> 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
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
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mail
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
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
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: G
> 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
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 hav
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
m
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 ge
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
> ___
> --Bandwidth and Colocation prov
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 * doe
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
>>> ch
> 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 ca
> 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 simultaneo
> 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
[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
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, Sep
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
Benjamin Jacob wrote:
> Rushowr wrote:
>
>> ___
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.
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?
>
> Pl
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
> con
bilal ghayyad wrote:
> Hi list;
>
> Does asterisk work with fedora because redhat
> enterprise is licensed and costly.
>
> Regards
> Bilal
>
> __
> Do You Yahoo!?
> Tired of spam? Yahoo! Mail has the best spam protection around
> http://mail.yaho
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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Benjamin Jacob wrote:
> Rushowr wrote:
>
>> Benjamin Jacob wrote:
>>
>>
>>> Hello ppl,
>>> Wanted to know your experiences, if you've worked with Asterisk Realtime
>>> Architecture.
&g
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 h
Rich Adamson wrote:
> Rushowr wrote:
>> -BEGIN PGP SIGNED MESSAGE-
>> Hash: SHA1
>>
>> Christopher Corn wrote:
>>> thanks for the reply. why are residential lines cheaper than businesses?
>>> say for unlimited, it always costs more for
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Hash: SHA1
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 bein
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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 pho
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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 t
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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 d
: ;tag=as4db2b55c
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Contact:
> Content-Length: 0
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[
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Hash: SHA1
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 Aster
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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 sa
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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 metho
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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
he string using REGEX?
>
>
> On 9/6/06, *Rushowr* <[EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>> wrote:
>
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Rushowr wrote:
> > Steve Hsieh wrote:
> >> Greetings,
>
-BEGIN PGP SIGNED MESSAGE-
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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 wou
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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
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] Zapt
>-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 i
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: [asterisk
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-Commerci
> Cheers,
>
> Dean
>
>
>
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Rushowr
> > Sent: Monday, 28 August 2006 2:09 PM
> > To: 'Asterisk Users Mailing Li
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
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]
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 simp
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/Trix
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
ShafferSe
;m attempting to have multiple phones (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
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:
[a
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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 - Non-Co
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
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
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 O
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:
Asterisk-Users@li
-BEGIN PGP SIGNED MESSAGE-
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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 balance.
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
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
D
day, August 22, 2006 7:26 PM
>Cc: '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
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 INVI
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]
>[mailt
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
>S
I believe you want to use ${ENV(variable)}.. From asterisk's CLI:
*CLI>show function ENV
-= Info about function 'ENV' =-
[Syntax]
ENV()
[Synopsis]
Gets or sets the environment variable specified
Note that ENV is a function...you need to encase the argument inside
parentheses
>-Origina
>I now need to remove the 9 but then prefix another number onto
>the phone number before dialing now but am unsure how to do
>this is the dialplan.
Simple...for instance, if you wish to prefix 123 before the number just do:
Dial(SIP/123${EXTEN}
>
>Would someone be able to point me in the ri
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. I
Just gotta check, I've never seen a complete day with no posts
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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
SSH connection hangs 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
>
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 S
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:
>> ${CALLERID(number)}
>>
>>
>>> -Original Message-
>>> From: [EMAIL PROTECTED]
&g
Have you run SIP DEBUG PEER 192.168.1.60? It may help...tcpdump is also one
of my personal favorites
>-Original Message-
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of
>Diego Andres Asenjo G.
>Sent: Tuesday, August 22, 2006 6:50 PM
>To: asterisk-users@lists.digium.com
EMAIL PROTECTED] On Behalf Of Rushowr
>Sent: Tuesday, August 22, 2006 8:55 AM
>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>Subject: [asterisk-users] Setting RPID privacy?
>
>Hello all,
>
>Just had a question that I've not been able to find a suita
Hello all,
Just had a question that I've not been able to find a suitable answer for.
When we receive calls on SIP, we can get SIP_HEADER(Remote-Party-ID) and
check the privacy flag for what privacy is requested. Now, since SIP_HEADER
is not writable, how can I set the privacy flag in the RPID hea
Gotoif($["${ISNULL(${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
>Sub
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 using
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
_
I tries:
>Action: GetVar
>Variable ${CALLERID(227)}
>
>Neither 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
${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
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'
*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 Message
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 - N
who use it feel it's useless, I'll drop it and
>> do other useful things for Asterisk-- 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 sta
Presentation Setting: ${CALLINGPRES}) ; shows CID
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
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: Re:
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, Augus
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: [EMAIL
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: [asterisk-users
-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.
-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: Dougl
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,
I
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
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,
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", Jo
M
To: Asterisk Users 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 Wie
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
Gro
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: [aste
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 Han
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
been around for 20
years 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 n
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,
Rus
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: [asterisk-u
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 a
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