running, with a quad PRI and a quad BRI.
This all works pretty well.
What i was wondering:
If i do a
show sip peers
I see all the ip addresses of the phones that
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Christopher Corn wrote:
thanks for the reply. why are residential lines cheaper than businesses?
say for unlimited, it always costs more for residential.
*/Michael Graves [EMAIL PROTECTED]/* wrote:
I'd just use a service that's being
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RR wrote:
I am currently running this with UnixODBC - FreeTDS - MSSQL Server
2K ( please don't hate me for using an 'evil empire' product amongst
the pure sanctity of open source :D). But the results are, well...So
far so good. But I can't say
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Mike wrote:
Thanks Tim.
I've been trying to find out what's happening. Basically, somehow, it seems
that my Polycom 501 knows what extensions are valid and which aren't in my
dialplan. Obviously, the 501 doesn't really know that, but Asterisk
]
Content-Length: 0
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rushowr
Sent: September 8, 2006 4:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] What don't I get about SIP?
Mike wrote:
Thanks Tim
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Mike wrote:
But that's the whole freaking problem!!!
If I could do that, I would. But Asterisk keeps on sending the 484 Address
incomplete message, and the Polycom keeps on waiting silently and patiently
for me to put in the needed extra
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Hall, Eric M. wrote:
Hello group
I have a customer that has asked me to build an auto dialer that will
call customer a few day before an appt and remind them of the time and
date of the appt.
Does anyone have any good links for apps that
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Joe Shmoe wrote:
You say its not your code. But yet, why would you
actually admit to one of your own leaking it. Well
some research has been done one the code.. here's what
we found..
the g723.1 library code that was posted matches the
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Hey all,
I'm looking into setting up a system or two with either IMAP or ODBC
storage of Voicemail messages and wanted to hear about your experiences,
gather tips or warnings, etc, before I go diving too deep into it. Are
either of those storage
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Steve Hsieh wrote:
Greetings,
Is it possible to create a conditional IF inside extensions.conf based
on the source IP address of a SIP phone (as opposed to extension)? What
I would like to do is the following:
1. If SIP phone IP belongs
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Rushowr wrote:
Steve Hsieh wrote:
Greetings,
Is it possible to create a conditional IF inside extensions.conf based
on the source IP address of a SIP phone (as opposed to extension)? What
I would like to do is the following:
1. If SIP
/06, *Rushowr* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rushowr wrote:
Steve Hsieh wrote:
Greetings,
Is it possible to create a conditional IF inside extensions.conf
based
on the source IP
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Benjamin Jacob
Sent: Monday, September 04, 2006 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] includes in realtime ??
Hello ppl,
Is it possible to include
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vidura
Senadeera
Sent: Monday, September 04, 2006 5:15 AM
To: asterisk-users@lists.digium.com
Cc: asterisk-dev@lists.digium.com
Subject: [asterisk-users]
You need to install libmysqlclient15dev, it's saying it can't find the
header files it requires.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Christopher Aloi
Sent: Friday, August 25, 2006 8:36 PM
To: Asterisk Users Mailing List - Non-Commercial
Try changing the DTMF mode for that line, I've found that if rfc2833 doesn't
work, inband will
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lenny
Sent: Saturday, September 02, 2006 4:28 AM
To: Asterisk-Users@lists.digium.com
Subject:
In short, yes...
The wiki (http://www.voip-info.org) has documentation
on how to configure your servers, how to configure the dialplan, etcI don't
mean to single you out mate, but has anyone else noticed an increase in the
number of questions being asked that could have been answered
That's very very odd...that should work fine :(
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Larry Alkoff
Sent: Tuesday, August 29, 2006 11:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
Then entire OLD extension must be removed so the new one will match
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Larry Alkoff
Sent: Tuesday, August 29, 2006 6:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rushowr
Sent: Monday, 28 August 2006 2:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Asterisk with PABX
Too true too true Personally, I think trying to use
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To a single extension?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brandon
Galbraith
Sent: Sunday, August 27, 2006 8:16 PM
To: Asterisk Users Mailing List - Non-Commercial
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Actually, isn't there SLA work being done in the trunk right now?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: Monday, August 28, 2006 9:16 AM
To: Asterisk Users Mailing List -
IIRC, you'll want to look at 'hint' extensions, and
possibly subscriptions to get status updates
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
MirSent: Monday, August 28, 2006 9:34 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
(geographically seperated)
register to a single extension, so when the extension is dialed, any phone can
pick up the call. Is this better handled by having each phone have a seperate
extension, and handle the call routing in a dial plan?
-brandon
On 8/28/06, Rushowr
[EMAIL PROTECTED]
wrote
You'll want to put them in the _additional.conf files,
because AAH/TB/FPBX doesn't always play nice with changes to the configuration
files that it modifies directly.
Rushowr / SKM
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Curt
ShafferSent: Monday, August
Too true too true Personally, I think trying to use Trixbox to learn
Asterisk is akin to a monkey humpin' a footballIt's just not right.
Anywhohad to do my smartass deed for the day
Rushowr
(Hates getting contracts to fix someone's AAH/TrixBox/FreePBX phone
system
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First big question is are you checking beforehand how long the limit should
be by calculating ((BALANCE / RATE) / 1000)
If you're not, that would be why it doesn't disconnect the customer within a
time period that wouldn't result in a negative
from within asterisk, just run the following
command:
show application Verbose
That'll fill you in. Your other solid option is to search
the wiki
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
AbdulSent: Sunday, August 27, 2006 4:05 AMTo:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Personally I've used the shared database method previously, I've even setup
a mysql cluster and had each asterisk host be a query node.
SKM
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Set(TIMEOUT(absolute)=seconds)
Change seconds to the number of seconds you want to allow a
call to last
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
AbdulSent: Sunday, August 27, 2006 1:21 AMTo:
Asterisk-Users@lists.digium.comSubject: [asterisk-users] Call
I think he actually needs show channels verbose
*CLI help show channels
Usage: show channels [concise|verbose]
Lists currently defined channels and some information about them. If
'concise' is specified, the format is abridged and in a more easily
machine parsable format. If
I now need to remove the 9 but then prefix another number onto
the phone number before dialing now but am unsure how to do
this is the dialplan.
Simple...for instance, if you wish to prefix 123 before the number just do:
Dial(SIP/123${EXTEN}
Would someone be able to point me in the right
I believe you want to use ${ENV(variable)}.. From asterisk's CLI:
*CLIshow function ENV
-= Info about function 'ENV' =-
[Syntax]
ENV(envname)
[Synopsis]
Gets or sets the environment variable specified
Note that ENV is a function...you need to encase the argument inside
parentheses
Download the asterisk-addons package. It contains several addons, including
all the mysql additions.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Diego Quintana Cruz
Sent: Thursday, August 24, 2006 4:06 PM
To: asterisk-users@lists.digium.com
I wish I could offer some direct help on whether or not your method with a
comma separated list would work, but I can't. However, you could always
create a few entries using different formats and then run some tests against
them
-Original Message-
From: [EMAIL PROTECTED]
Not last I heard...I just fought with this
yesterday
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
LunsfordSent: Tuesday, August 22, 2006 8:10 PMTo:
asterisk-users@lists.digium.comSubject: [asterisk-users] Setting
the contact header on outbound
: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Strange SIP response
Rushowr wrote:
Have you run SIP DEBUG PEER 192.168.1.60? It may
help...tcpdump is also
one of my personal favorites
Yes, I have used it. The lines are extracted from a sip debug
Do you have the development libraries installed too? I
believe on Debian it's something like libmysqlclient
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Christopher AloiSent: Friday, August 25, 2006 8:36
PMTo: Asterisk Users Mailing List - Non-Commercial
on logout?
On Wed, Aug 23, 2006 at 11:03:23PM -0700, Steve Edwards wrote:
On Thu, 24 Aug 2006, Jeremy McNamara wrote:
Rushowr wrote:
Hey all, I have an interesting issue that just recently
started when
I grabbed a copy of the trunk about a week ago and
compiled it. Ever
since
Just gotta check, I've never seen a complete day with no posts
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Hey all, I have an interesting issue that just recently started when I
grabbed a copy of the trunk about a week ago and compiled it. Ever since
that compile, if I start Asterisk (disconnected terminal, using
safe_asterisk to launch) and then continue on about my work with it, when I
disconnect my
Have you run SIP DEBUG PEER 192.168.1.60? It may help...tcpdump is also one
of my personal favorites
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Diego Andres Asenjo G.
Sent: Tuesday, August 22, 2006 6:50 PM
To: asterisk-users@lists.digium.com
id passed to it as part of the call?
W
Rushowr wrote:
${CALLERID(number)}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Warren (mailing lists)
Sent: Monday, August 21, 2006 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
You can't use that realtime field in an include statement... However, you
could use context names like caller-conference and caller-longdistance and
then call the context dynamically with Goto(caller-${key}).
Otherwise, you're going to have to do it with logic routing. May I suggest
at LEAST
Gotoif($[${ISNULL(${CALLERID(number)})} = 1]?ask4cardnum:doagi_astcc)
:-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ronald Wiplinger
Sent: Tuesday, August 22, 2006 7:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hello all,
Just had a question that I've not been able to find a suitable answer for.
When we receive calls on SIP, we can get SIP_HEADER(Remote-Party-ID) and
check the privacy flag for what privacy is requested. Now, since SIP_HEADER
is not writable, how can I set the privacy flag in the RPID
Of Rushowr
Sent: Tuesday, August 22, 2006 8:55 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Setting RPID privacy?
Hello all,
Just had a question that I've not been able to find a suitable
answer for.
When we receive calls on SIP, we can get
${CALLERID(number)}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Warren (mailing lists)
Sent: Monday, August 21, 2006 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Variable to show caller id for a
returned anything.
How can I do this? Alternately... Is there a way to have a
program fired off when an extension rings that will have the
caller id passed to it as part of the call?
W
Rushowr wrote:
${CALLERID(number)}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hey all,
I've done some peeking around and can't find a GOOD listing of what the
currently supported SIP headers are that Asterisk supports. My main reason
is to get the CallerID/RPID settings for whether or not to display, but
there's others as well.
Anyone have a link?
SKM
*steps slowly to the soapbox*
Can we please get this pissing match over with? The horse is dead, stop
beating it and bury the corpse for chrissake
*steps down from soapbox*
That's all I got
*checks the fire extinguisher and awaits the flames to be redirected*
SKM
-Original
Oh my gawdwhy are my emails taking so long to publish?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rushowr
Sent: Thursday, August 17, 2006 9:43 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users
-- there's plenty to do!
murf
Please, don't! Even if it last only a few versions, it will be
worth it!
BarZ
Murf, I think you know where I stand on this ;-)
Rushowr
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asterisk-users
IIRC, You can use REGEXes in your extension matchingDon't have a handy
link, but if I find it, I'll forward
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
William Moore
Sent: Friday, August 18, 2006 1:04 PM
To: Asterisk Users Mailing List -
Sounds like a sessions error
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sebastian
Milioto
Sent: Thursday, August 17, 2006 2:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] astbill white screen!!
Hi all,
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, August 17, 2006 4:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Asterisk 'Hosting'
-Original Message-
From:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Thursday, August 17, 2006 4:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Asterisk 'Hosting'
Hi. I only just stumled across it myself.
Instead of SYSTEM(), you could use an AGI possibly.
Cheers,
SKM
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damien
Gabrielson
Sent: Thursday, August 17, 2006 6:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Realtime configuration is when you tell Asterisk to use the database for
reading the sip global configuration items.
Static configuration is when you use the sip.conf file to store the sip
global configuration items.
You cannot mix the two. That's all.
-Original Message-
From:
I use Asterisk Realtime a LOT, it's pretty much the core of all my
consulting jobs in the last year. If you still need help, I'll try to assist
you as much as possible.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Wednesday,
What's the Dial command being used to pass the call to the
Softphones?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy
BoySent: Wednesday, August 16, 2006 3:23 AMTo:
[EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
DiscussionSubject:
presentationexten = s,n,SetMusicOnHold(default)exten =
s,n,Set(TIMEOUT(digit)=5)exten = s,n,Set(TIMEOUT(response)=10)exten
= s,n,Background(/tmp/virg2)exten = s,n,Goto(s,1)exten =
s,n,Hangup()include = leader
Hope this is helpful in
some way...
Rushowr
From: [EMAIL PROTECTED]
[mailto
You CAN use both. You cannot use both if you tell asterisk to get the WHOLE
sip configuration file from the database. But, in your case, realtime peers
and users
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, August 16,
Just in case Murf doesn't get around to answering this one, I'll stab it...
For one thing, I can code in a style that is similar to many programming
languages, which can reduce the learning curve for many people, and
personally I think it makes the code MORE readable because If statements
follow
I have to say that I'm experiencing the same issues, using the latest SVN
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chan Kwang
Mien
Sent: Monday, August 14, 2006 8:26 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problems with
Discussion
Subject: RE: [asterisk-users] Problems with Hangup
- Original Message -
From: Rushowr
[mailto:[EMAIL PROTECTED]
To: [EMAIL PROTECTED], 'Asterisk Users Mailing List - Non-Commercial
Discussion'
[mailto:[EMAIL PROTECTED]
Sent: Mon, 14 Aug 2006 09:28:29
-0300
Subject: RE: [asterisk-users
Hey Attilla, thanks for the update. I'm also working on a solution, but
unfortunately the system I'm working with needs the separate macros. I'll
update the list if anything gets worked out.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Attilla De
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Macro inside macro
Any reason that you can't set variables before you use Gosub, then access
them in the subroutine?
Attilla De Groot wrote:
On Aug 14, 2006, at 6:39 PM, Eric ManxPower Wieling wrote:
Rushowr wrote:
Hey
AGI+PHP would be a good
place to do all of this. However, be aware that interpreted code such as PHP
incurs a performance hit and may not be suitable for very large installations,
in addition to the issue of passing call control away from Asterisk in general.
(ref: "Asterisk Performance",
It's because the standard CDR engine uses the last ${EXTEN} value as the
destination number
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthias
Fechner
Sent: Friday, August 11, 2006 6:08 AM
To: asterisk-users@lists.digium.com
Subject:
The reason he might want it is because it's a feature offered by many POTS
and Mobile Telcos. I know that's why I've played with it, the ITSPs and SIP
Termination providers I consult for want to have as many if not more
features to offer than the POTS and Mobile guys.
Cheers,
Rushowr - Sherwood
are lacking
John Novack
Rushowr wrote:
The reason he might want it is because it's a feature offered by many POTS
and Mobile Telcos. I know that's why I've played with it, the ITSPs and SIP
Termination providers I consult for want to have as many if not more
features to offer than the POTS
Uh, what's your Register statement for those SIP DIDs look like? If you
don't specify the number after a /, you'll be handed calls for that line,
but specifying 's' as the extension.
register = user[:secret[:[EMAIL PROTECTED]:port][/extension]
I consider that last argument required anymore
username + secret
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]Sent: Thursday, August 10, 2006 7:53
AMTo: asterisk-users@lists.digium.comSubject:
[asterisk-users] Realtime SIP Authentication
Hi All,I'm using Realtime for SIP users
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