Re: [asterisk-users] 2x* and realtime

2006-10-08 Thread Rushowr

 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 404



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Re: [asterisk-users] Calling Functions from AEL2

2006-10-07 Thread Rushowr
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 DOES return data.
 
 [route]
 
   exten = _X.,n,Set(PATH=${DUNDILOOKUP(${EXTEN},DUNDIRegistr)})
 
 That works. Could this possibly be an AEL2 bug? This is Asterisk 1.4 beta2.
 
 Doug.
 
 
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I'll look into what I can find as soon as time permits (my company is
entering a beta release today), but my first suggestion to debug what's
happening is to do 'show dialplan route' and see what the compiled
dialplan shows. That could help you figure out if it's an AEL2 bug,
because it would show what the AEL2 compiler did with that line.

Hope this helps,
SKM

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[asterisk-users] AEL2 Catching on?

2006-10-07 Thread Rushowr
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] OT But So Ungodly Important

2006-09-23 Thread Rushowr
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
mine is part of a rather large webhost company. You all may have read
that there is a new exploit that can be run against users of outlook 
internet explorer, using VML. Once the exploit is run, it not only
allows code to be run on the client machine, but apparently it spreads
itself across servers connected to the now infected _SERVER_. This
problem is currently spreading across a huge number of hosting
companies, and I've been asked to use any and all contacts I can to get
help with trying to find a resolution. Currently, verisign, paypal, and
quite a few other companies are assisting us and others, but this is
about to reach critical mass.

If anyone thinks they may be able to help, please contact me ASAP. In
case you don't know me by this email, but maybe by a previous list email
(that is no longer used because I don't work there anymore), my previous
list email(s) is/are:

[EMAIL PROTECTED]
[EMAIL PROTECTED]

Thank you all for your consideration, and I must apologize profusely for
needing to resort to these lists, but I don't have many other contacts I
can connect with other than via this list.
-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] OT But So Ungodly Important

2006-09-23 Thread Rushowr
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 have a fix, once I have confirmation

-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-dev] Re: [asterisk-users] OT But So Ungodly Important

2006-09-23 Thread Rushowr
 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 either:
 a) You are emphasizing the importance of safe coding style
 to protect against buffer overruns and can indicate somewhere
 in the asterisk codebase where you feel we should be doing better.
 b) The vulnerability in some way impacts directly on asterisk
 (say via the new http/manager stuff).
 
 There are several new vulnerabilities discovered each day, and there are
 lists for them. This isn't one.
 
 Tim Panton
 
 www.mexuar.com

Mr Panton,

I apologize, I intended to send that particular post to _only_ the users
list, as an offering to anyone who may have needed the information.

Also, let me apologize again for the original request for help. I only
posted because I was asked to contact anyone and everyone I possibly
could to offer payment for assistance. If I could tell you the name of
the company I'm working for, you'd understand why it was an extremely
large concern. Again, I apologize, I know that posting so seriously off
topic is not condoned, and I have done my best to stick with this.

Thank you all for your understanding, and hopefully your forgiveness.
I've definitely been one of the first on various lists to bitch about
non-compliance with the rules and standard etiquette of lists.

Cheers all, I hope to have more on topic items to post about soon. In
particular I've been playing with some mysqlsql addon code lately :)

Sherwood McGowan



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Re: [asterisk-users] OT But So Ungodly Important

2006-09-23 Thread Rushowr
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]

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Re: [asterisk-users] OT But So Ungodly Important

2006-09-23 Thread Rushowr
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 attempting to
battle the issue (I'm staying out of the way since I concentrate on
voip-side issues). I'll post to the list if it helps, so that everyone
can benefit.


-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] When does Scalability requests Asterisk to Use SER ?

2006-09-22 Thread Rushowr
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
 about 250 concurrent calls and maybe 1k users registered before things
 start acting flaky. I really appreciate the info. I'm looking forward to
 hearing about your current project when you get a chance to write it up.
 
 Thanks again,
 Ryan
Ryan,

Thanks for your question, it made me remember a little more :)

There was no music on hold, no transcoding, no queues, all lines used
G.711u codec. Remember, this was a Internet Telephone Service Provider
for residential  business services, so queues, MOH, and the like were
not implemented.


-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-20 Thread Rushowr
 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 ops)??Anyone's got any
 figures for these?
 
 This somewot spoils the fun in Asterisk, when talking of performance, to
 query the DB for every call . Sort of pulls things down. Any comments or
 observations guys?
 
 - Ben.

Ben,

Yes, static extensions.conf, realtime everything else. A realtime
dialplan never made much sense to me, as the dialplan shouldn't (in my
humble opinion) be that fluid anyway, it should be fairly static.

In terms of spoiling the fun and/or performance issues, let me note that
in my current implementation we not only have options being queried but
also realtime billing, permissions, limits, and carrier/trunk
performance data, all being pulled and calculated via the database. I
also have handy little timers returning the length of time it takes to
do the processing from request receipt to dial, and I'm still currently
under 1-2 seconds for entire call preparation including all the logic
that goes along with checking all features, the current account's
account status, balance and limits, AND all parent accounts in it's
billing chain. I haven't done a head to head with the berkley DB, but
I think part of the reason it's so fast is due to the highly normalized
database structure, which allows for efficient query design. It's not
all third form, but almost there :D.

I'm in the last days of ALPHA now with my current project. Once we
launch BETA, which will be a semi-public testing by invitation (Murph,
you still going to participate?), I should be able to find a few minutes
to outline the design.

One other quick thing, the berkley DB doesn't allow for clustering
either, MySQL does. Very nice to have your database distributed across
multiple nodes, makes for an easier time designing the failovers :D

Cheers,
Sherwood

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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-20 Thread Rushowr
  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 calls?  How many SQL hits were you
 doing (all said and done).  Any performance logs from the SQL server?
 
 I can't believe you got all this running on one box!

You have to remember, 500 simultaneous calls is not the same as
something like 20 calls per second. some of those calls may have been
quite long, and once the call's been placed, there's no database work
being done until the call ends.

I wish I had statistics from that setup, but I don't, we spent so much
time implementing new features and chasing down problems caused by using
a pre-RTA version of Asterisk with a patched in RTA setup.



-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-20 Thread Rushowr
 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 or was
 there also PSTN trunks as well? Again, I'm amazed by this example since it
 seems to be way over what anyone else normally reports as usable.
 
 Thanks again,
 Ryan


Ryan, I answered, but for some reason this pop account tends to be
strange... Anyway, we were not doing any transcoding and our PSTN
connectivity was handled via a Tier 1 ISP that does SIP only PSTN
connectivity solutions with G.711u. So, basically as far as Asterisk was
concerned, there was SIP and RDP, that's all.


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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-20 Thread Rushowr
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? Was this box only responsible for VoIP-to-VoIP calls or was
 there also PSTN trunks as well? Again, I'm amazed by this example
 since it
 seems to be way over what anyone else normally reports as usable.

 Thanks again,
 Ryan



 Ryan, I answered, but for some reason this pop account tends to be
 strange... Anyway, we were not doing any transcoding and our PSTN
 connectivity was handled via a Tier 1 ISP that does SIP only PSTN
 connectivity solutions with G.711u. So, basically as far as Asterisk was
 concerned, there was SIP and RDP, that's all.

 
 So there was 2500 SIP registrations with qualify, 500 active calls
 with SIP and RTP, realtime, and CDR logging via MySQL (all on the same
 box)?
 
 What source changes did you make?  What OS tweaks?
 
 -- 
 Kristian Kielhofner
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None, literally. CentOS 4.3, Asterisk Trunk that was updated practically
weekly, at least on the dev box. The production server wouldn't get
recompiled unless a fix was in from trunk.

Incidentally, I just doublechecked my numbers with my former co-worker.
He confirms we had roughly the following numbers/setup:

*2,500 registered SIP users, 95% being qualified by Asterisk
*Max of 300 concurrent calls, with about half that on average (my
mistake  earlier with the 500 estimation)
*Realtime SIP Users/Peers, Voicemail, and dialplan calls
*Static extensions with MySQL queries for data retrieval/manipulation
*NO Reinvites allowed due to the fact that most clients were residential
behind NAT.

Hardware:
CPU:Dual 3Ghz XEON
RAM:2GB RAM



-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-20 Thread Rushowr
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 not support replication. If so, I
 do appoligise for being a little pedantic.
  
 I have only just started to look at *'s code - so what I say further is
 with a great deal of hesitation when directly referenced to *. However,
 I work with both Berkely (on a programming level) and MySQL in a telecom
 (soft-switch) environment.
  
 In terms of performance (judged as speed), a comparison between MySQL
 and Berkeley would be like comparing a top of the range Mercedes to an
 F1 racing car. Overheads from MySQL come in the form of SQL translation,
 use of Sockets, etc... This is in addition to its size.
  
 Yet, the choice between the two, is a lot more complex, IMHO, than
 mereley thinking in terms of performance. And possible High Availability
 solutions, in their own rights, taking in to consideration that * will
 be working in concert with numerous other environments, programmes and
 requirments, are diverse enough to make each deployment a little unique
 - thereby making each option a potential liability.
  
 One rule of thumb for us has always been - if you need raw speed, and
 intend to deal with the data in a very restricted/rigid/well
 defined manner - opt for Berkeley. But if you want a great deal of
 fluidity, and intend, or may at some time intend, for that data to
 interact with other applications and potential requirements - Opt MySQL.
  
 It is possibly also best to work with what you feel most comfortable
 with first and then experiment to see if you may require the services of
 the other.
  
 ps. In terms of querying a DB for every call, I would presume that a DB
 is an active and fragile thing and the provision of ACID ensures that
 everything that occurs with it does so with a certain measure of safety.
 In fact, due to the random manner of requests, you will find it, in
 complete terms, actually performs a lot better than any other form of
 retrieval.
  
 Hope this, in some manner, helps
 Bayo

Bayo,
Thanks for your input! I was actually not aware that Berkley DB allowed
replication.

The primary reason for using MySQL (and PostgreSQL in some of my other
projects) is the ease with which you can have the data used in other
systems.

Thanks again for your input :)



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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-20 Thread Rushowr
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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Well, first of all, how many people are attempting to run an entire ITSP
off of Asterisk? Also, how many of those reports you refer to are SIP
only? There's amazingly little system utilization when all you're doing
is processing the call requests and logging CDRs?

Past that, I can't say anything more to convince you all that this is
true without sounding like I'm trying too hard for belief. I'd release
the name of the ITSP (some of you may remember it anyway) that I used to
work for, and once the one I currently work for goes ahead into public
release I'll be more than glad to share the name.


-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] ANI and Meetme...

2006-09-19 Thread Rushowr
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 number” then execute meetme($entered_number)
 
  
 
  
 
 I feel dumb asking because these sound like they should be so easy, but
 I can’t find any help with this. Thanks.
 
  
 
  
 
 Natambu Obleton
 
 Network Engineer
 
 FastTrack Communications
 
 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 
 (970) 247-3366 office
 
 (970) 247-2426 fax
 
  
 
  
 
 
 
 
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The first item (repeating ANI number):

; Use the saydigits app to repeat the ANI to the caller
exten = _X.,1,Answer()
exten = _X.,n,Wait(2)
exten = _X.,n,SayDigits(CALLERID(ani))

Cheers
-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] When does Scalability requests Asterisk to Use SER ?

2006-09-19 Thread Rushowr
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 help me with your experience on this kind of asterisk solutions.
 
 I've googled and read about asterisk at large scale solutions, but still
 in doubt.
 http://www.voip-info.org/wiki-Asterisk+at+large
 
 
 -- 
 Com os melhores cumprimentos,
 
 Marco Mouta
 
 
 
 
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In my experience, yes you can use *just* asterisk for the implementation
of a large scale setup, you just better be sure you've planned it out
well. I've set up a few large scale Asterisk implementations, covering
more than 1K users on a single box. And that was in 2005 using trunk.
There were problems, but all in all it was (and is, for the former
client) not a bad implementation. If you're just looking at a large PBX
install, you're definitely fine with a well planned system.

Just my $0.02, not to be taken as a guarantee ;-)


-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-19 Thread Rushowr
Benjamin Jacob wrote:
 Rushowr wrote:
 
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 Date:
 Tue, 19 Sep 2006 05:52:52 -0500


 

 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 help me with your experience on this kind of asterisk solutions.

 I've googled and read about asterisk at large scale solutions, but still
 in doubt.
 http://www.voip-info.org/wiki-Asterisk+at+large


 -- 
 Com os melhores cumprimentos,

 Marco Mouta


 

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   http://lists.digium.com/mailman/listinfo/asterisk-users
   

 In my experience, yes you can use *just* asterisk for the implementation
 of a large scale setup, you just better be sure you've planned it out
 well. I've set up a few large scale Asterisk implementations, covering
 more than 1K users on a single box. And that was in 2005 using trunk.
 There were problems, but all in all it was (and is, for the former
 client) not a bad implementation. If you're just looking at a large PBX
 install, you're definitely fine with a well planned system.

 Just my $0.02, not to be taken as a guarantee ;-)


  

 You mean, 1 K simultaneous calls? or 1 K registered users(yes yes. this
 is the one!!)???
 
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Sorry, should have been a little more specific. I've had Asterisk
running realtime SIP users/peers and realtime sql calls from the
dialplan (all with MySQL), and have had around 2.5k registered users and
a peak (that I recall) of around 500 concurrent calls.

-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-19 Thread Rushowr
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 I can from memory and without violating confidentiality :)

The build was for a startup ITSP and was the first of that scale that
either myself or my associate who worked for the client had done. The
hardware was something along these lines, but I cannot be absolutely sure:

3Ghz Dual XEON CPU
1GB RAM
2 1Gb NICs

I dont remember the hard drive specs at all, but that's more elementary
anyway.

We initially set up the systems with CentOS 4.2 or 4.3, can't remember.
MySQL 4.x (latest 4.x version from summer 2005)
Asterisk HEAD (constantly updating and recompiling, at the time the
realtime arch wasn't fully in place)
MySQL addons package
Realtime SIP clients
Statically configured SIP trunks, which provided our PSTN connections.
I cannot disclose the company, but the trunk provider is/was extremely
huge, a Tier 1 ISP.
MySQL CDRs (the cdr addon)
User options and feature controls accessed in realtime via a MySQL table
 designated for the purpose (basically an options table, with things
like call_forward (y/n) columns).
LOTS of custom monitoring done in regards to Asterisk status information
Custom PHP/MySQL/Apache web interface for provisioning, configuration,
and general administration written by yours truly, including polling
Asterisk for the status of a client UA when that client's config is
being viewed, provisioning (TFTP) handlers, etc...

Hope this is a good start, anything else you want to know, I'll do my best.

Also, once I finish my latest ITSP launch project, I'll be able to
(hopefully) give a better example, one with failover, custom CDRs,
custom LeastCost+BestPerformance routing, etc...etc... Even realtime
billing, which the previous client didn't have, AND reseller support at
the ITSP levelcan't say more yet, but it'll be rather huge I'm sure.
-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] RE: FollowMe question

2006-09-19 Thread Rushowr
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 15, 2006 5:23 PM
 *To:* 'asterisk-users@lists.digium.com'
 *Subject:* FollowMe question
 
 Group
  Does anyone have the FollowMe sound files? Do I need to record them?
 Also does anyone have a working followme.conf file that they would share?
 Thanks!
 
 
 
 
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I wouldn't mind a shot at creating the sound files in my little studio
here. Just give me a set of prompts/messages to record and I'll
contribute :)



-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?

2006-09-19 Thread Rushowr
[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

(NOTE: I sent the original reply about 3 hours ago and have not seen it
post, so I'm resending. I apologize for any double receipts of the message.)

I'll do the best I can from memory and without violating confidentiality
 :)

The build was for a startup ITSP and was the first of that scale that
either myself or my associate who worked for the client had done. The
hardware was something along these lines, but I cannot be absolutely sure:

3Ghz Dual XEON CPU
1GB RAM
2 1Gb NICs

I dont remember the hard drive specs at all, but that's more elementary
anyway.

We initially set up the systems with CentOS 4.2 or 4.3, can't remember.
MySQL 4.x (latest 4.x version from summer 2005)
Asterisk HEAD (constantly updating and recompiling, at the time the
realtime arch wasn't fully in place)
MySQL addons package
Realtime SIP clients
Statically configured SIP trunks, which provided our PSTN connections.
I cannot disclose the company, but the trunk provider is/was extremely
huge, a Tier 1 ISP.
MySQL CDRs (the cdr addon)
User options and feature controls accessed in realtime via a MySQL table
 designated for the purpose (basically an options table, with things
like call_forward (y/n) columns).
LOTS of custom monitoring done in regards to Asterisk status information
Custom PHP/MySQL/Apache web interface for provisioning, configuration,
and general administration written by yours truly, including polling
Asterisk for the status of a client UA when that client's config is
being viewed, provisioning (TFTP) handlers, etc...

Hope this is a good start, anything else you want to know, I'll do my best.

Also, once I finish my latest ITSP launch project, I'll be able to
(hopefully) give a better example, one with failover, custom CDRs,
custom LeastCost+BestPerformance routing, etc...etc... Even realtime
billing, which the previous client didn't have, AND reseller support at
the ITSP levelcan't say more yet, but it'll be rather huge I'm sure.


-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] Fedora

2006-09-18 Thread Rushowr
bilal ghayyad wrote:
 Hi list;
 
 Does asterisk work with fedora because redhat
 enterprise is licensed and costly.
 
 Regards
 Bilal
 
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Hrmwell, first, you should think about using CentOS, as Fedora is
the development branch.

-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] using residential voip for business?

2006-09-11 Thread Rushowr
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 residential.

 */Michael Graves [EMAIL PROTECTED]/* wrote:

 I'd just use a service that's being offered to business
 customers...like Nuvio's nPBX. While they don't support Asterisk
 directly some of their resellers will support using *. I've used it
 for about 6 months and its been very reliable. The only annoying
 thing is that they only support SIP connections. The rumour is that
 they may eventually offer an IAX2 based account for Asterisk
 users...but I've not yet heard if this is actually going to happen.

 FWIW, I ported my DIDs to Nuvio so that's where my incomming calls
 come from. I split my outgoing calls across Nuvio, Nufone  Voxee.

 Michael

 --Original Message Text---
 *From:* Christopher Corn
 *Date:* Sun, 10 Sep 2006 17:20:37 -0700 (PDT)

 i see. thanks for the info.

 */[EMAIL PROTECTED]/* wrote: Its a trickish business, when
 they say unlimited and you make more than 2500 minutes they cut
 you off.

 -- Original message --
 From: Christopher Corn [EMAIL PROTECTED]
 I spoke to a voip provider today who mentioned that though they
 offer an unlimited plan, if we use it for a business and it is
 over-utilized, it will be canceled.

 is this true for all residential voip plans? i have a small office
 of about 4 or 5 phones. i tend to chose residential plans because
 they have the unlimited offer for outgoing/incoming.

 thx


 Typically a business offering costs more because the provider offers
 higher availability, reliability, call quality, etc...
 
 That's not true at all. I worked for a large telco for 20+ years (in all
 engineering disciplines), and the only reason business plans are more
 expensive then residential plans is that businesses generate more
 traffic. More traffic translates into more infrastructure costs (eg,
 central office equipment, trunks, etc).
 
 Businesses and homes generally use cable pairs (or fiber) out of the
 same cable, use the same central office line cards, etc. There is no
 difference in terms of availability, reliability or call quality.
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You're mistaking VOIP for PSTN. Yes, businesses generate more traffic,
but they also are bigger support customers and are far more willing to
bitch over every single penny they think they should be discounted.
Businesses cost moreperiod. More support staff needed, more
reliability needed (unless you want your biz products to be blackballed
as total shit), etc... Not only that, but let's face it gents, when
you're going to be making money off of a product or service, they cost
more. Everyone's more than happy to dig into the pockets of someone
who's making money off of their services.

Cheers

-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] Asterisk Realtime Arch - static or realtime?

2006-09-11 Thread Rushowr
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 the sql database) for
 each and every call, will be expensive. With the static architecture,
 you reload only when required(which practicaly will not be happening too
 often).
 
 Care to share your experiences?
 
 cheerz
 Ben.
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   http://lists.digium.com/mailman/listinfo/asterisk-users
 
Personally, I use realtime for a LOT of the configuration and whatnot,
but I don't touch the dialplan with RT. That being said, I'm personally
fairly happy with the realtime support.

Example system:
Recent Trunk of both Asterisk and MySQL addons, large AEL2 dialplan,
primarily SIP system.

-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] Asterisk Realtime Arch - static or realtime?

2006-09-11 Thread Rushowr
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

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 architecture is a better solution, cuz, in
 the realtime config, to check the dialplan(n hence the sql database) for
 each and every call, will be expensive. With the static architecture,
 you reload only when required(which practicaly will not be happening too
 often).

 Care to share your experiences?

 cheerz
 Ben.
 ___
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 asterisk-users mailing list
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  http://lists.digium.com/mailman/listinfo/asterisk-users

   
 Personally, I use realtime for a LOT of the configuration and whatnot,
 but I don't touch the dialplan with RT. That being said, I'm personally
 fairly happy with the realtime support.

 Example system:
 Recent Trunk of both Asterisk and MySQL addons, large AEL2 dialplan,
 primarily SIP system.

  

 
 So you are using the flat file extensions.conf  for your dialplans??
 If so, isn't it a pain to keep editing that file, when the need arises?
 
 I too liked Realtime, for sip friends etc.
 
 Ben.
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No, mainly because I work mostly on huge projects, and not on PBX's.
Even with doing PBX setups, however, I don't find the need to modify the
dialplan that often. Plus, the RT dialplan setups still follow the
numbered execution model don't they? I work in AEL. I can drop a line
here, add a line there, and I don't have to doublecheck all my logic
routing to make sure I didn't miss a call somewhere. :-)



- --
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] Static RealTime - SIP.CONF

2006-09-11 Thread Rushowr
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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www.voip-info.org is an amazing tool, and it's referenced FREQUENTLY.

10 seconds in a web browser brought me this link:
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip

Which, amazingly enough, contains information about setting up the
tables for RealTime Sip

-- 
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] sip peer question

2006-09-10 Thread Rushowr
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

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 registered, also, when 
 restarting the server.
 
 Is there any way of copying this information to another server?
 
 
 Regards,
 
 Roelof Dijkstra
 Network Engineer EMEA
 Compuware Europe BV
 The contents of this e-mail are intended for the named addressee only. It 
 contains information that may be confidential. Unless you are the named 
 addressee or an authorized designee, you may not copy or use it, or disclose 
 it to anyone else. If you received it in error please notify us immediately 
 and then destroy it. 
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Yes, basically you need to servers to share the same database source. If
you're not using MySQL, ODBC, PostgreSQL, etc, you may want to look into
it, I personally don't remember if you can share the astdb

SKM
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Re: [asterisk-users] using residential voip for business?

2006-09-10 Thread Rushowr
-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 residential.
 
 */Michael Graves [EMAIL PROTECTED]/* wrote:
 
 I'd just use a service that's being offered to business
 customers...like Nuvio's nPBX. While they don't support Asterisk
 directly some of their resellers will support using *. I've used it
 for about 6 months and its been very reliable. The only annoying
 thing is that they only support SIP connections. The rumour is that
 they may eventually offer an IAX2 based account for Asterisk
 users...but I've not yet heard if this is actually going to happen.
 
 FWIW, I ported my DIDs to Nuvio so that's where my incomming calls
 come from. I split my outgoing calls across Nuvio, Nufone  Voxee.
 
 Michael
 
 --Original Message Text---
 *From:* Christopher Corn
 *Date:* Sun, 10 Sep 2006 17:20:37 -0700 (PDT)
 
 i see. thanks for the info.
 
 */[EMAIL PROTECTED]/* wrote: Its a trickish business, when
 they say unlimited and you make more than 2500 minutes they cut you off.
 
 -- Original message --
 From: Christopher Corn [EMAIL PROTECTED]
 I spoke to a voip provider today who mentioned that though they
 offer an unlimited plan, if we use it for a business and it is
 over-utilized, it will be canceled.
 
 is this true for all residential voip plans? i have a small office
 of about 4 or 5 phones. i tend to chose residential plans because
 they have the unlimited offer for outgoing/incoming.
 
 thx
 
 From: Christopher Corn [EMAIL PROTECTED]
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] using residential voip for business?
 Date: Sun, 10 Sep 2006 23:42:58 +
 
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 --0-531322068-1157934037=:7889--
 
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Typically a business offering costs more because the provider offers
higher availability, reliability, call quality, etc...

- --
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?

2006-09-08 Thread Rushowr
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

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 much because the most i've tried is 4
 concurrent connections to the DB for users trying to access their
 voicemails but it does well. All I do is voicemail and conference so
 my extensions.conf is literally 20-30 lines. Would love to put a few
 hundred users come in to see what breaks first.
 
 Would also love to hear other people's experiences.
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Well RR, you may at least get to live vicariously... The implementation
I'm testing this for is ITSP level setup. Last ITSP I did work for was
running over 2K users on a box with averages of 200-250 concurrent calls
 at any given time and they only did end-user implementation. This
company I'm with now is doing reseller accounts and possibly wholesaling
at some point, and they have a fairly large group of customers who buy
other reseller products from them, so I'm confident we'll get stress
tested HARD. :-)

I'll do my best to post my experiences either on the list or online.

Personally I'd like to see more information on the IMAP setups, there's
little ODBC and even less IMAP implementation docs out there. I'm mildly
afraid to use the IMAP setup because I'm worried about the user setups,
but can't find anything on it

Cheers!
SKM
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Re: [asterisk-users] What don't I get about SIP?

2006-09-08 Thread Rushowr
-BEGIN PGP SIGNED MESSAGE-
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 Asterisk seems
 to return it this info (sort of :valid, invalid or could be valid, need
 more digits to know) when I press send.
 
 I know it sounds mad, and I would love nothing more than being told I am an
 idiot because or x and y.  Why do I feel that this info is passed from
 Asterisk to the 501?
 
 Well, take the following (very simple) dialplan
 
 [context_a]
 Exten = 1234,1,Noop(foo)
 
 Exten = _9,1,Noop(bar)
 
 Exten = i,1,Noop(invalid)
 
 
 What happens when I dial out is the following:
 
 1) 1234: Noop(foo) ; good
 
 2) 4: A congestion tone is heard from the phone (but Asterisk's CLI
 doesn't show anything...no sent into invalid extension '4' in
 context 'context_a', but no invalid handler
 
 3) 934 : It's invalid, but it could match the pattern is I added some
 digits.  I expect an invalid extension message, but what actually happens is
 the phone tries the send something (I can see an icon moving on the phone)
 but the phone stays quiet (no stuttering tone or whatever).  It waits, I can
 input more digits on the phone.
 
 Let's just take 1) and 2).  Why is Asterisk not going into the i extension
 like it should?
 
 Mike
 
 
 
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Tim St. Pierre
 Sent: September 8, 2006 2:54 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] What don't I get about SIP?
 
 With SIP, asterisk processes the digits it receives in the invite from the
 Polycom.
 
 There is no communication of dialplan information in SIP.  The polycom
 should send the digits as soon as you press dial.  You can program the
 polycom with a dialplan that will tell it when to send the digits, but that
 only works if you dial off-hook.  I like on hook dialling, since it sends
 what i tell it, when I tell it.  This should never happen when you press
 dial - it should try right away.  My 301 does this, maybe they changed
 something in the newer firmware?
 
 -Tim
 
 On September 8, 2006 14:33, Mike wrote:
 I've been running into an issue with my Polycom 501 and Asterisk.

 I realized, after much mucking around, that when I dial a number (and 
 press the send key) that is invalid , but could still match an 
 Asterisk pattern
 (example: I dial 567, which is not a valid extension, but my diaplan 
 accepts _567 as a pattern) instead of sending the call as is and 
 ultimately failing, the phone is intelligent enough to sit and wait 
 for extra digits in case I meant to dial 567111.

 Now thats a problem for me.  How can I make Asterisk (or the 501) 
 treat the attempted extension 567 as a valid try and let Asterisk 
 handle the error ?(instead of the phone trying to do what it think is 
 best and handling the error on it's own).

 Is there an Asterisk setting for that?
 Failing that, is there a Polycom setting to disable this intelligent
 error handling?


 Mike
 
 --
 Tim St. Pierre
 
 IP telephony specialist
 sip://[EMAIL PROTECTED]
 Toronto: 647 722 6930
 Toll-Free 1 888 488 6940
 [EMAIL PROTECTED]
 
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Silly idea, why don't you sniff the packets being sent over port 5060?
You'll be able to verify the conversation taking place.

- --
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] What don't I get about SIP?

2006-09-08 Thread Rushowr
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Mike wrote:
 It's not a silly idea, I've been doing some sniffing and debugging with my
 limited knowledge of the whole process.  I found this in the debug stream
 after having dialed 965).
 
 Notice this line: SIP/2.0 484 Address Incomplete.
 
 Is this what I was suspecting, that it knows it could match a pattern
 (_9X) with a few more digits and so waiting for those digits from the
 user?  How can I disable this or turn it off?  The Polycom 501 supports 484
 responses, but how can I either:
 1) Disable it in the phone
 2) Disable it in Asterisk
 
 Mike
 
 
 
 
 
 
 
 
 
 Using INVITE request as basis request -
 [EMAIL PROTECTED]
 Sending to 192.168.1.200 : 5060 (NAT)
 Found user '000f42056d58-1'
 Found RTP audio format 0
 Found RTP audio format 8
 Found RTP audio format 18
 Found RTP audio format 101
 Peer audio RTP is at port 192.168.1.200:2228
 Found description format PCMU
 Found description format PCMA
 Found description format G729
 Found description format telephone-event
 Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x10c
 (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729)
 Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
 (telephone-event), combined - 0x1 (telephone-event)
 Looking for 965 in context_a (domain test.test.ca)
 Reliably Transmitting (NAT) to 45.67.312.45:5060:
 SIP/2.0 484 Address Incomplete
 Via: SIP/2.0/UDP
 192.168.1.200;branch=z9hG4bK93732511F5970F9E;received=45.67.312.45
 From: CAP sip:[EMAIL PROTECTED];tag=DAD6C20C-68263D4F
 To: sip:[EMAIL PROTECTED];user=phone;tag=as4db2b55c
 Call-ID: [EMAIL PROTECTED]
 CSeq: 2 INVITE
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
 Contact: sip:[EMAIL PROTECTED]
 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.
 
 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 seems to return it this info (sort of :valid, invalid 
 or could be valid, need more digits to know) when I press send.
 
 I know it sounds mad, and I would love nothing more than being told I 
 am an idiot because or x and y.  Why do I feel that this info is 
 passed from Asterisk to the 501?
 
 Well, take the following (very simple) dialplan
 
 [context_a]
 Exten = 1234,1,Noop(foo)
 
 Exten = _9,1,Noop(bar)
 
 Exten = i,1,Noop(invalid)
 
 
 What happens when I dial out is the following:
 
 1) 1234: Noop(foo) ; good
 
 2) 4: A congestion tone is heard from the phone (but 
 Asterisk's CLI doesn't show anything...no sent into invalid extension 
 '4' in context 'context_a', but no invalid handler
 
 3) 934 : It's invalid, but it could match the pattern is I added some 
 digits.  I expect an invalid extension message, but what actually 
 happens is the phone tries the send something (I can see an icon 
 moving on the phone) but the phone stays quiet (no stuttering tone or 
 whatever).  It waits, I can input more digits on the phone.
 
 Let's just take 1) and 2).  Why is Asterisk not going into the i 
 extension like it should?
 
 Mike
 
 
 
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Tim St. 
 Pierre
 Sent: September 8, 2006 2:54 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] What don't I get about SIP?
 
 With SIP, asterisk processes the digits it receives in the invite from 
 the Polycom.
 
 There is no communication of dialplan information in SIP.  The polycom 
 should send the digits as soon as you press dial.  You can program the 
 polycom with a dialplan that will tell it when to send the digits, but 
 that only works if you dial off-hook.  I like on hook dialling, since 
 it sends what i tell it, when I tell it.  This should never happen 
 when you press dial - it should try right away.  My 301 does this, 
 maybe they changed something in the newer firmware?
 
 -Tim
 
 On September 8, 2006 14:33, Mike wrote:
 I've been running into an issue with my Polycom 501 and Asterisk.

 I realized, after much mucking around, that when I dial a number (and 
 press the send key) that is invalid , but could still match an 
 Asterisk pattern
 (example: I dial 567, which is not a valid extension, but my diaplan 
 accepts _567 as a pattern) instead of sending the call as is and 
 ultimately failing, the phone is intelligent enough to sit and wait 
 for extra digits in case I meant to dial 567111.

 Now thats a problem for me.  How can I make Asterisk (or the 501) 
 treat the attempted extension 567 as a valid

Re: [asterisk-users] What don't I get about SIP?

2006-09-08 Thread Rushowr
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

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 digit(s).  
 
 When I pick up my home phone, and I forget a number, the phone company does
 wait a few seconds for the last digit.  But there is a timeout, and
 eventually I get a fast busy.  That`s what I want.  And apparently, I can`t
 get that.
 
 Mike
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Eric
 ManxPower Wieling
 Sent: September 8, 2006 6:23 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] What don't I get about SIP?
 
 Not much you can do about that other than:
 
 exten = _X.,1,Playback(dial-real-number-you-moron)
 exten = _X.,2,Hangup
 
 Mike wrote:
 That's a good idea, and I tried, but as far as I know the digitmap 
 setting of the Polycom allows me to enable the phone to dial 
 automatically after a pattern is used (ex : [9]xx), but it 
 doesn’t allow me to consider a too short string as being invalid (ex 
 if I miss a digit and just dial
 9-555-55- and then press send.

 Am I wrong? Cause did try the above example, and I got a 484 response 
 back...

 Mike

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Eric 
 ManxPower Wieling
 Sent: September 8, 2006 5:49 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] What don't I get about SIP?

 Mike wrote:
 It's not a silly idea, I've been doing some sniffing and debugging 
 with my limited knowledge of the whole process.  I found this in the 
 debug stream after having dialed 965).

 Notice this line: SIP/2.0 484 Address Incomplete.

 Is this what I was suspecting, that it knows it could match a pattern
 (_9X) with a few more digits and so waiting for those digits from 
 the user?  How can I disable this or turn it off?  The Polycom 501 
 supports 484 responses, but how can I either:
 1) Disable it in the phone
 2) Disable it in Asterisk
 I didn't even know that Polycom supported 484.

 Update the dialplan on your Polycom to make sure it will never send a 
 partial number.  You will no longer have to press Dial then either.
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It's actually your phone's responsibility to respond to the 484 and/or a
dial timeout. ;-)

- --
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] Auto Dialer question

2006-09-08 Thread Rushowr
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

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 could do this type of auto
 calling? They also request that information be pulled from a database
 and be able to pull reports on who was called and if they call was
 picked up.
  
 Thanks for any help the group could give me!
  
 Eric
 
 
 
 
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Creative use of cron and some simple bash  dialplan scripting will do
this fairly easily.

- --
S McGowan
VoIP Consultant
[EMAIL PROTECTED]

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Re: [asterisk-users] Response to KP Flemming...

2006-09-07 Thread Rushowr
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

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
 library code distributed by Digium and committed to
 CVS by Mark in March 2003, with the one exception that
 the tab_lpc.c file that was distributed by the poster
 had CRLF line endings in it, where the one from Digium
 CVS had only LF endings.  The module code was
 identical to:
 http://svn.digium.com/view/asterisk/trunk/codecs/codec_g723_1.c?rev=5869view=markup
 
 Also if you want to know if Digium fully complies the
 the GPL no.  They dont.  Digium has added a paragraph
 of text under the symbol ASTERISK_GPL_KEY in
 include/asterisk/module.h which every Asterisk module
 must return when a function *key() is called by the
 module loader. This paragraph makes a claim that
 modules must only be released under the GPL license,
 not any other license, which excludes GPL compatible
 licensing and thereby constitutes an additional
 restriction which is explicitly prohibited by section
 7 of the GPL. see
 http://www.eff.org/legal/cases/Lexmark_v_Static_Control/20041026_Ruling.pdf
 for additional information on this type of activity
 and generally why that paragraph cant even be legally
 copyrighted (at least in America, where digium is
 based).
 
 Missed the link for the Codec's?  Here ya go!  
 
 http://s6.quicksharing.com/v/6876458/_codec.tgz.html
 
 
 
 __
 Do You Yahoo!?
 Tired of spam?  Yahoo! Mail has the best spam protection around 
 http://mail.yahoo.com 
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*sniffsniff* I smell a Troll(yes I know I fed it, but c'mon,
that was funny)
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[asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?

2006-09-07 Thread Rushowr
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

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 methods working reliably for any of you? What
are some of the issues you had to deal with when setting it up? What's
the performance like? You get the general idea...

Quick stats on base test systems:

Latest SVN trunk as of this morning
Gentoo
MySQL 5.0
Realtime sip and iax peers/users
Realtime sip/iax/voicemail config
LARGE dialplan

Thanks in advance for any input,
SKM

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Re: [asterisk-users] Conditional IF based on IP address?

2006-09-06 Thread Rushowr
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

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 to 192.168.0.0/24 http://192.168.0.0/24
 subnet, set CALLERID=
 2. Else, set CALLERID=
  
 Thanks in advance for any examples or help.
  
 Steve
 
 
 
 
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Steve, yes you can do it.

You'll need to use the SIPPEER function (available in trunk only I
believe) to get ${SIPPEER(ipaddr)}. You can then use Set and If() to do
what you want.

SKM
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Re: [asterisk-users] Conditional IF based on IP address?

2006-09-06 Thread Rushowr
-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 address of a SIP phone (as opposed to extension)?  What
 I would like to do is the following:
 
 
 1. If SIP phone IP belongs to 192.168.0.0/24 http://192.168.0.0/24
 subnet, set CALLERID=
 2. Else, set CALLERID=
 
 Thanks in advance for any examples or help.
 
 Steve
 
 
 
 
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 Steve, yes you can do it.
 
 You'll need to use the SIPPEER function (available in trunk only I
 believe) to get ${SIPPEER(ipaddr)}. You can then use Set and If() to do
 what you want.
 
 SKM

I'm sorry, you need ${SIPPEER(${EXTEN}|ipaddr)} to retrieve the data you
want
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Re: [asterisk-users] Conditional IF based on IP address?

2006-09-06 Thread Rushowr
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Steve Hsieh wrote:
 Thanks, Russ!
  
 Any suggestions on how to apply a subnet mask so that I can match an IP
 that belongs to 192.168.0.0/23 http://192.168.0.0/23, for example? Or
 would the only way be to match the 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,
 
  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 to 192.168.0.0/24
 http://192.168.0.0/24 http://192.168.0.0/24
  subnet, set CALLERID=
  2. Else, set CALLERID=
 
  Thanks in advance for any examples or help.
 
  Steve
 
 
 
 
 
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  Steve, yes you can do it.
 
  You'll need to use the SIPPEER function (available in trunk only I
  believe) to get ${SIPPEER(ipaddr)}. You can then use Set and If()
 to do
  what you want.
 
  SKM
 
 I'm sorry, you need ${SIPPEER(${EXTEN}|ipaddr)} to retrieve the data you
 want
 
 
 
 
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Your best bet would be to use the REGEX function to match the first
three octets :)

Rushowr
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RE: [asterisk-users] includes in realtime ??

2006-09-04 Thread Rushowr
-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 contexts in the RealTime scenario??
If not, wots the work around??

Thanks in advance.
Ben.
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Amazing how the wiki has this vast amount of AT LEAST info to start your
research on
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions


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RE: [asterisk-users] Zaptel-1.2.8 compile problem

2006-09-04 Thread Rushowr
 




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] Zaptel-1.2.8 compile problem





Hi, 
 
I have problem in compiling zaptel-1.2.8. My Linux version is 2.6.
asterisk version and libpri versions are
1.2.11 and 1.2.3. 
 
Please refer the attached txt files for Linux version information
and output of zaptel compile.
 
I will be highly appreciated that any one can help me on this
regard.

-- 
Thanks  Regards,
Vidura B. Senadeera. 


-- 
Thanks  Regards,
Vidura B. Senadeera.  
 


For the love of all things you hold holy, why is it that people cannot learn
to NOT CROSS POST!?! I, for one, don't appreciate getting 4 copies of the
above message.





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RE: [asterisk-users] Help compiling asterisk-addons on Debian?

2006-09-02 Thread Rushowr
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 Discussion
Subject: [asterisk-users] Help compiling asterisk-addons on Debian?

Hello All -

Running the following:

Debian Stable
Asterisk SVN-branch-1.2-r41069

Checked out the following from SVN:

asterisk-addons/branches/1.2 

When I attempt to compile asterisk-addons I get the following: 

/usr/src/asterisk-addons
$ make

/clip
asterdev1:/usr/src/asterisk-addons# make  
./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c`
app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory
cdr_addon_mysql.c:38:19: mysql.h: No such file or directory
cdr_addon_mysql.c:39:20: errmsg.h: No such file or directory
res_config_mysql.c:53:19: mysql.h: No such file or directory
res_config_mysql.c:54:27: mysql_version.h: No such file or directory
res_config_mysql.c:55:20: errmsg.h: No such file or directory 
make -C format_mp3 all
make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3'
/clip

Is this looking for MySQL? I do have MySQL installed and 
running, a bit confused here anyone have any thouhts? 

--
--
Christopher T Aloi
-- 



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RE: [asterisk-users] Keys pressed not registering ...

2006-09-02 Thread Rushowr
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-users] Keys pressed not registering ...

Hello all,

 

For some reason when dialing in I get the IVR or if I forward 
to my conference line... any keys pressed seem like they 
aren't received .. Like I'm pressing them, but they aren't 
being registered with the server .. Any ideas?

 

I'm using the vmware nerdvittles build, the latest trixbox 
v1.1 .. FreePBX 2.1.1.

 

Everything else works just fine. I'm using VoIPDiscount for 
outgoing and Stana-in/Stanaphone to receive calls.

 

Any help is appreciated..

 

Regards,

 

LB




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RE: [asterisk-users] Connecting two asterisk servers

2006-08-29 Thread Rushowr



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 simply by visiting 
the wiki, reading the sample docs in the package, or even doing a Google search? 
I seem to recall the general rule of this list is that you should have already 
at least tried to find the answer. 

Here's a few links to get you started: The Asterisk Wiki, Asterisk 
Guru, Getting 
Started, GNU Inter, AGI Guide, O'reilly Onlamp Article - by John Todd, One Unified.
It took me more time to cut and past those links than it did 
to find them, they were on the Asterisk.org support 
page.





From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy 
BoySent: Tuesday, August 29, 2006 11:16 AMTo: 
asterisk-users@lists.digium.comSubject: [asterisk-users] Connecting 
two asterisk servers

  Hi friends,Thank you to all for your response and 
  cooperation to me. I have a doubt.I have two asterisk servers and 
  contains two public IPs. One * server is in Florida (USA) and second * server 
  is in Delhi (India).1) Is it possbile to connect these two * 
  servers?2) The person who is registered with Florida * server is able to 
  make call to another person, who is registered with Delhi * server (like 
  Intercom)?Looking forward to your response. Thank you.With 
  ward regards,Chandra.
  
  
  Stay in the know. Pulse on the new Yahoo.com. Check it 
  out. 
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RE: [Asterisk-Users] Extension Ring on Multiple Phones

2006-08-29 Thread Rushowr
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] Extension Ring on Multiple Phones
 
 This is a reply to a fairly old thread.
 
 My EXTEN string is meant to ring 3 phones (will increase to 12) thus:
 old: exten =_879677[67],1,Dial(SIP/120)  ; works fine
 new: exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124)
 
 I edit extensions.conf to the new line above, type 'reload' 
 into the CLI, see the new line with 'show dialplan' and 
 actually see the new line above, but when I dial the DID 
 879-6777 it rings on extension 120 only.
 
 Have I missed a step?
 
 Larry
 
 Jonathan k. Creasy wrote:
  EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE)
  
   
  
  
  
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Dave 
  Morrow
  Sent: Tuesday, November 08, 2005 1:51 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [Asterisk-Users] Extension Ring on Multiple Phones
  
   
  
  Hi all.  I wonder if anyone out there has a dial-plan which 
 will ring 
  an extension on multiple phones.
  
  David A. Morrow
 
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RE: [Asterisk-Users] Extension Ring on Multiple Phones

2006-08-29 Thread Rushowr
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: [Asterisk-Users] Extension Ring on Multiple Phones
 
 Color me puzzled.  What part of: exten = 
 _879677[67],1,Dial(SIP/120) should be deleted?
 
 Larry
 
 William Piper wrote:
  Sounds like you still have the old exten still there.
  Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120)
  
  bp
  
  On 8/29/06, Larry Alkoff [EMAIL PROTECTED] wrote:
 
  This is a reply to a fairly old thread.
 
  My EXTEN string is meant to ring 3 phones (will increase 
 to 12) thus:
  old: exten =_879677[67],1,Dial(SIP/120); works fine
  new: exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124)
 
  I edit extensions.conf to the new line above, type 
 'reload' into the 
  CLI, see the new line with 'show dialplan' and actually 
 see the new 
  line above, but when I dial the DID 879-6777 it rings on 
 extension 120 only.
 
  Have I missed a step?
 
  Larry
 
  Jonathan k. Creasy wrote:
   EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE)
  
  
  
   
  
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On 
 Behalf Of Dave 
   Morrow
   Sent: Tuesday, November 08, 2005 1:51 PM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: [Asterisk-Users] Extension Ring on Multiple Phones
  
  
  
   Hi all.  I wonder if anyone out there has a dial-plan which will
  ring an
   extension on multiple phones.
  
   David A. Morrow
 
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  --
  
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 --
 Larry Alkoff N2LA - Austin TX
 Using Thunderbird on Linux
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RE: [asterisk-users] Asterisk with PABX

2006-08-29 Thread Rushowr
*dunks email in bucket*

Heheh...Gee, ya think, Dean? Pardon my possession of an opinion. 

*Cautiously waits for next flame*

SKM

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Dean Collins
 Sent: Monday, August 28, 2006 9:38 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] Asterisk with PABX
 
 Hey if you don't need the work just say no, I'm sure someone 
 else will be happy to take the money from them.
 
 Maybe the monkey?
  
 
 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 List - Non-Commercial Discussion'
  Subject: RE: [asterisk-users] Asterisk with PABX
  
  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)
  
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On 
 Behalf Of Eric 
   ManxPower Wieling
   Sent: Monday, August 28, 2006 1:47 PM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [asterisk-users] Asterisk with PABX
  
   Dean Collins wrote:
Yes it is possible.
   
May I suggest you spend more time with 
 www.voip-info.org Or even 
better download www.trixbox.org on an old server to get an
   idea of how configs work.
  
   Getting Trixbox would help him understand how Trixbox 
 configs work, 
   not how Asterisk configs work.
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RE: [asterisk-users] Max number of SIP devices registered to anextension

2006-08-28 Thread Rushowr
-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 Discussion
Subject: [asterisk-users] Max number of SIP devices registered to
anextension


Is there a maximum number of SIP devices that can be registered to
an extension?

-brandon

-- 
Brandon Galbraith
Email: [EMAIL PROTECTED] 
AIM: brandong00
Voice: 630.400.6992
A true pirate starts drinking before the sun hits the yard-arm.
Ya. --thelost 

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Comment: ENCRYPTED WITH GPG

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AvXmh2VtjGAJPvixfnpwEWM=
=cyAY
-END PGP SIGNATURE-



To a single extension? 

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Brandon 
  GalbraithSent: Sunday, August 27, 2006 8:16 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
  [asterisk-users] Max number of SIP devices registered to 
  anextension
  Is there a maximum number of SIP devices that can be registered to 
  an extension?-brandon-- Brandon 
  GalbraithEmail: [EMAIL PROTECTED] 
  AIM: brandong00Voice: 630.400.6992"A true pirate starts 
  drinking before the sun hits the yard-arm. Ya. --thelost" 



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RE: [asterisk-users] Max number of SIP devices registered to anextension

2006-08-28 Thread Rushowr
-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 - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Max number of SIP devices 
 registered to anextension
 
 1
 
 On 8/27/06, Brandon Galbraith [EMAIL PROTECTED] wrote:
  Is there a maximum number of SIP devices that can be 
 registered to an 
  extension?
 
  -brandon
 
  --
  Brandon Galbraith
  Email: [EMAIL PROTECTED]
  AIM: brandong00
  Voice: 630.400.6992
  A true pirate starts drinking before the sun hits the 
 yard-arm. Ya.
  --thelost
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iD8DBQFE8uzVlfQsv7FBhp8RAhPAAJ9U0DSatH/FMxbCdosRnCuDB1zcagCZAYfO
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RE: [asterisk-users] RE: [asterisk-dev] Phone status

2006-08-28 Thread Rushowr



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: 
  [asterisk-users] RE: [asterisk-dev] Phone status
  
  Your are right, I dont have to invent the wheel again, and I'm getting 
  cleverer by looking at other peoples code.
  
  But this does not solve my problems, I have worked in the PABX business 
  as a software developer for about 8 years, and coming to * is not all that 
  easy. 
  
  For instance, * does not give you very good information of the state of 
  extensions (like we are used to in the "old-fashioned" PABX business), or 
  maybe I'm not good at finding the information.
  
  I'm trying to port an existing Windows application to *, its a dialer, 
  used to dial and se information about received calls.
  
  I know how to dial new calls, by using ORIGINATE on the AMI.
  I can receive some status information via the AMI, but consider this 
  example:
  
  I receive a call, which I accept. I get an event from 
  theAMI,that the call is now in the UP state.
  I receive another call, I get en event from the AMI, that the new call is 
  in the RINGING state.
  
  So far, so good.
  
  I now answer the other call (for instance by the line button on my 
  phone).
  Both calls are now in the UP state, who am I talking to?
  
  This, and many other questions, are currently making me even more thin 
  haired than normal :-)
  
  
  Michael
  2006/8/25, C F [EMAIL PROTECTED]: 
  So 
how about inventing a car? The auto industry is much more 
profitable.The point; there is no point in reinventing the wheel, 
why are you writing this from scratch?On 8/24/06, Mir [EMAIL PROTECTED] 
wrote: What do you mean? I'm not looking for 
someone elses work, I'm developing an application from  
scratch. Michael 2006/8/24, Andrew 
Kirch [EMAIL PROTECTED]: 
 Umm Flash operator 
panel? 
Andrew 
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] 
On Behalf Of Mir Sent: Thursday, August 24, 2006 2:18 PM 
 To: asterisk-users@lists.digium.com; 
asterisk-dev@lists.digium.comSubject: 
[asterisk-dev] Phone status  
Hi I'm working on a project, 
where I need the status of every telephone on the system. 
(Idle,ringing,busy) If a phone is busy, I also need 
to know the callerid of the other 
end. I have made a deamon, 
which query Asterisk every second for active calls, this works by 
issuing a "Status" to the manager-interface, and processing  the 
return data and then put the result into a MySQL 
table. The clients will 
query the MySQL table every second for the state of their phone, if 
there are no records with their numbers in it, they are considered  
idle. This works fine for 
calls from one SIP-phone to the other, this is for instance what it 
look like when extension 310 is connected to extension 
311: Event: Status 
Privilege: Call Channel: SIP/310-08697fb8 CallerID: 
310 CallerIDName: unknown Account: State: 
Up Link: SIP/311-0868fd98  Uniqueid: 
1156442804.74 Event: Status Privilege: 
Call Channel: SIP/311-0868fd98 CallerID: 311 
CallerIDName: Snom Account: State: Up Context: 
macro-vm  Extension: s Priority: 5 Seconds: 
13 Link: SIP/310-08697fb8 Uniqueid: 
1156442804.73 That is pretty easy to decode. 
However when an external call is made to a SIP-phone, the result is  
different, this is a call from another Asterisk via an IAX 
trunk: Event: Status Privilege: Call 
Channel: SIP/311-08695698 CallerID: 35254390 CallerIDName: 
unknown  Account: State: Up Link: 
IAX2/MR-1 Uniqueid: 1156442974.76 Event: 
Status Privilege: Call Channel: IAX2/MR-1 CallerID: 
35436121 CallerIDName: unknown  Account: 
State: Up Context: macro-vm Extension: s Priority: 
5 Seconds: 9 Link: SIP/311-08695698 Uniqueid: 
1156442974.75 The actual callerid of the caller is 3536121, 
35254390 is the called number.  How do I get the 
information, that 35436121 is connected to 311? Am I doing 
it in a stupid way, I'm aware that the Manager can give me realtime 
events, but I'm under the impression, that it is not very stable in  
a high traffic environment? Any help or good ideas would be 
appriceated. 
Michael 
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RE: [asterisk-users] Max number of SIP devices registered toanextension

2006-08-28 Thread Rushowr



Well, since you can technically only have one phone 
registered to an extension, you'll need to do a simultaneous ring setup in your 
dial:

Dial(SIP/1SIP/2SIP/3.)

I may be having a momentary brain freeze about the '' 
but I believe that's right...

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Brandon 
  GalbraithSent: Monday, August 28, 2006 11:35 AMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [asterisk-users] Max number of SIP devices registered 
  toanextension
  I'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 8/28/06, Rushowr 
  [EMAIL PROTECTED] 
  wrote:
  -BEGIN 
PGP SIGNED MESSAGE-Hash: SHA1Actually, 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-Commercial Discussion Subject: Re: 
[asterisk-users] Max number of SIP devices registered to 
anextension 1 On 8/27/06, Brandon Galbraith 
 [EMAIL PROTECTED] 
wrote:  Is there a maximum number of SIP devices that can 
be registered to an  extension?  
 -brandon--  Brandon 
Galbraith  Email: [EMAIL PROTECTED] 
 AIM: brandong00  Voice: 630.400.6992  "A true 
pirate starts drinking before the sun hits the yard-arm. 
Ya.  --thelost"  
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visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Brandon GalbraithEmail: [EMAIL PROTECTED]AIM: 
  brandong00Voice: 630.400.6992"A true pirate starts drinking before the 
  sun hits the yard-arm. Ya. --thelost" 
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RE: [asterisk-users] manual mods with GUI in place

2006-08-28 Thread Rushowr



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 28, 2006 2:54 PMTo: 
  'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
  [asterisk-users] manual mods with GUI in place
  
  
  This post spurred off of the 
  comment of Michael Collins on the Asterisk with PABX thread. I am going to 
  post the relevant information here:
  
  I started w/ AAH, then 
  went back and learned the dialplan apps, scripting, etc. For some guys 
  like me, it's easier to start with a working (if limited) system, and then 
  tinker with it, break it, etc.
  After breaking a few 
  systems I then went back and did a vanilla install to learn some more. I 
  ended up settling on a compromise: I load Trixbox and then make a bunch of 
  manual mods. I get the best of both worlds - a system that has all of 
  the prereqs loaded for me, plus a GUI for stuff that I don't want to do a cmd 
  line and also the power and flexibility of hand-editing my .conf files to get 
  exactly what I want out of the dialplan.
  
  For those wondering how to 
  get started, I can highly recommend STARTING with Trixbox, but definitely 
  don't STOP with Trixbox. After you play with a pre-installed, working 
  system, go out and get your hands dirty on a plain install. You'll be 
  better off for it in the long run. Having both GUI and cmd line 
  experience will make you a well-rounded Asterisk 
  user.
  
  -MC
  I started w/ AAH, then 
  went back and learned the dialplan apps, scripting, etc. For some guys 
  like me, it's easier to start with a working (if limited) system, and then 
  tinker with it, break it, etc.
  After breaking a few 
  systems I then went back and did a vanilla install to learn some more. I 
  ended up settling on a compromise: I load Trixbox and then make a bunch of 
  manual mods. I get the best of both worlds - a system that has all of 
  the prereqs loaded for me, plus a GUI for stuff that I don't want to do a cmd 
  line and also the power and flexibility of hand-editing my .conf files to get 
  exactly what I want out of the dialplan.
  
  For those wondering how to 
  get started, I can highly recommend STARTING with Trixbox, but definitely 
  don't STOP with Trixbox. After you play with a pre-installed, working 
  system, go out and get your hands dirty on a plain install. You'll be 
  better off for it in the long run. Having both GUI and cmd line 
  experience will make you a well-rounded Asterisk 
  user.
  
  -MC
  
  
  My question to everyone is 
  this..This is where I am at now. I have been using FreePBX for about a year, 
  after moving from [EMAIL PROTECTED] I am starting to need some manual changes and modules. 
  My question is can anyone point me in a direction on how to learn how to 
  create these. I read the ORiley book and thumbed though some of the others, 
  although I plan on reading them all the way through as time permits. I guess 
  my question is where do I add these things. I would still like to use FreePBX 
  because it just saves a ton of coding but I want to add my own things too. Do 
  I put them in the *_additional configs (which appear to be written over by 
  freePBX), the .conf files or the features.conf? Any web links with beginner 
  how tos or more info on this would be appreciated as 
  well!
  
  I didnt want to cross post 
  ;)
  
  Thanks
  
  Curt
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RE: [asterisk-users] Asterisk with PABX

2006-08-28 Thread Rushowr
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)

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Eric ManxPower Wieling
 Sent: Monday, August 28, 2006 1:47 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Asterisk with PABX
 
 Dean Collins wrote:
  Yes it is possible.
  
  May I suggest you spend more time with www.voip-info.org Or even 
  better download www.trixbox.org on an old server to get an 
 idea of how configs work.
 
 Getting Trixbox would help him understand how Trixbox configs 
 work, not how Asterisk configs work.
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RE: [asterisk-users] Call Max Time

2006-08-27 Thread Rushowr
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

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. 
 
Other than that, you might want to possibly check if your script is getting
the dialstring properly. Do you need to escape the / characters in it? What
I'd personally do is set up some Verbose() statements in my scripts to
output debugging data.
 
Hope this helps!




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Abdul
Sent: Sunday, August 27, 2006 2:54 AM
To: Asterisk-Users@lists.digium.com
Subject: RE: [asterisk-users] Call Max Time


Hello,

Could you tell how i can use it in PERL AGI script?

currently i am using in my AGI with this format, but some time call
is not disconnecting customers talking without money.

$dialstr = SIP/terminator/15745405022|350|tTL(653044:7000:5000);
$AGI-exec('Dial', $dialstr);

regards,








Get your own web address for just $1.99/1st yr
http://us.rd.yahoo.com/evt=43290/*http://smallbusiness.yahoo.com/domains .
We'll help. Yahoo! Small Business
http://us.rd.yahoo.com/evt=41244/*http://smallbusiness.yahoo.com/ . 

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WclJ5IBoFFF1NBdDb3P/oXM=
=9Jxz
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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. 


Other than that, you might want to possibly check if your 
script is getting the dialstring properly. Do you need to escape the / 
characters in it? What I'd personally do is set up some Verbose() statements in 
my scripts to output debugging data.

Hope this helps!

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  AbdulSent: Sunday, August 27, 2006 2:54 AMTo: 
  Asterisk-Users@lists.digium.comSubject: RE: [asterisk-users] Call 
  Max Time
  Hello,Could you tell how i can use it in PERL AGI 
  script?currently i am using in my AGI with this format, but some time 
  call is not disconnecting customers talking without money.$dialstr = 
  "SIP/terminator/15745405022|350|tTL(653044:7000:5000)";$AGI-exec('Dial', 
  $dialstr);regards,
  
  
  Get your own web 
  address for just $1.99/1st yr. We'll help. Yahoo! 
  Small Business. 


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RE: [asterisk-users] Call Max Time

2006-08-27 Thread Rushowr



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@lists.digium.comSubject: RE: [asterisk-users] Call 
  Max Time
  Hi,i am using the same calculating ((BALANCE / RATE) / 
  1000) method to return tTL.and i am sure my GAI is working well. but could 
  u tell me how i can set Verbose() sepecial for my 
  dialstring?Regards,
  
  
  Get your own web 
  address for just $1.99/1st yr. We'll help. Yahoo! 
  Small Business. 
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RE: [asterisk-users] Shared NFS or Shared MySQL for redundant secondaryserver?

2006-08-27 Thread Rushowr
-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 Of Christopher
Aloi
Sent: Sunday, August 27, 2006 5:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Shared NFS or Shared MySQL for redundant
secondaryserver?


Hey List!

What are your thoughts on redundancy?? 
Is it best to have the two Asterisk boxes share a /etc/asterisk
directroy; so if one falls out of service the other takes over or is it best
to have each node pull from a shared DB?? 

Cheers!
-- 
--
Christopher T Aloi
-- 

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=nhUW
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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 Of 
  Christopher AloiSent: Sunday, August 27, 2006 5:31 
  PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: [asterisk-users] Shared NFS or Shared MySQL for 
  redundant secondaryserver?
  Hey List!What are your thoughts on redundancy?? Is it 
  best to have the two Asterisk boxes share a /etc/asterisk directroy; so if one 
  falls out of service the other takes over or is it best to have each node pull 
  from a shared DB?? Cheers!-- --Christopher T 
  Aloi-- 


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RE: [asterisk-users] Call Max Time

2006-08-26 Thread Rushowr



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 Max 
  Time
  Hi All,Could anyone give me idea, How i can set Call Max 
  Time, so in pariticular time the call should disconnect 
  automatically.I will be appriciate for your helps.Abdul
  
  
  Get your email and more, right on the new 
  Yahoo.com 
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RE: [asterisk-users] Re: column width in CLI

2006-08-25 Thread Rushowr
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 'verbose' is specified, the output
includes
   more and longer fields.

Cheers
SKM 

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Sent: Wednesday, August 23, 2006 6:59 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: column width in CLI

Try show channels concise

--
--
Steven

http://www.glimasoutheast.org



Shaun Hofer [EMAIL PROTECTED] wrote in message 
news:[EMAIL PROTECTED]
 Hi,

 Can the column width for commands run in the Asterisk CLI be 
increased?
 Currently when I run 'show channels' I can't see the whole channels 
 id/name as its to long for the columns width and is cut off. 
I need to 
 grab a list of active channels, which is currently not do able.

 Thanks
 Shaun
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RE: [asterisk-users] Adding/Removing Prefixes

2006-08-25 Thread Rushowr

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 direction or 
provide an example diaplan that does this?

Many Thanks in Advance
SP
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RE: [asterisk-users] How to set externip in sip.conf automatically?

2006-08-25 Thread Rushowr
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



-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Larry Alkoff
Sent: Wednesday, August 23, 2006 9:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to set externip in sip.conf 
automatically?

As stated in the original post, when I entter the IP with an 
editor directly into sip.conf calls work just fine but I am 
looking for a way to have that done _automatically_.

The Asterisk - Future of Telephony book says it is possible 
for Asterisk to access a Linux environment variable containing 
the IP information in the form of ${ENV{variable}}.

It doesn't seem to work.  I am asking how to make it work.

Larry

Watkins, Bradley wrote:
 If you already have the IP in a file, why don't you set it up so the 
 file itself says:  externip=xx.xx.xx.xx and then do a #include in 
 sip.conf for the /etc/myip file?  I believe you'll have to do a sip 
 reload either way (which can obviously be part of your cron job) if 
 you're not already, but that should do what you're looking to do.
 
 - Brad
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Larry 
 Alkoff
 Sent: Tuesday, August 22, 2006 9:34 PM
 To: Asterisk-users; Austin-asterisk-users
 Subject: [asterisk-users] How to set externip in sip.conf 
automatically?
 
   I need to give Asterisk access to my external IP address 
to prevent 
 the NAT problem where caller cannot hear the callee's voice.
 
 According to Asterisk - The Future of Telephony page 92 Environment
 Variables:
 
Environment variables are a way of accessing Unix environment 
 variables from within Asterisk.  They are referenced in the form of
${ENV{var}}
 where var is the Unix environment variable you wish to reference.
 
 My external IP is placed each night in a file call /etc/myip and 
 placed in the $MYIP variable by /etc/bashrc when an shell is loaded.
 
 So I have /etc/myip refreshed each night in a cron job and when a 
 shell is opened /etc/bashrc does:
 export MYIP=`cat /etc/myip`
 
 To access the variable in sip.conf I have tried:
 
  externip=${ENV(EXTERNIP)}
 and
  ${ENV($EXTERNIP)}
 but neither seems to work.
 Is this the correct syntax?  Did I misinterpret the book?
 
 I say neither seems to work because When I hard code
 externip=69.91.84.176
 there are no NAT problems but when I try to access the $MYIP 
variable 
 either of the ways above NAT prevents me hearing the callee's voice.
 
 I have tried but not found a way to directly access the contents of 
 MYIP to the console using the CLI.  Is there a way to see or 
set _any_ 
 Linux enviromnent variable using the CLI?  More generally, how do I 
 access the Linux shell from the CLI?
 
 The problem with simply using
 externip=69.91.94.176
 is that number is subject to change and I don't know an easy way to 
 automatically write the value into sip.conf programatically.
 
 I could have just said how do I do this but wanted to show 
that I've 
 done my homework.
 Thanks for any help.
 
 Larry
 
 --
 Larry Alkoff N2LA - Austin TX
 Using Thunderbird on Linux
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--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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RE: [asterisk-users] MySQL CDR

2006-08-25 Thread Rushowr
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
Subject: [asterisk-users] MySQL CDR

Hi everyone,

I finished installing the Xorcom Rapid's Asterisk Packages 
with amportal (1.10.10), but i wasn't able to find the 
asterisk-mysql package. Any idea what happened there?, Is 
there another reposiitory for that package for asterisk 
1.0.11. Or could somebody send me the cdr_addon_mysql.so file?

Thanks for your responses,
--
Diego Quintana a.k.a. RouterMaN
Ingeniería de las Telecomunicaciones
PUCP
Linux Registered User #382615 - http://counter.li.org/ SIP # 
1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 
http://routerman.blogsome.com http://planeta.debianperu.org 
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RE: [asterisk-users] Trunk with multiple IPs?

2006-08-25 Thread Rushowr
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] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Benjamin Lawetz
Sent: Wednesday, August 23, 2006 9:42 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Trunk with multiple IPs?

Still no answers huh?

I've asked a couple of time how to do this, and by the lack of 
answers, I'm guessing there is no way.
The workaround unfortunately is to create an entry for each IP 
address in the range (I hope you don't have to open up a whole 
C class) 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of 
Warren (mailing lists)
Sent: August 22, 2006 3:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Trunk with multiple IPs?

How do I enter a trunk with multiple IPs.

xyz voip provider has 4 IPs and I want to allow incoming from any of
them: 1.1.1.1, 1.1.1.2, 1.1.1.3 and 1.1.1.4

Do I put 4 separate host= lines, do I put a single host=line 
that is comma separated or do I have to set up 4 separate 
incoming trunks?

TIA,
Warren

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RE: [asterisk-users] Setting the contact header on outbound INVITE

2006-08-25 Thread Rushowr



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 INVITE
  
  
  Is there anyway to set the Contact 
  header on outbound INVITEs such as there is for the REGISTER? I would also 
  like to be able to set the Contact header on 
  responses.
  
  Thanks,
  Michael
  
  
  

  This email may contain confidential information. If 
  you are not the intended recipient, please advise by return email and delete 
  immediately without reading or forwarding to others. -- Cbeyond 
  

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RE: [asterisk-users] Strange SIP response

2006-08-25 Thread Rushowr
Diego,

I've encountered this before, let me review a couple of old logs and notes
and I'll get back to regarding this.

Cheers,
SKM 

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Diego Andrés Asenjo González
Sent: Tuesday, 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 favorites
  


Yes, I have used it. The lines are extracted from a sip debug 
on the CLI. I'm going to paste more lines:

Sip read:
SIP/2.0 480 Temporarily Unavailable
To: sip:[EMAIL PROTECTED]:6198;tag=e4331437
From: 24307022sip:[EMAIL PROTECTED];tag=as288765a2
Via: SIP/2.0/UDP 
172.16.1.3:5060;branch=z9hG4bK73129cd1;received=172.16.1.3
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Content-Length: 0


7 headers, 0 lines
-- Got SIP response 480 Temporarily Unavailable back 
from 192.168.1.50
Transmitting:
ACK sip:[EMAIL PROTECTED]:6198 SIP/2.0
Via: SIP/2.0/UDP 172.16.1.3:5060;branch=z9hG4bK73129cd1
From: 24307022 sip:[EMAIL PROTECTED];tag=as288765a2
To: sip:[EMAIL PROTECTED]:6198;tag=e4331437
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.1.50:6198
-- SIP/EXT25-a454 is circuit-busy
  == Everyone is busy/congested at this time

I have not detected packet losses even.

Thanks for your response.

  

-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
Subject: [asterisk-users] Strange SIP response

Hi,

I am getting the following message on the CLI:

-- Got SIP response 480 Temporarily Unavailable back from 
192.168.1.60
-- SIP/EXT23-d910 is circuit-busy

and the call hangs up.

The peer is correctly registered and I'm not getting unavailable 
messages.

I really need help with this error.

--
MENSAJE ENVIADO CON WMAIL 1.01
UNIVERSIDAD DEL CAUCA


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RE: [asterisk-users] Help compiling asterisk-addons on Debian?

2006-08-25 Thread Rushowr



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 
  DiscussionSubject: [asterisk-users] Help compiling asterisk-addons 
  on Debian?
  Hello All -Running the following:Debian 
  StableAsterisk SVN-branch-1.2-r41069Checked out the following from 
  SVN:asterisk-addons/branches/1.2 When I attempt to compile 
  asterisk-addons I get the following: /usr/src/asterisk-addons$ 
  make/clipasterdev1:/usr/src/asterisk-addons# 
  make ./mkdep -fPIC -I../asterisk 
  -D_GNU_SOURCE `ls *.c`app_addon_sql_mysql.c:23:19: 
  mysql.h: No such file or directory cdr_addon_mysql.c:38:19: mysql.h: No 
  such file or directorycdr_addon_mysql.c:39:20: errmsg.h: No such file or 
  directoryres_config_mysql.c:53:19: mysql.h: No such file or 
  directoryres_config_mysql.c:54:27: mysql_version.h: No such file or 
  directory res_config_mysql.c:55:20: errmsg.h: No such file or 
  directorymake -C format_mp3 allmake[1]: Entering directory 
  `/usr/src/asterisk-addons/format_mp3'/clipIs this looking for 
  MySQL? I do have MySQL installed and running, a bit confused here anyone 
  have any thouhts? -- --Christopher T 
  Aloi-- 
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RE: [asterisk-users] SSH connection hangs on logout?

2006-08-24 Thread Rushowr
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Tzafrir Cohen
 Sent: Thursday, August 24, 2006 2:32 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [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 
  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 SSH terminal (using latest 
 version of 
  PuTTY) the session no longer closes it just hangs. I've 
 even changed 
  the Putty setting to close the window even on unclean exit but it 
  still hangs the connection... I had something similar once with 
  Zabbix a while back, but never Asterisk.
  
  Anyone else experience this?
  
  Start asterisk using  safe_asterisk or via asterisk -f
  
  I prefer the safe_asterisk shell script, since if asterisk seg 
  faults, there is a good chance asterisk will get 
 automatically restarted.
  
  Jeremy McNamara
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  You may need to redirect stdin, stdout, stderr like:
  
  run_asterisk\
  0/dev/null\
  1/dev/null\
  2/dev/null\
  
  
 
 In other words: 
 
 A plain 'asterisk' (without '-c' and such) that daemonizes 
 and does exactly that for you, among others.
 
 Asterisk is a daemon, rather than an interactive program. 
 Thus its handling for SIGHUP is to re-read configuration 
 rather than detach from the terminal.
 
 -- 
 Tzafrir Cohen sip:[EMAIL PROTECTED]
 icq#16849755  iax:[EMAIL PROTECTED]
 +972-50-7952406  jabber:[EMAIL PROTECTED]
 [EMAIL PROTECTED] http://www.xorcom.com
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Shoots out own flames before emailing

Gents, asstated in my email, I am using safe_asterisk. Additionally, even
when I started asterisk by hand, it was always forked off of my tty.
However, even if I DID have it connected to my tty, I'd have to issue stop
now before getting to the command prompt and being able to issue logout
to bash. 

Try this sometime gents, you'll see what I mean...issue a ! From the
*CLI...then type logout...You'll be told that you're not in a login shell
and to use exit.


Wow..
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.5 (MingW32)
Comment: ENCRYPTED WITH GPG

iD8DBQFE7cEuwWoA8HY7JXYRAqtoAJwNX8/L7OFuXvTPobOvJ8cH0Iei9QCaApf0
S4BH1uc4ZxWxei0gRy+qKy0=
=6PsA
-END PGP SIGNATURE-



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[asterisk-users] Quiet on the list today?

2006-08-24 Thread Rushowr
Just gotta check, I've never seen a complete day with no posts


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[asterisk-users] SSH connection hangs on logout?

2006-08-23 Thread Rushowr
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 SSH terminal (using latest version of PuTTY) the session no
longer closes it just hangs. I've even changed the Putty setting to close
the window even on unclean exit but it still hangs the connection... I had
something similar once with Zabbix a while back, but never Asterisk.

Anyone else experience this?

SKM


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RE: [asterisk-users] Strange SIP response

2006-08-22 Thread Rushowr
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
Subject: [asterisk-users] Strange SIP response

Hi,

I am getting the following message on the CLI:

-- Got SIP response 480 Temporarily Unavailable back from 
192.168.1.60
-- SIP/EXT23-d910 is circuit-busy

and the call hangs up.

The peer is correctly registered and I'm not getting 
unavailable messages.

I really need help with this error.

--
MENSAJE ENVIADO CON WMAIL 1.01
UNIVERSIDAD DEL CAUCA


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RE: [asterisk-users] Variable to show caller id for a current call?

2006-08-22 Thread Rushowr
Wait a minutewhy are you putting 227 into the CALLERID function? You
should read this:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid

The (number) portion is the argument to CALLERID telling it what to give
you, not what to insert/write 

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Warren (mailing lists)
Sent: Monday, August 21, 2006 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Variable to show caller id for a 
current call?

But how do you get that with GetVar?  I am trying to do this 
through the API.  I tried:
Action: GetVar
Variable CALLERID(227)

and 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

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
 Subject: [asterisk-users] Variable to show caller id for a 
 current call?

 Is there a variable that can be gotten with GetVar to show the 
 callerid of the current incoming call in progress at a sip 
extension?

 For instance, a caller from 516-922-9463 calls extension 234.  
 I would like to be able to be able to get back the 
 516-922-9463 if I pass 234.

 Also, can this be done while the extension is ringing?

 TIA,
 Warren
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RE: [asterisk-users] re-writing the dial plan - some hints please

2006-08-22 Thread Rushowr
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 Subroutines (end the subroutines with the Return command, and
call them with Gosub), or maybe even doing a little logic magic with GotoIf
statements? Of course, all of that logic routing can be a pain, but to
further alleviate my personal dialplan code pains, I use AEL2 which makes
the code a little more like Perl/C/PHP code, and allows for If/Else
statements, etc...

Just my 0.02.


SKM

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Ronald Wiplinger
Sent: Tuesday, August 22, 2006 8:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] re-writing the dial plan - some hints please

My dialplan grew over the last months and I want to 
restructure it. What hints do you have for me?

There are some points I want to do, but none of my tests worked.

I use realtime, and have there a field called key, which can 
have several flags.
E.g. a flag if the user is allowed to use a conference room, 
can call long distance, can call overseas, can call local 
pstn, different tariffs, 

I tried something like:
[test-key]
exten = _.,1,NoOp(variable key is ${key}) exten = 
_.,2,Set(flag_int =${CUT(key,,1)}) exten = 
_.,3,Set(tarif=${CUT(key,,2)}) exten = _.,4,NoOP(flag_int is 
${flag_int} and tarif is ${tarif})

and wanted to use this variables in the next context, by using 
include statments, but it did not work.

[caller]
include = test-key
include = A
include = B
...



The idea was to set at each entrance point first all flags and 
variables. Than I can use a common dialplan.
If a flag is set, than I could include another context. 
Unfortunately there is no IF()include. I might be able to set 
a jump in each context to the end if the flag is not set.

Any idea how I can do that?
Any ideas of structuring the dialplan more efficiently?

bye

Ronald


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RE: [asterisk-users] if command for or missing callerid?

2006-08-22 Thread Rushowr
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: [asterisk-users] if command for or missing callerid?

I am looking for a way to make a decission in the dialplan if 
I have a caller id or not.

What I want to do with it:

Call on the PSTN line should either use astcc.agi with the 
caller-id in place as card number, or asking for the calling 
card number.

How can I make this gotoif ???

bye

Ronald
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[asterisk-users] Setting RPID privacy?

2006-08-22 Thread Rushowr
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 header? Should I
just use CallingPres? Just set the CID to RESTRICTED ? 

Any hints, suggestions?

SKM


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RE: [asterisk-users] Setting RPID privacy?

2006-08-22 Thread Rushowr
Nevermind Gents and Ladies, I looked AGAIN at the dialplan command list and
found the one word I missed before when scanning
SetCallerPres...Independent as in Channel Independent. Please excuse my
error.

SKM

-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[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 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 header? 
Should I just use CallingPres? Just set the CID to RESTRICTED ? 

Any hints, suggestions?

SKM


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RE: [asterisk-users] Variable to show caller id for a current call?

2006-08-21 Thread Rushowr
${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 
current call?

Is there a variable that can be gotten with GetVar to show the 
callerid of the current incoming call in progress at a sip extension?

For instance, a caller from 516-922-9463 calls extension 234.  
I would like to be able to be able to get back the 
516-922-9463 if I pass 234.

Also, can this be done while the extension is ringing?

TIA,
Warren
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RE: [asterisk-users] Variable to show caller id for a current call?

2006-08-21 Thread Rushowr
Well, for one, you could set something like CID = ${CALLERID(number)} in the
dialplan, and then GetVar CID



-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Warren (mailing lists)
Sent: Monday, August 21, 2006 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Variable to show caller id for a 
current call?

But how do you get that with GetVar?  I am trying to do this 
through the API.  I tried:
Action: GetVar
Variable CALLERID(227)

and 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

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
 Subject: [asterisk-users] Variable to show caller id for a 
 current call?

 Is there a variable that can be gotten with GetVar to show the 
 callerid of the current incoming call in progress at a sip 
extension?

 For instance, a caller from 516-922-9463 calls extension 234.  
 I would like to be able to be able to get back the 
 516-922-9463 if I pass 234.

 Also, can this be done while the extension is ringing?

 TIA,
 Warren
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[asterisk-users] Quick, hopefully easy, question

2006-08-21 Thread Rushowr
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


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RE: [asterisk-users] Asterisk 'Hosting'

2006-08-19 Thread Rushowr
*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-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
McNamara
Sent: Wednesday, August 16, 2006 3:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 'Hosting'

Douglas Garstang wrote:
 Well, we're talking about several dozen, maybe 100, companies, per 
 Asterisk box here.


Ok - And the problem is?


Jeremy McNamara
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RE: [asterisk-users] Asterisk 'Hosting'

2006-08-19 Thread Rushowr
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] Asterisk 'Hosting'

*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-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of 
Jeremy McNamara
Sent: Wednesday, August 16, 2006 3:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 'Hosting'

Douglas Garstang wrote:
 Well, we're talking about several dozen, maybe 100, companies, per 
 Asterisk box here.


Ok - And the problem is?


Jeremy McNamara
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RE: [asterisk-users] Re: what is the real use of AEL?

2006-08-18 Thread Rushowr
-Original Message-
From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Barzilai
Sent: Friday, August 18, 2006 2:55 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: what is the real use of AEL?

Steve Murphy wrote:
...
[a lot of well-written arguments]
...
 And, pardon the shameless plug here, but for all you fence 
sitters, I 
 invite you to try AEL in/for your dialplans, and give me 
feedback! If 
 the majority of those 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 stand on this ;-) 

Rushowr


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RE: [asterisk-users] Dialplan or matching

2006-08-18 Thread Rushowr
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 - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialplan or matching

On 8/18/06, David Cook [EMAIL PROTECTED] wrote:
 Maybe I'm daft, but can asterisk to 'or' logic in dialplan matches 
 sort of like the SPA's can?

 Tollfree numbers for example. I can have a line for each combination:
 exten = _1800NXX, Dial, 
 exten = _1866NXX, Dial, 
 exten = _1877NXX, Dial, 
 exten = _1888NXX, Dial, 

 But I want to do is something like this:
 exten = _18[0678][0678]NXX, Dial, .

This syntax is valid and would work for what you're doing, but 
as you said, there is a chance of logic error in it.

 Or to prevent the logic error which albeit small, the above 
would create:
 exten = _18[00,66,77,88:2]NXX, Dial, ..
 (representing that the next 2 chars must equal any of '00'.'66','77' 
 or '88'

As for this syntax, Asterisk does not respect the [], so it parses the
66 as the priority.  I have no idea how to properly do this in 
one line if it is possible.


Kinsey
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RE: [asterisk-users] astbill white screen!!

2006-08-17 Thread Rushowr
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've installed asterisk and astbill according with all recommendation
(mysql5, drupal included with astbill, php, apache2...).
When I write http://server_adress/astbill, I get a white screen page.
Browser doesn´t give me an error page, it just a white screen page.

However asterisk doesnt have any problem, and works well with mysql. I also
have installed Drupal 4.7.3 linked to other database with other user and
password working well. And I have phpMyAdmin too. All working very good at
the same server.

I tried changing index.php to phpinfo.php in the same directory and it works
well too.

Can anybody help me with that please? Any suggestion will be very
appreciated.

Thanks, very much in advance

Sebastian




On 7/14/06, varun [EMAIL PROTECTED] wrote:
 Hello,

 Our asterisk server is on Centos 4.2

 We want to use Astbill.
 Astbill requires Drupal and mysql 5.

 I could not find rpms mysql5 for centos.

 We are getting mysql extensions issues because of php-mysql.

 How do we solve this ?

 Any other billing software that similar to Astbill ?

 Thanks

 Varun




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RE: [asterisk-users] Asterisk 'Hosting'

2006-08-17 Thread Rushowr
-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: Douglas Garstang
 Sent: Thursday, August 17, 2006 2:17 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [asterisk-users] Asterisk 'Hosting'

**snip**

  I spent 8+ hours a day, 5+ days per week for over 6 months thinking 
  how these functions fit within the realm of Asterisk. At every 
  single turn, after going down every single path, there where 
  limitations that forced us to backtrack and evaluate a different 
  approach. A script that could handle call routing, in conjection 
  with MySQL and stored procedures was the only way to implement our 
  requirements. The MySQL command had limitations, realtime was way 
  too resource intensive, unreliable and undocumented and so on. 
  Yep... i definitely haven't thought about this at all.
 
 Oops. I almost forgot intra-organisational 4 digit extension dialling. 
 Not just company, but organisational, where a company may have 
 multiple organisational units. It might be possible to hack together a 
 flat intra-business 4 digit extension dial lookup in the native 
 dialplan, but trying to make it a multi-level organisation lookup 
 would be pure hell... unless you farm the task out to a more advanced 
 scripting langauge like python, perl whatever.

I see the MySQL dial plan command still doesn't support stored procedures
either, 
unless you hack around with the source.

I've just recently come up against this limitation. Care to share info/code
concerning making stored procs work with the addon?


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MySQL Addon and MySQL5 Stored Procs (WAS: RE: [asterisk-users] Asterisk 'Hosting')

2006-08-17 Thread Rushowr
-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. I was trying to prove a point to
Jeremy. 
On the  voip wiki:

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MYSQL

under a comment titled 'Calling MySQL 5 stored procedures from app_mysql',
it looks 
like someone has managed to modify the source to get it to work.
I haven't tried it yet...

Doug.

Thanks for the point, I hadn't noticed that comment. I'll be implementing it
on one of my dev systems tonight and testing a multitude of stored
procedures that we had planned. If it works reliably, I'll be reporting it
as a bug in the digium bug tracker and submitting the modification as a
patch (with proper credit to the original poster of the mod of course).

S McGowan


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RE: [asterisk-users] Sending Email From A Dial Plan

2006-08-17 Thread Rushowr
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] Sending Email From A Dial Plan

Hi,

I'm looking for a simple way to send email from a dial plan. I have searched
around quite a bit looking for a solution for this and I'm surprised that I
haven't found anything useful yet other than using the
System() application. I would like to be able to change the subject
dynamically based on ${EXTEN} and the body is not important. I was hoping to
have a one line command from the System() application without having to
write a script or any other dependency. Has anyone implemented anything like
this?

Thanks,
Damien
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RE: [asterisk-users] Asterisk Real Time and sip.conf file used at

2006-08-17 Thread Rushowr
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 PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yusuf
Sent: Wednesday, August 16, 2006 3:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Real Time and sip.conf file used at


 On Wed, 2006-08-16 at 19:03 +0100, Ricardo Carvalho wrote:
 Is it possible to use Asterisk RealTime and also config files (like
 sip.conf) at the same time?
 As much as I know, only one thing can be used and I need them both 
 working!...


   Yes, you can use both at the same time.  The only restriction is
that 
 you cannot use the realtime static configuration and realtime 
 configuration.

 --

Hi

realtime static configuration and realtime configuration???
What is the difference, can you please explain?

thanks,
yusuf


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RE: [Asterisk-Users] OPENSER / SER and Asterisk

2006-08-17 Thread Rushowr
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, August 16, 2006 10:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] OPENSER / SER and Asterisk

*lol* The cryptic replies have been exactly my problem as well! 

 -Original Message-
 From: kjcsb [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, August 16, 2006 12:37 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk
 
 
 Absolutely. The SER/OpenSER documentation is terrible, and if you post 
 to the OpenSER mailing list, you get very cryptic replies.
 ___
 
 Whilst I would agree with you regarding SER, the documentation of 
 OpenSER is far better.
 
 Documentation of Asterisk Realtime on the other hand. Now
 *that's* terrible.
 
 Cameron
 
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RE: [asterisk-users] CallerID is not displaying for my incoming calls

2006-08-17 Thread Rushowr



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: [asterisk-users] CallerID is not displaying 
  for my incoming calls
  Hi,As you said, I have changed my configurations. But, 
  callerid is not displaying. What I have to do? Please tell 
  me.ThanksRegards,Chandra.Rich Adamson 
  [EMAIL PROTECTED] wrote:
  Crazy 
Boy wrote: Hi Friends,  We have installed Asterisk 
with Digium 04B card (4 FXO ports). Now, I  have connected my PSTN 
line directly to first port. I am making outgoing  calls and 
receiving incoming calls successfully through my Asterisk. The  
problem is: When I am receiving a call from outside (PSTN), I am not 
 getting the callerid number and getting callerid as "Asterisk" in 
my  softphones (XLite). Here I am giving my configuration 
files.  zaptel.conf file contents:  loadzone 
= us defaultzone=us fxsks=1-4  zapata.conf 
file contents:  [channels] context=incoming 
signalling=fxs_ks busydetect=1 busycount=7 
relaxdtmf=yes callwaiting=yes 
callwaitingcallerid=yes threewaycalling=yes 
cancallforward=yes echocancelwhenbridged=yes 
rxgain=0.0 txgain=0.0 callerid=asreceived 
language=en usecallerid=yes hidecallerid=no 
echocancel=yes transfer=yes immediate=no 
group=1 callgroup=9 pickupgroup=9 channel = 
1The above entries appear to be reasonable and correct. If you have 
not properly set rxgain and txgain, it "could" impact callerid. If those 
gains are too high/low, asterisk will not properly read the callerid 
data when sent to you. extensions.conf file 
contents:  [incoming] exten = s,1,Answer 
exten = s,2,SetMusicOnHold(default) exten = 
s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 exten 
= s,5,Background(/tmp/virg2) exten = s,6,Goto(s,1) 
include = leader Got event 18 (Ring Begin)... Aug 14 
14:11:58 WARNING[26744]: pbx.c:5869 pbx_builtin_dtimeout:  
DigitTimeout is deprecated, please use Set(TIMEOUT(digit)=timeout) 
instead. Aug 14 14:11:58 WARNING[26744]: pbx.c:5845 
pbx_builtin_rtimeout:  ResponseTimeout is deprecated, please use 
Set(TIMEOUT(response)=timeout)  instead.The above two 
WARNING statements are telling you that either you are copying those 
exten= statements from someone's old config files, or, you haven't 
read the asterisk documentation. The message is telling you that your 
statement "exten = s,3,DigitTimeout,5" should be replaced with the 
Set(TIMEOUT(digit)=timeout) syntax. Your statements are still executing 
properly today, but the next time you upgrade asterisk code, they are 
likely to fail due to the old syntax not being supported.Try 'show 
function TIMEOUT' from your CLI and read it. What I have to do 
to display the PSTN caller number on my softphones?  Please tell me. 
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RE: [asterisk-users] CallerID is not displaying for my incoming calls

2006-08-17 Thread Rushowr



Chandra,

Unfortunately, I can't help you too much, because I've not 
worked a lot with Zap. However, this message:

Aug 17 19:45:41 ERROR[10449]: callerid.c:276 callerid_feed: 
fsk_serie made mylen  0 (-8)

Seems interesting...My guess is that the callerid 
information is corrupted or something, because it's a negative value, not a 0 or 
positive. Possibly you have your CID Signalling set to the wrong value... One 
thing you could try just to get a better idea of what (if anything) is actually 
read from the callerid and what the presentation is set to, is to modify the 
your dialplan to output the data to your console (I use verbose 2 so I don't 
have to read the extra info:

[incoming]exten = s,1,Wait(4)exten = 
s,n,Answer
exten = s,n,Verbose(2|CallerID info received: 
${CALLERID(all)}) ; shows CID info
exten = 
s,n,Verbose(2|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 is helpful in 
some way...
Rushowr

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Crazy 
  BoySent: Friday, August 18, 2006 1:14 AMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [asterisk-users] CallerID is not displaying for my incoming 
  calls
  Hi Rushowr,Thank you for response.Here I am giving 
  my config files and error message. Please see it.zaptel.conf contents:loadzone = 
  usdefaultzone=usfxsks=1-4zapata.conf 
  contents:[channels]context=incomingsignalling=fxs_ksbusydetect=1busycount=7relaxdtmf=yescallwaiting=yescallwaitingcallerid=yesthreewaycalling=yescancallforward=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0callerid=asreceivedlanguage=enusecallerid=yeshidecallerid=noechocancel=yestransfer=yesimmediate=nomusiconhold=defaultringtimeout=8000cidsignalling=dtmfcidstart=ringgroup=1callgroup=1pickupgroup=1channel 
  = 1sip.conf 
  contents:[105]type=friendusername=105secret=ravicallerid="RaviKanth"host=dynamiccontext=leadercanreinvite=nonat=yesdtmfmode=rfc2833allow=allextensions.conf 
  contents:[incoming]exten = s,1,Wait(4)exten = 
  s,n,Answerexten = 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[leader]exten = 
  105,1,Dial(SIP/105,15)exten = 105,2,Voicemail(u105)exten = 
  105,3,Voicemail(b105)exten = 105,4,Hangupexten = 
  _9XX,1,Dial(Zap/1/${EXTEN:1}) ; Mobile phoneexten 
  = _5,1,Dial(Zap/1/${EXTEN:1})  ; 
  Local Landlineinclude = internal[internal]exten = 105, 
  1, Dial(SIP/105,15)When somebody calls from outside (Eg: mobile), I am 
  getting this below error message on Asterisk console:Error Message:Aug 17 19:45:41 
  ERROR[10449]: callerid.c:276 callerid_feed: fsk_serie made mylen  0 
  (-8)Aug 17 19:45:41 WARNING[10449]: chan_zap.c:6087 ss_thread: 
  CallerID feed failed: SuccessAug 17 19:45:41 WARNING[10449]: 
  chan_zap.c:6131 ss_thread: CallerID returned with error on channel 
  'Zap/1-1'Please tell me the solution. Looking forward to your kind 
  response. Thank you.Regards,Chandra.Rushowr 
  [EMAIL PROTECTED] wrote:
  

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: [asterisk-users] CallerID is not 
  displaying for my incoming calls
  Hi,As you said, I have changed my configurations. But, 
  callerid is not displaying. What I have to do? Please tell 
  me.ThanksRegards,Chandra.Rich Adamson 
  [EMAIL PROTECTED] wrote: 
  Crazy 
Boy wrote: Hi Friends,  We have installed 
Asterisk with Digium 04B card (4 FXO ports). Now, I  have 
connected my PSTN line directly to first port. I am making outgoing 
 calls and receiving incoming calls successfully through my 
Asterisk. The  problem is: When I am receiving a call from 
outside (PSTN), I am not  getting the callerid number and 
getting callerid as "Asterisk" in my  softphones (XLite). Here I 
am giving my configuration files.  zaptel.conf file 
contents:  loadzone = us defaultzone=us 
fxsks=1-4  zapata.conf file contents:  
[channels] context=incoming signalling=fxs_ks 
busydetect=1 busycount=7 relaxdtmf=yes 
callwaiting=yes callwaitingcallerid=yes 
threewaycalling=yes cancallforward=yes 
echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 
callerid=asreceived language=en usecallerid=yes 
hidecallerid=no echocancel=yes transfer=yes 

RE: [asterisk-users] Asterisk Real Time and sip.conf file used at thesame time

2006-08-16 Thread Rushowr
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, 2006 7:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Carlos Chavez
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Real Time and sip.conf file used at
thesame time

I guess my problem might be that, because I pretend Asterisk to use my
sip.conf static configuration file and also MySQL tables referenced in
extconfig.conf like this:

[settings]
sipusers = mysql,asterisk,sip
sippeers = mysql,asterisk,sip
voicemail = mysql,asterisk,voicemail

While I'm using one thing I can't use the other right???

Thanks once more,

Ricardo.






Quoting Carlos Chavez [EMAIL PROTECTED]:

 On Wed, 2006-08-16 at 19:03 +0100, Ricardo Carvalho wrote:
  Is it possible to use Asterisk RealTime and also config files (like
  sip.conf) at the same time?
  As much as I know, only one thing can be used and I need them both 
  working!...
 

   Yes, you can use both at the same time.  The only restriction is
that 
 you cannot use the realtime static configuration and realtime 
 configuration.

 --
 Carlos Chavez Prats
 Director de Tecnología
 Telecomunicaciones Abiertas de México S.A. de C.V.
 Tel: +52-55-91169161 Ext 2001




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RE: [asterisk-users] Recent additions to the DigiumAsterisk development team

2006-08-16 Thread Rushowr
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 a structure familiar to me and many others. Additionally, I
personally think it's a little more manageable and have had clients remark
on this as well after I show them comparable code from AEL2 and the
traditional Asterisk DP code.

As to your question about what can AEL do that an AGI/FastAGI app can't?
I'll name the number one and number two items on my list: 
AEL doesn't require calling an external application
AEL doesn't use an interpreted language such as PHP or Perl, both of
which seem to be the languages of choice for most of you cats out there
writing AGIs. 

AGI+Perl and AGI+PHP take a double performance hit compared to anything
written purely in the dialplan, because an interface has to be opened
between the two apps, control is passed back and forth, and the interpreter
has to run against the script. AEL2 is pre-parsed by the AEL2 compiler and
converted into Asterisk dialplan code when loaded into the system, not just
in time.

Anywho, that's just my two cents...

Sherwood McGowan
Consultant, AEL2 Zealot

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel
Hiver
Sent: Wednesday, August 16, 2006 8:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Recent additions to the DigiumAsterisk
development team


Steve Murphy joined our development team at the beginning of June. Steve
(murf on IRC/Mantis) had rewritten Asterisk's expression parser and the AEL
language parser as a volunteer community member, along with various other
bug fixes and improvements.
  

Which makes me think, what is the real use of AEL. I have taken a look at
it, and it makes asterisk's config file almost as unreadable as SER.

What exactly does AEL do that a well written AGI / FastAGI app doesn't?

I would think (but I'm surely wrong) that it would be better to do work on
having well defined APIs that allow us to script Asterisk (such as AGI and
the Manager interface) rather than invent Yet Another Pseudo Programming
Language - which is going to be an endless task... Don't you think?

That being said, just like the rest of the community, I'm very happy with
Kevin's exciting announcement!

Cheers,
Jean-Michel.
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RE: [asterisk-users] Problems with Hangup

2006-08-14 Thread Rushowr
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 Hangup

Hi,

my test-bed is :

sipphone -- Asterisk PBX -- PSTN -- Cell Phone

sipphone was able to setup a connection to Cell Phone. When sipphone hangs
up, Cell Phone also hangs up. However, when Cell Phone hangs up, sipphone
was not able to hang up.

Could it be that Asterisk was not able to recognise the hangup tone when the
Cell Phone hangs up ?

Does anyone know what the reason is ?

Thank you.

regards,
Kwang Mien


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RE: [asterisk-users] Problems with Hangup

2006-08-14 Thread Rushowr
My PSTN termination is through a provider, with a SIP connection between
myself and their systems. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp
Sent: Monday, August 14, 2006 6:55 AM
To: Asterisk Users Mailing List - Non-Commercial 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] Problems with Hangup


 I have to say that I'm experiencing the same issues, using the latest 
 SVN

You both need to enlighten us on what technology is in use to get to the
PSTN. If it is analog and disconnect supervision is not available, then
you'll be having fun.

 -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 Hangup
 
 Hi,
 
 my test-bed is :
 
 sipphone -- Asterisk PBX -- PSTN -- Cell Phone
 
 sipphone was able to setup a connection to Cell Phone. When sipphone 
 hangs up, Cell Phone also hangs up. However, when Cell Phone hangs up, 
 sipphone was not able to hang up.
 
 Could it be that Asterisk was not able to recognise the hangup tone 
 when the Cell Phone hangs up ?
 
 Does anyone know what the reason is ?
 

Joshua Colp
Digium
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RE: [asterisk-users] Macro inside macro

2006-08-14 Thread Rushowr
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
Groot
Sent: Monday, August 14, 2006 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Macro inside macro

Well I solved the problem, by just making it one macro, not a macro inside
another one.

[macro-record]
exten = s,1,Setvar(CALLFILENAME=CALL-${ARG1}-${MACRO_EXTEN:4}-$
{TIMESTAMP})
exten = s,2,Monitor(wav,${CALLFILENAME},m})
exten = s,3,setcallerid(${ARG2})
exten = s,4,dial(${ARG3})
exten = h,1,System(/etc/asterisk/mail.sh [EMAIL PROTECTED] $
{CALLFILENAME} )


Regards,
Attilla



On Aug 13, 2006, at 11:21 PM, Attilla De Groot wrote:

 On Aug 13, 2006, at 7:53 PM, Gonzalo Servat wrote:

 I think what you probably want is:

 exten = _*22*X.,1,Macro(record,conference,${EXTEN:4})

 If you have _*23*., it means it will match *23 as well as 
 *23*, but not *23*123456 which is probably what you 
 want. Try:

 exten = _*23*X.,1,Macro(record|dialout|31455200025|SIP/${EXTEN:4}
 @voipbuster)

 I made both changes, but that wasn't a problem. If I understand it 
 correctly, this only changes that the extension got more specific, 
 right ?



 Also, from memory, the h extension gets executed from the main
 context. After making the above changes, try adding this:

 exten = _*23*X.,h,System(/etc/asterisk/mail.sh [EMAIL PROTECTED]
 ${CALLFILENAME} )

 .. and remove the h extension from macro-record.

 Let me know if the above helps.

 I did make the changes, but it didn't solve the problem. It should  
 be  h,1,System by  the way ;)
 I have some log here:

  -- Executing Macro(SIP/attilla-0dce, record|dialout|31455200025| 
 SIP/[EMAIL PROTECTED]) in new stack
 -- Executing SetVar(SIP/attilla-0dce, CALLFILENAME=CALL- 
 dialout-08001234-20060813-231921) in new stack
 -- Executing Monitor(SIP/attilla-0dce, wav|CALL- 
 dialout-08001234-20060813-231921|m}) in new stack
 -- Executing GotoIf(SIP/attilla-0dce, 0?macro-record|s| 
 4:macro-record|s|5) in new stack
 -- Goto (macro-record,s,5)
 -- Executing Macro(SIP/attilla-0dce, dialout|31455200025|SIP/ 
 [EMAIL PROTECTED]) in new stack
 -- Executing SetCallerID(SIP/attilla-0dce, 31455200025) in  
 new stack
 -- Executing Dial(SIP/attilla-0dce, SIP/ 
 [EMAIL PROTECTED]) in new stack
 -- Called [EMAIL PROTECTED]
 -- SIP/sip1.voipbuster.com-f715 is making progress passing it  
 to SIP/attilla-0dce
 -- SIP/sip1.voipbuster.com-f715 answered SIP/attilla-0dce
   == Spawn extension (macro-dialout, s, 2) exited non-zero on 'SIP/ 
 attilla-0dce' in macro 'dialout'
   == Spawn extension (macro-dialout, s, 2) exited non-zero on 'SIP/ 
 attilla-0dce' in macro 'record'
   == Spawn extension (macro-dialout, s, 2) exited non-zero on 'SIP/ 
 attilla-0dce'

 As you can see it executes everything perfectly and I was expecting  
 that after this the script would be executed.


 Regards,
 Attilla


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RE: [asterisk-users] Macro inside macro

2006-08-14 Thread Rushowr
I know I could do that, and I do for some instances, but the biggest point
of writing the macros (at least in my case) is to reduce typing etc... 

Murf and I have uncovered a bug relating to this. Macros can call macros
just fine pre-dial, but once there's been a hangup, we've discovered that
Asterisk no longer processes the macros properly. A bug has been submitted
to the digium bug system.

Cheers,
Sherwood McGowan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Monday, August 14, 2006 1:26 PM
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 Wieling wrote:
 
 Rushowr wrote:
 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.

 pbx-1*CLI show application gosub
 pbx-1*CLI
   -= Info about application 'Gosub' =-

 [Synopsis]
 Jump to label, saving return address

 [Description]
 Gosub([[context|]exten|]priority)
   Jumps to the label specified, saving the return address.

 pbx-1*CLI
 
 Already considered this option, but I want to give it some arguments. 
 And that isn't possible with gosub.

--
Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga,
and Montgomery.
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RE: [asterisk-users] Asterisk and PHP?

2006-08-14 Thread Rushowr



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", Joachim Vanheuverzwijn/Zoa).

If I'm mistaken, somebody 
please correct me :)

I'm mistaken, please, 


  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Lennie De 
  VilliersSent: Monday, August 14, 2006 5:09 PMTo: 
  asterisk-users@lists.digium.comSubject: [asterisk-users] Asterisk 
  and PHP?
  
  

  
Hi,

I found:

http://www.voip-info.org/wiki/view/Asterisk+AGI+php

Is there more such examples or tutorials available that show me how 
to control Asterisk using PHP?
I want to be able to have full control over Asterisk using 
PHP.

For example:

* Execute PHP code when there's an incoming call.
* Handle or control the incoming call via PHP code.
* Execute PHP code when done with incoming call.
* Etc...

Kind Regards,

Lennie De Villiers


  

  
  



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RE: [asterisk-users] In CDR record not what I want

2006-08-11 Thread Rushowr
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-users] In CDR record not what I want

Hi,

I have the following rules:
exten = 4441,1,NoOp(--- ${CALLERID} calling on capi-extern (${EXTEN}) ---)
exten = 4441,2,Goto(dialin-privat,s,1) exten = 4441,3,Hangup

[dialin-privat]
; Log incoming calls
exten = s,1,LDAPget(CALLERIDNAME=daheim) exten =
s,2,NoOP(--CALLERID=-${CALLERID}-, CALLERIDNUM=-${CALLERIDNUM}-,
EXTEN=-${EXTEN}--) ...

my CDR records says now that a call from unkown to s happened.
Is it possible that in the CDR record the number which has been called is
saved and not s?
e.g.
number unkown called 4441


Best regards,
Matthias

-- 

Programming today is a race between software engineers striving to build
bigger and better idiot-proof programs, and the universe trying to produce
bigger and better idiots. So far, the universe is winning. -- Rich Cook
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RE: [asterisk-users] Auto retry on Busy

2006-08-11 Thread Rushowr
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 McGowan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Smith
Sent: Friday, August 11, 2006 2:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Auto retry on Busy

Why don't you just test for the dial status after the dial command
completes? I don't really see why you want something to keep dialing until
it gets through, but this would work.

[something]
1,1,Dial(zap/,sip/, etc/whatever, 10)
1,n,gotoif[${DIALSTATUS}=BUSY]?(LINEBUSY):(OTHER)
1,n(LINEBUSY), Wait(30)
1,n,goto(something,1,1)
1,n(OTHER), do something else

Sure it is pretty rough, but the basics are there. Also you might want to
read this: http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS

Kevin



Noah Silverman wrote:
 Hi,

 Does anybody have an easy solution for this.

 I want something that will keep trying a busy number every 30 seconds 
 until it gets through.

 I've tried retrydial, but can't get it to work.

 Any suggestions?

 Thanks,

 -N
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RE: [asterisk-users] Auto retry on Busy

2006-08-11 Thread Rushowr
Yep, my point exactly.

Since I'm in the middle of another ITSP project, I'll be hitting this again,
and will share anything I come up with. I've had thoughts, but never tested
it.


SHerwood 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Novack
Sent: Friday, August 11, 2006 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Auto retry on Busy

Also many so-called legacy hybrid PBX switches have had this for many a
year Hard to compete when well used features that have 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 not more
features to offer than the POTS and Mobile guys.

 Cheers,
 Rushowr - Sherwood McGowan

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Kevin 
 Smith
 Sent: Friday, August 11, 2006 2:30 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Auto retry on Busy

 Why don't you just test for the dial status after the dial command 
 completes? I don't really see why you want something to keep dialing 
 until it gets through, but this would work.

 [something]
 1,1,Dial(zap/,sip/, etc/whatever, 10)
 1,n,gotoif[${DIALSTATUS}=BUSY]?(LINEBUSY):(OTHER)
 1,n(LINEBUSY), Wait(30)
 1,n,goto(something,1,1)
 1,n(OTHER), do something else

 Sure it is pretty rough, but the basics are there. Also you might want 
 to read this: 
 http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS

 Kevin



 Noah Silverman wrote:
   
 Hi,

 Does anybody have an easy solution for this.

 I want something that will keep trying a busy number every 30 seconds 
 until it gets through.

 I've tried retrydial, but can't get it to work.

 Any suggestions?

 Thanks,

 -N
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RE: [asterisk-users] Inbound Calls SIP/2.0 404 Not Found

2006-08-11 Thread Rushowr
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


Sherwood 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Friday, August 11, 2006 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inbound Calls  SIP/2.0 404 Not Found

You might want to take a look at 'sip debug' to see what your provider is
actually sending you. Its likely they aren't sending you the
9495551212 sting as you are expecting.


 Thanks -
 
 Just to be clear - I just replaced the real digits with  - I want 
 to direct these to specific extensions. So maybe I should have used 
  or something else?
 
 I tried this:
 
 exten=_9495551212,1, Goto(mainmenu,s,1)
 
 But still to no avail.
 
 On 8/11/06, Vadim Berezniker [EMAIL PROTECTED] wrote:
 Perhaps the context in sip.conf doesn't match the context in the dial 
 plan.

 


 I'm trying to get inbound DIDs working via SIP.

 I have 20 DIDs coming in via a single SIP profile in sip.conf.

 I was hoping to have these matched in extensions.conf, so I have 
 setup lines like this:

 exten=949271,1, Goto(mainmenu,s,1)

 Unfortunately these aren't getting matched and I'm getting this error:

 Looking for s in druid-default (domain 949271) SIP/2.0 404 Not 
 Found

 Any hints or tips?

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RE: [asterisk-users] Realtime SIP Authentication

2006-08-10 Thread Rushowr



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 and I looking to 
  find a way to be able to authenticate users based on both the username and IP 
  of the incoming call the reason being I have different users connecting from 
  same IP but using different usernames.I have read that setting 
  type=peer is only matched on IP address/port.Is it possible to 
  configure Realtime to match on username and IP? Ron.
  
  
  Check Out the new free AIM(R) Mail -- 2 GB of storage 
  and industry-leading spam and email virus 
protection.
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