Re: [asterisk-users] AGI - Getting the passed parameters

[yusuf]

Mike D'Ambrogia wrote:

Need to figure out how to grab the passed variable in my php AGI script

I pass it in via the Dialplan like this:

exten => 420,1,Answer
exten => 420,n,DigitTimeout(5)
exten => 420,n,ResponseTimeout(10)
exten => 420,n,Flite("enter the one digit code")
exten => 420,n,Read(CODE,beep,1)
exten => 420,n,AGI(yy.php|${CODE})

Inside of yy.php how would I reference ${CODE} to expose it??  It
doesn't seem to come in with the standard variables that asterisk passes
to the AGI, at least the debugging loop that I have writing to log file
doesn't expose it as part of the std variables

mike


Hi,

in your AGI use: GET VARIABLE CODE

thanks,
Yusuf

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Re: [asterisk-users] No CDR from Outbound Call

[William Piper]

Try adding a forkcdr in just before your dial command.

bp


On 1/8/07, Peder @ NetworkOblivion <[EMAIL PROTECTED]> wrote:


I have a little call recording script that I am running and it works
fine, but I have one problem.  I get CDR when a user calls into the
extension, but I do not get CDR for the call that it makes outbound on #
17.  Any idea why?  Here is the extensions info:

[default]
exten => 2211,1,Answer
exten => 2211,2,Wait(1)
exten => 2211,3,Playback(/etc/asterisk/recording/getshop)
exten => 2211,4,playback(beep)
exten => 2211,5,Read(shopid)
exten => 2211,6,AGI,getnumber.agi|${shopid}
exten => 2211,7,Noop,${shopid}
exten => 2211,8,GotoIf($[${SHOPPHONE} = 1]?20:9)
exten => 2211,9,Noop,${SHOPPHONE}
exten => 2211,10,GotoIf($[${SHOPPHONE} = 2]?22:11)
exten => 2211,11,Noop,${SHOPNO}
exten => 2211,12,GotoIf($[${SHOPPHONE} = 3]?24:13)
exten => 2211,13,SetVar(CALLFILENAME=${SHOPNO}-${TIMESTAMP})
exten => 2211,14,AGI,startlog.agi|${SHOPPHONE}|${CALLFILENAME}
exten => 2211,15,SetCallerPres(prohib)
exten => 2211,15,SetCIDNum(2211)
exten => 2211,16,Monitor(wav,${CALLFILENAME},m)
exten => 2211,17,Dial(SIP/[EMAIL PROTECTED])
exten => 2211,18,wait(2)
exten => 2211,19,hangup
exten => 2211,20,playback(/etc/asterisk/recording/problem)
exten => 2211,21,goto(default,2211,2)
exten => 2211,22,playback(/etc/asterisk/recording/invalid)
exten => 2211,23,goto(default,2211,2)
exten => 2211,24,playback(/etc/asterisk/recording/syserror)
exten => 2211,25,goto(default,2211,2)

How it works is that a user calls in and enters a code.  It then does a
database lookup of the code to find a number to call.  It calls the
number and then bridges the two ends together, records it and mixes it
to an mp3.  As I said, it works fine and has for over a year.  The only
issue is that I don't get outbound CDR for some reason and I don't know
why

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RE: [asterisk-users] AGI - Getting the passed parameters

[Yuan LIU]

From: "Mike D'Ambrogia" <[EMAIL PROTECTED]>

Need to figure out how to grab the passed variable in my php AGI script

I pass it in via the Dialplan like this:

exten => 420,1,Answer
exten => 420,n,DigitTimeout(5)
exten => 420,n,ResponseTimeout(10)
exten => 420,n,Flite("enter the one digit code")
exten => 420,n,Read(CODE,beep,1)
exten => 420,n,AGI(yy.php|${CODE})

Inside of yy.php how would I reference ${CODE} to expose it??  It
doesn't seem to come in with the standard variables that asterisk passes
to the AGI, at least the debugging loop that I have writing to log file
doesn't expose it as part of the std variables


If you read the documentation, it says that parameters are passed like 
command line arguments.  So it's in the array that contains these arguments. 
 In fact, you can debug your AGI using command line.


Yuan Liu


mike



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[asterisk-users] AGI - Getting the passed parameters

[Mike D'Ambrogia]

Need to figure out how to grab the passed variable in my php AGI script

I pass it in via the Dialplan like this:

exten => 420,1,Answer
exten => 420,n,DigitTimeout(5)
exten => 420,n,ResponseTimeout(10)
exten => 420,n,Flite("enter the one digit code")
exten => 420,n,Read(CODE,beep,1)
exten => 420,n,AGI(yy.php|${CODE})

Inside of yy.php how would I reference ${CODE} to expose it??  It
doesn't seem to come in with the standard variables that asterisk passes
to the AGI, at least the debugging loop that I have writing to log file
doesn't expose it as part of the std variables

mike

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Re: [asterisk-users] Re: Best inexpensive home office router forVoIP (QoS with maybe PoE)

[Brian Capouch]

Jon Pounder wrote:


Buzzwrong answer!  Don't answer on things you have no idea.  and 
stop providing bad information.



you should take your own advice  - an acre is 200ft x 200ft - what idiot 
would
pay a consultant $7000 to tell them they need one access point in the 
middle.




This is getting ridiculous.

As a farmer's son, I must report that your acre is a bit too small. 
Perhaps you meant *approximately* 200x200?


A square acre is 208.710325571 feet on a side, or 43,560 square feet.

B.

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Re: [asterisk-users] RTP directly

[Benjamin Jacob]

Davida,
You would also want to look at canreinvite option in sip.conf
http://www.voip-info.org/wiki-Asterisk+SIP+canreinvite

cheerz
- Ben.


Eric "ManxPower" Wieling wrote:


David Alcott wrote:



Is there a way to configure the Asterisk so that the RTP goes 
directly between the Endpoints as opposed to going through the asterisk?



That is the default if Asterisk believes it will work.  Things that 
might not make it work is tTwW options to Dial, protocol transation 
(one leg is SIP, the other is IAX2, transcoding, NAT, or many other 
things that make the two legs of the call not compatible with reinvites.

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Re: [asterisk-users] Digium TE407P vs. Sangoma A104d

[Eric \"ManxPower\" Wieling]

Christopher Snell wrote:

Sorry for the old thread revival...I bought three Sangoma A104 cards
to use as T1 (not PRI) data cards in an OpenBSD router.  I was
disappointed to find out that trunking is not supported with this
configuration.  I contacted Sangoma and was told that they would look
into it but I haven't heard back from them since.  Sangoma has chapped
my ass a bit because of this.  I'm sitting on $4500 of useless
hardware.  Anybody want to by some A104s?  ;)

My advice: go with Digium.


I think that is more an issue with the OpenBSD drivers for Sangoma.  I 
use "trunking", i.e. Channelized voice T-1, with Sangoma with Asterisk 
on Linix for the corporate HQ Asterisk server.

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Re: [asterisk-users] Re: Is there a low cost cell phone base station for asterisk ?

[Eric \"ManxPower\" Wieling]

M.Hockings wrote:

Dumpolid Exeplish wrote:

It is true what Eric and Steve have said, you do need a licensed GSM
frequency to operate and sell GSM services (even for rural areas).
however, this link might be of interest to you

http://rfdesign.com/mag/radio_field_trials_allsoftware/



That is more what I was thinking of but it is still a cell provider type 
of hardware.  In my mind I was thinking of something very low powered 
and turning off the roaming, etc on the phone so they only work with the 
one base.   Think single cell base-station transceiver that can talk to 
a cell phone and turn it into a sip conversation to Asterisk.  Here in 
Canada, and back years ago, when I worked with radio I think the law was 
something like less than 100mw of input power didn't require a license. 
 However, with the advent of cell phones that could very well not be the 
case in those bands.  But one never knows...


In any case I'll probably lean towards something like the Engenius 
wireless phones.


http://www.engeniustech.com/telecom/products/details.aspx?id=107


Those things are somewhere around US$500

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Re: [asterisk-users] Random dropped calls...

[Carlos Chavez]

On Wed, 10 Jan 2007 21:49:50 +0200, Tzafrir Cohen wrote

> 
> Those are 6 channels of the 9?
> 
> What is the configuration of the other three?
> 
> What is the configuration of the 16 Astribank channels?
> 
> You don't set busycount. This uses the default value (3?). Can you 
> try setting it to a higher value?
> 

 The other three channels are used for a GSM adapter so they are not
included here and they do not seem to have the same problem.  I do not think
the Astribanks may be the problem because the problem also happens when
calling out from a SIP phone.

 I will try busycount but why would that matter?  The calls are dropped
and you never hear a busy tone.

--
Carlos Chavez
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] Re: Is there a low cost cell phone base station for asterisk ?

[Time Bandit]

The phone in question just prepended 010whatever to ALL phone numbers
dialled, which makes it pretty crappy to use with a line that does not
allow for network selection codes, or on lines that need a "0" for a
POTS line.

You could use it as an extension on Asterisk and strip that pre-pended
number from your dialed number.

Nice way to screw their attempt to screw you :)
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Re: [asterisk-users] Directory too difficult?

[Paul]

Leo Ann Boon wrote:

> Colin,
>
> Thanks mate for the first laugh of the day.
>
> Colin Anderson wrote:
>
>> I got a requirement list just now, with my comments inline: (showing
>> it just for a giggle)
>>  
>> User requirement: 1) Directory set up by name - If person calling
>> does not know employee's name, how will they access?
>>  
>> -Why, using app_telepathy.so of course!
>
> Is app_telepathy GPL'd (General Psychic License) or do we need a bunch
> of mind reading license lawyers. ^_^
>
>>  
>> User requirement: 3) Not all mobile phones have the albphabet on
>> their dialpads, how do they access our directory?
>>  
>> -Shout really loud. Telus should have a class action against it
>> for selling Razrs with no DTMF.
>
> Wow, didn't know there are pulse dialed mobile phone. ^_^
>
> What about speech recognition option for directory? Not necessary a
> full speech to text engine, just enough to recognize the names would do.
>
Yes let's use a very limited vocabulary engine to make it simple:

Ask - Does the first name begin with the letter A? Please answer yes or no.

If yes ask - Does the first name begin with the letters A B? Please
answer yes or no.

If no ask - Does the first name begin with the letter B? Please answer
yes or no.

And so on until you get a match or they just go away.

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Re: [asterisk-users] API: how to bridge originated call?

[chester c young]

Moises

what you've done here looks great, but some examples or a little doc
would be really helpful - following the bug report, although
informative, is a very difficult way to extract specs.

in my case I want a user to be on-line all the time - the system will
dial and connect them and, when they're done, they select the next one.
 what I'm doing now is putting them into a loop with a g-option on the
dial.  the number it dials is set thru the api.  if the number's not
set it waits one second and loops again.

from my limited knowledge using a Bridge() function is much more
elegant, but I am in the dark as to what the context of the user, what
happens on hangup (can it fall thru?), etc.

maybe after this demo is done I'll solve this correctly using Bridge,
but alas little time now for experimentations.

thanks
cy



--- Moises Silva <[EMAIL PROTECTED]> wrote:

> I have uploaded a working patch for version 1.2.12.1, and other that
> seems to work in Trunk, but few people is reporting results, you can
> help to get this into Asterisk, go here:
> 
> http://bugs.digium.com/view.php?id=5841
> 
> The patch I ported to 1.2.12.1 is working fine, I have tested in my
> servers, is the one called "bridge-1.2.12.1.patch", there are other
> ones that say trunk, obviously only work with the trunk version of
> Asterisk.
> 
> Kind Regards
> 
> On 1/3/07, chester c young <[EMAIL PROTECTED]> wrote:
> > (my pstn calls are coming in thru an upstream asterisk server, so
> the
> > called and calling phone number is passed as an extension.)
> >
> > when caller comes in on 555, he will go to extension 1234 where
> he
> > will wait for the API to make a call to 999 for him.  how do I
> > bridge the two calls?
> >
> > extensions.conf:
> >
> > ;context where caller comes in
> > [caller]
> > 555,s,1 Answer()
> > 555,s,n UserEvent(Init) ;this lets me know the connection for
> > 555
> > 555,1234,1 Noop(caller waits to be bridged)
> > 555,1234,2 Background(soothingmusic)
> >
> > ;context for connection - is this needed?
> > [connect]
> >
> >
> > from the API:
> >
> > (do I need to create a new context/extension first?)
> >
> > Action: Originate
> > Channel: IAX2/upstream/999  <-- calls 999222 thru upsteam IAX
> > Context: ??
> > Exten: ??
> > Priority: ??
> >
> >
> >
> > __
> > Do You Yahoo!?
> > Tired of spam?  Yahoo! Mail has the best spam protection around
> > http://mail.yahoo.com
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> >
> 
> 
> -- 
> "Su nombre es GNU/Linux, no solamente Linux, mas info en
> http://www.gnu.org";
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Yahoo! Music Unlimited
Access over 1 million songs.
http://music.yahoo.com/unlimited
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Re: [asterisk-users] Digium TE407P vs. Sangoma A104d

[Christopher Snell]

Sorry for the old thread revival...I bought three Sangoma A104 cards
to use as T1 (not PRI) data cards in an OpenBSD router.  I was
disappointed to find out that trunking is not supported with this
configuration.  I contacted Sangoma and was told that they would look
into it but I haven't heard back from them since.  Sangoma has chapped
my ass a bit because of this.  I'm sitting on $4500 of useless
hardware.  Anybody want to by some A104s?  ;)

My advice: go with Digium.

Chris

On 12/4/06, Michael Collins <[EMAIL PROTECTED]> wrote:





Has anyone had experience with one or both of these cards?  I'm in a
position where I might need to recommend one over the other.  I've read
everything that I can find online, so now I'd like to hear of personal
experiences.  Everything I read on both cards is "5 stars! Awesome! It
Rocks!"  They both seem to have similar capabilities, similar pricing, etc.



Could those of you who have seen these in action please give us some
feedback?  I'm interested in anything that might help me decide, be it
warranty info, vendor responsiveness, etc.



Thanks!



-MC
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[asterisk-users] Extensions in a macro

[Yuan LIU]

I have a question regarding how a macro interacts with the "main" call flow.

In my observation, if I Goto an extension (without explicitly referencing 
context) that exists within a marco, call flows to that extension.  But if I 
Goto an extension that does not, presumably invalid, instead of flowing to 
the "i" extension inside the macro, the call seems to flow back to 
${MACRO_CONTEXT}.  In this manner, a macro is not a container like a 
subroutine; but instead, it's like an "include" that simply extends 
available extensions, except that it has variable substitution.  However, 
it's not totally like an "include" because it can have extensions that the 
calling context already has (such as "s").


Can someone enlighten me?

Yuan Liu


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[asterisk-users] Priority + 101 exit

[Yuan LIU]

Many built-in applications jump to current priority + 101 upon unfavourable 
output.  Does this apply to macros and AGI's also?  Precisely, when an AGI 
returns non-zero, will Asterisk automatically take the call to +101?  And 
when a macro hits an invalid extension, will Asterisk automatically take the 
call to +101? (So far the "i" extension could not be reached from within a 
macro even with Goto, but that's another question.)


Also, what's behind this magic number 101?  Any tradition or arbitrary? 
(Some apps take 51 and 101 as two conditional exits.)


Yuan Liu


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Re: [asterisk-users] how to realize chief - secretary (or Manager - Assistant) setup with Asterisk?

[Rosli Sukri]

check this out,

http://snom.com/wiki/index.php/Snom300/Web_Interface/Function_Keys#Dialog_state_.26_call_pickup

we are using this snom feature on our box, while enabling hints for the
extensions.. So basically after rebooting the snom the extension monitored
(i,e managers ext) if he is on the phone the LED will light up. If his
extension is ringing or someone is calling his extension the LED will blink.

On 1/11/07, Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote:


Am Mittwoch, den 10.01.2007, 16:37 +0100 schrieb Michael Hamann:
> Hello,
>
> we are running a Asterisk (1.2) installation with about 80 snom phones
> (300,320,360).
>
> Now have the demand for a special manager - assistant setup for a few
> extensions.
>
> Since Shared Line Appearance is not available in 1.2 I´m wondering how
> to realize this...
>
> What we need is that the manager can decide whether he wants to get
> calls or not. If not he must have the possibility to redirect all
> incoming calls to his secretary. The secretary itself answers all calls
> and decides if the call is important enough to disturb the manager. If
> so she/he transfers the call to the manager. So the secretary can filter
> the calls for the manager...
>
> The only way I can imagine so far is via a redirect by AstDB on the
> manager extension. The managers phone has two different lines - the
> official and a secret one only the secretary uses...
>
> Or are there any other solutions?
>
> Any hint will be appreciated ...

Hello Michael,

as I see it, the most obvious setup would be

- have SIP accounts, e.g. sip123 for the secretary phone, sip456 and
sip789 for the manager phone.
- the "official"/"public" extension number for the manager might be
"4321", so

exten => 4321,1,Dial(SIP/sip123&SIP/sip456)

would ring both the secretary phone and the manager phone on the
"public" id (which most probably can have a separate ringtone than the
"private" id). You would also want a "private" extension like

exten => 4901,1,Dial(SIP/sip789)

for the secretary to reach the manager. A few thoughts:
- The Callerid setting for both secretary and chief should be "4321", no
matter which line the chief chooses to call out through.
- Do not choose an obvious private number, like 4321 and 4322
- You could even choose a "real long" number, that only is available
from internal phones, and put it to a speed dial button on the secretary
phone

If you want the manager to be able to selectively not be disturbed by
"public number" calls, but only by his secretary, some AstDB logic could
come into the game. This can be highly dynamic, or you just configure a
few extensions by hand to do exactly this:

exten => 770/4321,1,Set(DB(list/4321)=SIP/sip123&SIP/sip456)
exten => 770/4321,2,Playback(feature-donotdisturb-off)
exten => 771/4321,1,Set(DB(list/4321)=SIP/sip123)
exten => 771/4321,2,Playback(feature-donotdisturb-on)
exten => 4321,1,Dial(${DB(list/4321)})

So either the chief or the secretary could activate do-not-disturb by
dialing 771, and deactivate with 770. Just examples; choose those codes
from a range that is not in use as extensions; for my personal setup,
the 2*/3*/4*/5*/6* internal numbering for SIP devices, OOH devices, IAX
devices etc.pp., 8* being applications (like 888 the talking clock), 9*
experimental and 0* PSTN calls (how 80's! :-). A somehow similar
function (divert to VoiceMail delay in seconds can be set from any
phone, between 0 and 60 seconds) is available here as 811x.
Choose whatever suits you best.

Of course one could imagine also that the manager phone number NOT rings
the secretary while the manager is there and ready to take calls - just
edit the 770/771 lines (or add 772 for that function) - in that case,
the secretary could make use of an extension number for him/herself, but
her phone also has several lines, so why not.

HTH&BR
Anselm

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Re: [asterisk-users] Asterisk build for Suse 10.1

[Doug Crompton]

With the RPM you get what you get. Why not get the latest source at digium
and compile it. It is not hard to do.

Doug

On Wed, 10 Jan 2007, Robert A. Rawlinson wrote:

> Yes you are correct. I do NOT plan to use it again. I have downloaded
> the latest version and plan to do an install. I was hoping there might
> be an rpm for it but does not seem to be. Thanks all.
> Bob Rawlinson
>
> On Wed, 2007-01-10 at 18:25 +0100, Anton Frolov wrote:
> > he is probably tried to install one of these "All in one Asterisk" CDs, that
> > effectively formats the hard drive and installs everything from scratch,
> > including the OS ;)
> >
> > And, yes, it will happen again, if he re-runs this CD...
> >
> > AF.
> >
> >
> > Doug Crompton wrote:
> > > Formated your hardisk... wow that is nasty, but I also cannot understand
> > > how that could ever happen. There must be some other HW problem going on
> > > or you got a hold of some really bad code.
> > >
> > > What code (source or binary) and what were you doing when that happenned?
> > >
> > > Doug
> > >
> > > On Wed, 10 Jan 2007, Robert A. Rawlinson wrote:
> > >
> > >> Thanks for the help. I was concerned because I tried once before and it
> > >> formatted my hard disk. I wanted to be sure that did not happen again.\
> > >> Bob Rawlinson
> > >>
> > >> On Tue, 2007-01-09 at 11:04 -0500, Robert A. Rawlinson wrote:
> > >>> Has anyone heard of a build or instructions for installing Asterisk on a
> > >>> Suse 10.1 system?
> > >>> Bob Rawlinson
> > >>>
> > >>>
> > >>> ___
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> > > k
> > >
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"Those that sacrifice essential liberty to obtain a little temporary safety
 deserve neither liberty nor safety."  -- Ben Franklin (1759)


*  Doug Crompton   *
*  Richboro, PA 18954  *
*  215-431-6307*
*  *
* [EMAIL PROTECTED]*
* http://www.crompton.com  *



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Re: [asterisk-users] How to test VOIP quality?

[Rosli Sukri]

you can test out hammer suite of products - it is quite pricey
http://empirix.com

On 1/10/07, Doug <[EMAIL PROTECTED]> wrote:


I did a search:
<
http://www.google.com/search?q=voip+quality+%28test+OR+testing%29+asterisk-users+site%3Amail-archive.com
>

and found this:
http://www.testyourvoip.com/

This seems to have quite a bit of detail.

Does anyone have a better solution for testing
VOIP quality?

Comments?

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[asterisk-users] Festival Problems

[Robert Norton - SophMedia LLC]

Hello,
Hopefully I'm posting to the correct list, but if not, please shun me ;).

I'm running Asterisk 1.4, with Festival 1.4.1. I've got a test extension setup, 
Festival configured and for some reason, when I dial that extension I get this:

[Jan 10 17:16:05] WARNING[9082]: app_festival.c:511 festival_exec: Festival 
returned ER

See the full debug below:

[Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:3381 sip_answer: SIP answering 
channel: SIP/SMEDIA-300-086da000
[Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:6255 transmit_response_with_sdp: 
Setting framing from config on incoming call
[Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:6003 add_sdp: ** Our capability: 
0x10c (ulaw|alaw|g729) Video flag: True
[Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:6004 add_sdp: ** Our prefcodec: 0x0 
(nothing) 
[Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:6155 add_sdp: -- Done with adding 
codecs to SDP
[Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:6200 add_sdp: Done building SDP. 
Settling with this capability: 0x10c (ulaw|alaw|g729)
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1767 pbx_extension_helper: Launching 
'Festival'
[Jan 10 17:16:05] DEBUG[9082]: config.c:733 config_text_file_load: Parsing 
/usr/local/etc/asterisk/festival.conf
[Jan 10 17:16:05] DEBUG[9082]: app_festival.c:370 festival_exec: Text passed to 
festival server : This is just a test at extension 111
[Jan 10 17:16:05] DEBUG[9082]: app_festival.c:433 festival_exec: Cache file 
exists, strln=36, strlen=36
[Jan 10 17:16:05] DEBUG[9082]: app_festival.c:435 festival_exec: Size OK
[Jan 10 17:16:05] DEBUG[9082]: app_festival.c:452 festival_exec: Reading from 
cache...
[Jan 10 17:16:05] DEBUG[9082]: app_festival.c:473 festival_exec: Passing data 
to channel...
[Jan 10 17:16:05] WARNING[9082]: app_festival.c:511 festival_exec: Festival 
returned ER
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:2363 __ast_pbx_run: Spawn extension 
(local-phone,111,2) exited non-zero on 'SIP/SMEDIA-300-086da000'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is '"x300" <4806265449>'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is '4806265449'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is '111'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is 'local-phone'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is 'SIP/SMEDIA-300-086da000'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is ''
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is 'Festival'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is 'This is just a test at extension 111'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is '2007-01-10 17:16:05'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is '2007-01-10 17:16:05'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is '2007-01-10 17:16:05'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is '0'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is '0'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is 'ANSWERED'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is 'DOCUMENTATION'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is ''
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is '1168474565.2'
[Jan 10 17:16:05] DEBUG[9082]: pbx.c:1621 pbx_substitute_variables_helper_full: 
Function result is ''
[Jan 10 17:16:05] DEBUG[9082]: channel.c:1558 ast_hangup: Hanging up channel 
'SIP/SMEDIA-300-086da000'
[Jan 10 17:16:05] DEBUG[9082]: chan_sip.c:3233 sip_hangup: Hangup call 
SIP/SMEDIA-300-086da000, SIP callid [EMAIL PROTECTED])

Any ideas?


--
Robert Norton
SophMedia LLC Operations Manager
Cell: 480-234-4312 Office: 480-626-5449 (x300)
P.O. Box 7755 Tempe, AZ 85281
http://www.XStreamHost.com - Web Hosting
http://www.SophMedia.com - Consulting & Web Development





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Re: [asterisk-users] Re: Is there a low cost cell phone base station for asterisk ?

[Anselm Martin Hoffmeister]

Am Mittwoch, den 10.01.2007, 18:44 -0500 schrieb cb:
> On Jan 10, 2007, at 6:33 PM, M.Hockings wrote:

> But as far as I can tell, there is nothing requiring you to use the  
> service, if you choose not to, they work just like a normal cordless  
> phone.

There was a similar business model here in Germany: Cordless Phones for
near-to-nothing. The catch was they had a fixed prefix code.

As explanation: You can choose one of about 100 or so network operators
to put your call through by pre-pending 010xx or 0100xx to the regular
phone number.

The phone in question just prepended 010whatever to ALL phone numbers
dialled, which makes it pretty crappy to use with a line that does not
allow for network selection codes, or on lines that need a "0" for a
POTS line.

Besides, you were fixed to their per-minute prices, instead of being
able to choose other providers with up to 70% lower prices.

I was not too sad when the base station power adaptor deceaded... :)

Regards
Anselm

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Re: Fwd: [asterisk-users] Some queries on g729 license.

[Leo Ann Boon]

David Thomas wrote:

This is by far the most volotile list I have ever been on. I'm not
sure that's exactly the reputation Digium/Asterisk is shooting for,
but even so it does provide some much needed comedy relief.
Alas, it was't even related to the OP's problem. He was just trying to 
figure out why his licenses are invalid after a server failover. It just 
happens that the backup server's license was overwritten by the primary 
server license (because of disk mirroring).


Regarding the IPP-based unlicensed codec. IIRC when it first surfaced - 
the general consensus then was that we should not be telling people 
where to download the binaries. Anyone can download IPP from Intel and 
compile Readytech's codec on their own. But, please don't use this list 
to propagate the unlicensed (and not to mention mostly illegal) 
binaries. The same goes for requests for Cisco phone firmware, etc. If 
you don't have a TAC accout, get a smartnet contract or find someone who 
has TAC access.




After seeing the G.729 pricing direct from SIPRO, I now take the
"shut-up and be thankful" position. I think Digium has done us a great
service by working out favorable pricing with SIPRO.

Agreed.

Leo
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Re: [asterisk-users] Re: Is there a low cost cell phone base station for asterisk ?

[cb]

On Jan 10, 2007, at 6:33 PM, M.Hockings wrote:

But it would have been slick to be able to use the old analogue  
cell phones as we have several unused here and I am sure they would  
be cheap or free to pick up more.


I don't know about in Canada, but around here (NJ, USA), there has  
been a company selling cordless phones for next to nothing recently.  
I think the company was ATS or something like that. They appear to be  
fairly standard 2 GHz cordless phones that are meant to be plugged  
into a traditional POTS line. They have been available anywhere from  
free after rebates to as low as just a few dollars at the register.


I think the catch with them is, they all have a speed dial button  
that activates some cheap prepaid calling service. So the company is  
pushing the phones out as cheap as possible in order to get them into  
people's hands, with the hopes that they will then use the prepaid  
calling service that is built into them.


But as far as I can tell, there is nothing requiring you to use the  
service, if you choose not to, they work just like a normal cordless  
phone.



Could be an easy way to pick up a few cheap phones as long as you  
have some kind of FXS device to connect them to Asterisk (and that  
also assumes you can get them at all in Canada).


-chris



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Re: [asterisk-users] Possibility to catch DTMF when 2 users are in a conversation

[Leo Ann Boon]

exten => 1234,1,Dial(SIP/1234)
exten => 5678,1,Dial(SIP/5678)

The SIP phones (X-lite) are configured to send DTMF's using RFC 2833 
mechanism.


I want to know if it is possible in Asterisk to catch a DTMF event 
sent by one of the phone to trigger an action, for example to play a 
sound/video clip to one of the phones.
google for features.conf, But you'll need to keep asterisk in the 
callpath, i.e. canreinvite=no, otherwise the RFC2833 DTMF codes will 
only be sent between the end points. If you need to reinvite, then you 
might have to try using SIP-INFO for DTMF instead of RFC2833.


Leo


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Re: [asterisk-users] Re: Is there a low cost cell phone base station for asterisk ?

[Steve Kennedy]

On Wed, Jan 10, 2007 at 06:33:05PM -0500, M.Hockings wrote:

> That is more what I was thinking of but it is still a cell provider type 
> of hardware.  In my mind I was thinking of something very low powered 
> and turning off the roaming, etc on the phone so they only work with the 
> one base.   Think single cell base-station transceiver that can talk to 
> a cell phone and turn it into a sip conversation to Asterisk.  Here in 
> Canada, and back years ago, when I worked with radio I think the law was 
> something like less than 100mw of input power didn't require a license. 
>  However, with the advent of cell phones that could very well not be 
> the case in those bands.  But one never knows...

PicoCell have a reference design for a pico GSM basestation, but any
country allowing cell phones will require licensing (even for low
power).

You'd have to pick frequencies not used by any network and that may be
problematic.


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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Re: [asterisk-users] Directory too difficult?

[Leo Ann Boon]

Colin,

Thanks mate for the first laugh of the day.

Colin Anderson wrote:
I got a requirement list just now, with my comments inline: (showing 
it just for a giggle)
 
User requirement: 1) Directory set up by name - If person calling does 
not know employee's name, how will they access?
 
-Why, using app_telepathy.so of course!
Is app_telepathy GPL'd (General Psychic License) or do we need a bunch 
of mind reading license lawyers. ^_^
 
User requirement: 3) Not all mobile phones have the albphabet on their 
dialpads, how do they access our directory?
 
-Shout really loud. Telus should have a class action against it 
for selling Razrs with no DTMF.

Wow, didn't know there are pulse dialed mobile phone. ^_^

What about speech recognition option for directory? Not necessary a full 
speech to text engine, just enough to recognize the names would do.


Leo

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[asterisk-users] Re: Is there a low cost cell phone base station for asterisk ?

[M.Hockings]

Dumpolid Exeplish wrote:

It is true what Eric and Steve have said, you do need a licensed GSM
frequency to operate and sell GSM services (even for rural areas).
however, this link might be of interest to you

http://rfdesign.com/mag/radio_field_trials_allsoftware/



That is more what I was thinking of but it is still a cell provider type 
of hardware.  In my mind I was thinking of something very low powered 
and turning off the roaming, etc on the phone so they only work with the 
one base.   Think single cell base-station transceiver that can talk to 
a cell phone and turn it into a sip conversation to Asterisk.  Here in 
Canada, and back years ago, when I worked with radio I think the law was 
something like less than 100mw of input power didn't require a license. 
 However, with the advent of cell phones that could very well not be 
the case in those bands.  But one never knows...


In any case I'll probably lean towards something like the Engenius 
wireless phones.


http://www.engeniustech.com/telecom/products/details.aspx?id=107

But it would have been slick to be able to use the old analogue cell 
phones as we have several unused here and I am sure they would be cheap 
or free to pick up more.


Thanks to all for your input

Mike

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Re: [asterisk-users] Asterisk build for Suse 10.1

[Hans Witvliet]

On Tue, 2007-01-09 at 11:04 -0500, Robert A. Rawlinson wrote:
> Has anyone heard of a build or instructions for installing Asterisk on a
> Suse 10.1 system?
> Bob Rawlinson
> 
> 
Hi Bob,

Afair, asterisk was not on the cdrom's (which some people use to make
their own dvd), but it was on the original DVD, aswell as on several
ftp-sites.
As long as you don't intend to use isdn-bra (isdn-2) the rpm's seems to
contain all you need.
Not just the binaries, but also about 50 config files in /etc/asterisk
you need to configure. The graphical config-tool "yast" won't assist you
here... Neither for 10.2, nor will there be for 10.3 :-(
(btw, for 10.2 is asterisk-1.2 included on the cdrom's)

Hans

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Re: [asterisk-users] Re: Best inexpensive home office router forVoIP (QoS with maybe PoE)

[Mark Coccimiglio]

Jon Pounder wrote:



you should take your own advice  - an acre is 200ft x 200ft - what 
idiot would
pay a consultant $7000 to tell them they need one access point in the 
middle.




I have a BA in Electronic Engineering, a Masters in Computer Science and 
I'm an FCC licensed

radio operator.  I think I know what I'm talking about.

 Life isn't always as simple as that.  What if  its a warehouse that is 
60x800ft.  still about an acre
(I've seen this one myself).  How will the system perform once the empty 
space is occupied with
inventory?  How will metal shelving effect performance.  What hardware 
should you use?  Netgear,
dLink, Linksys, Cisco (they are different), Alvarion, Proxima? If its an 
outside area an AP in the
"middle" is not necessarly practicle.  You can't just use any antenna 
combination you want  There are
rules governing use.  Are you certified to assemble and test such a 
system for Part 15 compliance?
Do you know the specs and ERP limits?  Who has presidence FCC or OSHA 
regs?  What about
other ISM bands?  How long can you make your ethernet runs or should you 
use Fiber Optics? 
These are the types of things that an Engineer addresses. 

"...one access point in the middle. "  It may work it may not.  One 
thing for sure is that the system

probably won't perform as you expect it.

Mark C


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Re: [asterisk-users] Adding 4000 Lines to asteriskdb via asterisk -rx ?

[Ron Wellsted]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

What is needed is a family of astdb manipulation commands:
astdbput family key value
astdbget family [key]
astdbdel family [key]

any others?


- --
Ron Wellsted
[EMAIL PROTECTED] http://www.wellsted.org.uk
N 52.567623, W 2.137621 Linux Counter No. 202120
FWD:519961
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.3 (GNU/Linux)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iQEVAwUBRaVpXUtP/KMNOfRbAQLagQf+LVq3xPgwatLShzkm53+Uy+/oRN3IfnY6
bW1OcO1fhy0uhQXVY9BysDiJxvryqCOZBNMQqGpeqQA9jzvAVuGxf7heJeqDSeo4
hfidqyW+o2N1VtvhLEKNLsxucgZ76dzkvnKv6+zPVtOArSc4XTMveDFMj6CSM5yQ
3ljCzCSZpNviZjZpSXAIo3PozKaKlWJtMw9FyBQP2BPzULIOVR2VAaq4T7jEyFoT
tT5PUIRKUIzRuCRUBR+2DPdRZeif+RGd8vb9ScOROFiMmmuIxLy4UpGjFRuJajaM
pxLO2rAgLnWVhGzXQMCk6gx1hj0ovP63hXmEpUrScCR7q2J479XwGQ==
=kECV
-END PGP SIGNATURE-

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Re: [asterisk-users] Asterisk build for Suse 10.1

[Anton Frolov]

but some of the packages are labeled to be for SuSe 10.1 ...

AF.


Tzafrir Cohen wrote:
> On Wed, Jan 10, 2007 at 09:43:01PM +0100, Anton Frolov wrote:
>> well, I'm not rpm user anymore for several years already... Isn't it
>> http://www.rpmfind.com/ that is used to find the rpms?
> 
> It's meant to find rpm pckages not from your distribution that are not
> supported and may be incompatible with it. Yeah.
> 
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RE: [asterisk-users] Directory too difficult?

[Colin Anderson]

I got a requirement list just now, with my comments inline: (showing it just
for a giggle)
 
User requirement: 1) Directory set up by name - If person calling does not
know employee's name, how will they access? 
 
-Why, using app_telepathy.so of course!
 
User requirement: 2) Directory set by first &/or last name?? 
 
-Yes. Now decide which one.
 
User requirement: 3) Not all mobile phones have the albphabet on their
dialpads, how do they access our directory? 
 
-Shout really loud. Telus should have a class action against it for
selling Razrs with no DTMF.
 
 
 

-Original Message-
From: Bryan M. Johns [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 10, 2007 1:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Directory too difficult?


Exactly. 

ESU = Equipment Superior to Users

;-)


Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
  http://www.sheltonjohns.com


On Jan 10, 2007, at 12:47 PM, Alexander Lopez wrote:


More like a ID-10-T error…..








  _  


From: [EMAIL PROTECTED] [
mailto:[EMAIL PROTECTED]
 ] On Behalf Of Bryan M.
Johns
Sent: Wednesday, January 10, 2007 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Directory too difficult?



I wish had some pearl of wisdom here, but I don't.  I will simply share my
sympathy.



Sounds like an ESU situation to me.





Bryan M. Johns

Partner

Shelton | Johns Technology Group

office: 678:248:2637 x:1500

direct: 678:229:1809

mobile: 404.259.9216

iaxtel: 700:248:2637 x:1500

  http://www.sheltonjohns.com







On Jan 10, 2007, at 11:10 AM, Colin Anderson wrote:





I have a group of users whos complaint about Asterisk is that the directory

application is too hard too use. (yeah, yeah, I know. For the record,

they're Calgarians) Now I'm in a pickle: I don't want to have to create a

custom directory for these guys. Anyone have any tips for making the

directory easier, maybe re-record the prompts so they are more verbose? We

go by first name. 

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Re: [asterisk-users] Problem with zaptel drivers or card

[Joel Vandal]

Hi,

How did you, or do go about reversing the patch?
  

I have put the patch (simple) available at :

http://www.scopserv.com/download/patches/zaptel-1.2.12-reverse7860.patch

Go on your zaptel src directory and do :

patch -p0 < zaptel-1.2.12-reverse7860.patch


It is a T1 and I am not sure what you mean by behaves like an E1. The
connection is a T1 with 23 b-channels and 1 d-channel. I think it just so
happens that the problem channel is 16 on the card. This worked fine for
over a year before the upgrade to the zaptel drivers.
  

I`ve got similar problem and look like the patch #7860 is responsable of 
this issue... like if this patch doesnt check if the line is an E1 or 
T1. I have reverse the patch on 1.2.12 and all work perfectly now.


  


--
Joel Vandal, CTO
ScopServ Inc.
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Re: [asterisk-users] Get dialed numbers in AGI

[Lee Jenkins]

Ralph Liebessohn wrote:

Using:
fwrite(STDOUT,"exec read 
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15 
\n");

fwrite(STDOUT,"get variable my_var \n");
fflush(STDOUT);
$my_var=STDIN;
fwrite(STDOUT,"exec saydigits $my_var \n");

I got it:



Also you might try concatenating the values together like this:

fwrite(STDOUT,"exec saydigits " + $my_var  + "\n");

Of course, that might not be the correct operator (+) to glue together 
strings, but I bet this has something to do with it.  Your version above 
puts the variable name in the string itself and probably the php engine 
ignores it (unlike asterisk which seems to replace ${VAR} symbols within 
quotes).  So try bringing the variable out of the quoted string like the 
example that I gave above.


Just another suggestion.

--

Warm Regards,

Lee

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Re: [asterisk-users] Get dialed numbers in AGI

[Lee Jenkins]

Ralph Liebessohn wrote:

Hi Lee,

thanks for the tip. I tried other methods trying to get the variable 
value, but no success.
Doing a GET VARIABLE my_var after READ the "get variable" returns the 
value I dialed, but doesn't give the exact value to it. I got Resource 
ID #1 instead.

Using:
fwrite(STDOUT,"exec read 
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15 
\n");

fwrite(STDOUT,"get variable my_var \n");
fflush(STDOUT);
$my_var=STDIN;
fwrite(STDOUT,"exec saydigits $my_var \n");

I got it:

AGI Rx << exec read 
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
-- AGI Script Executing Application: (read) Options: 
(my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15)

-- Accepting a maximum of 5 digits.
-- Playing 
'//usr/share/asterisk/sounds/please-wait-connect-oncall-eng' (language 'en')

-- User entered '85214'
AGI Tx >> 200 result=0
AGI Rx << get variable my_var
AGI Tx >> 200 result=1 (85214)
AGI Rx << exec saydigits Resource id #1
-- AGI Script Executing Application: (saydigits) Options: (Resource)
AGI Tx >> 200 result=0
AGI Rx << exec Resource id #1
-- AGI Script Executing Application: (Resource) Options: (id)
Jan 10 17:31:33 WARNING[4867]: res_agi.c:1147 handle_exec: Could not 
find application (Resource)

AGI Tx >> 200 result=-2


I also tried:
$my_var=fwrite(STDOUT,"get variable my_var \n");

But always I get 21 as value.
More tries?



Again, I'm not familiar with php, but can you try enclosing your 
variable in either single or double quotes?  Like this?


fwrite(STDOUT,"exec saydigits \"$my_var\" \n");

...or whatever it is that you guys used to escape literals.  I use 
pascal mostly and it's strings are encased in single quotes so it's easy ;)


I looks almost like the php interpreter is handing over the literal 
pointer to the string instead of the string reference itself.  That is 
why I suggested the quotes around the string.


As another posted suggested, you should consider using a wrapper 
class/object if you're using PHP.  They've done all the work for you 
already.  If I had to write every single little piece of code that I 
used to develop software, I'd never get anything done!


Sorry can't help you more.  Hopefully someone with real php experience 
will see your post and give you a hand.


--

Warm Regards,

Lee

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RE: [asterisk-users] caller id not transferred to SIP device

[Yuan LIU]

From: Tobias Unsleber <[EMAIL PROTECTED]>

Hello,

I'm wondering why asterisk is not transferring the callerid to the sip 
device.

Scenario as follows:

sangoma <---> zaptel <---> asterisk <---> sip <---> SIP-Device

zaptel is reporting the callerid, but in the sip packages the sip-address
shows unknown as user part, as this sip debug package shows:


Have you set up a "callerid" for your Asterisk box? (Could be anything.)  I 
got "Asterisk" as caller ID before setting callerid.  Afterward (as I recall 
the sequence of events) I get caller's ID.


Yuan Liu


Executing Dial("Zap/62-1", "SIP/123|25|d") in new stack
We're at 172.31.253.80 port 10460
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP

...

--
Tobias Unsleber
VoIP Consultant



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Re: [asterisk-users] Re: Best inexpensive home office router forVoIP (QoS with maybe PoE)

[Jon Pounder]

Quoting Mark Coccimiglio <[EMAIL PROTECTED]>:




shadowym wrote:


Regardless of the 1600's spec's which are outdated in many ways by todays
standards, ANY cheap 1600 or PIX etc. you buy on Ebay will likely have MANY
MANY hours on it.  Sure, they are built to last but they do not last
forever.  I would consider ANY of these boxes as somewhat unreliable for
high availability requirements.


Buzzwrong answer!  Don't answer on things you have no idea.  and 
stop providing bad information.


you should take your own advice  - an acre is 200ft x 200ft - what idiot would
pay a consultant $7000 to tell them they need one access point in the middle.






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Jon Pounder

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   _/_/_/  _/  _/ _/_/_/  _/  _/_/
  _/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


This message was sent using IMP, the Internet Messaging Program.


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[asterisk-users] Round Robin Queue

[Felipe Neuwald]

Hi Folks,

I implemented an Asterisk 1.2.10 on a Debian GNU/Linux, and I have
implemented a round robin queue (and a memory round robin queue too).

Here I have one simple problem:

- agent 1 (busy)
- agent 2 (busy)
- agent 3 (free)

When someone call to my queue, the action of the queue is this:
call agent 1, then call agent 2, and then call agent 3, that is free and
finally ring. There is someway to my queue only call free agents?

Thank you,

Felipe Neuwald.
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[asterisk-users] generating SIP errors

[Steve Cayona]

I have a DID vendor that wants me to be able to generate specific SIP 
error messages under certain conditions and I'm completely stumped on 
how to do these:


#1 - They want to see a SIP 503 error response(service unavailable) when 
they send the call in to an active extension and and the service is not 
available, I don't have a

clue on how to simulate this.

#2 - When they send in a call to an extension that doesn't exist they 
want to see a SIP 100 TRYING message before the receive the 404 NOT 
FOUND error.  Currently

I have only been able to generate the 404 error.

Any help, clues, tips or tricks are greatly appreciated in advance.  
I've searched the web for hours begging for scraps and still have come 
up empty handed

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Re: [asterisk-users] Re: Best inexpensive home office router forVoIP (QoS with maybe PoE)

[Mark Coccimiglio]

shadowym wrote:


Regardless of the 1600's spec's which are outdated in many ways by todays
standards, ANY cheap 1600 or PIX etc. you buy on Ebay will likely have MANY
MANY hours on it.  Sure, they are built to last but they do not last
forever.  I would consider ANY of these boxes as somewhat unreliable for
high availability requirements. 

 



Buzzwrong answer!  Don't answer on things you have no idea.  and 
stop providing bad information.


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RE: [asterisk-users] Problem with zaptel drivers or card

[Administrator]

How did you, or do go about reversing the patch?

 
James 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Vandal
Sent: Wednesday, January 10, 2007 11:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with zaptel drivers or card

Administrator a écrit :
> It is a T1 and I am not sure what you mean by behaves like an E1. The
> connection is a T1 with 23 b-channels and 1 d-channel. I think it just so
> happens that the problem channel is 16 on the card. This worked fine for
> over a year before the upgrade to the zaptel drivers.
>   
I`ve got similar problem and look like the patch #7860 is responsable of 
this issue... like if this patch doesnt check if the line is an E1 or 
T1. I have reverse the patch on 1.2.12 and all work perfectly now.

--
Joel Vandal, CTO
ScopServ Inc.

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Re: [asterisk-users] Proper use of the Local channel

[lists]

No, I haven't. I'll start there.

Thanks

On Wed, January 10, 2007 2:38 pm, "Eric \"ManxPower\" Wieling" <[EMAIL 
PROTECTED]> said:

> [EMAIL PROTECTED] wrote:
>> Is there any documentation you guys can point us to in order to learn more 
>> about
>> the proper use of the Local channel? We don't currently use it. However, 
>> while
>> evaluating other people's billing and management systems for Asterisk, we 
>> noticed
>> they make extensive use of it.
> 
> Did you read localchannel.txt in the asterisk docs directory in the
> source tree?
> 

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Re: [asterisk-users] Asterisk build for Suse 10.1

[Tzafrir Cohen]

On Wed, Jan 10, 2007 at 09:43:01PM +0100, Anton Frolov wrote:
> 
> well, I'm not rpm user anymore for several years already... Isn't it
> http://www.rpmfind.com/ that is used to find the rpms?

It's meant to find rpm pckages not from your distribution that are not
supported and may be incompatible with it. Yeah.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Random dropped calls...

[Andres]

Carlos Chavez wrote:


I have a customer running Asterisk 1.2.13 with Zaptel 1.2.11 that is
having calls dropped.  Sometimes you can stay on the phone for a long
time and sometimes the call is dropped within a minute.

There are 9 lines connected to 3 TDM04B cards.  The Panasonic Pbx we
replaced did not have this problem.  There are 8 SIP phones and 16
analog phones connected to two Astribank-8 units and everyone claims
that their calls are dropped several times a day.

Any suggestions?  Here is my zapata.conf:

language=es
context=default
;rwink=300
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echotraining=yes
;echocancelwhenbridged=yes
rxgain=-1.0
txgain=0.0
busydetect=yes
 

I would put the finger on busydetect.  This need to be fine-tuned with 
settings for busycount and busypattern.



callprogress=no
accountcode=Telmex
amaflags=default
signalling=fxs_ls
group=1
faxdetect=none
callerid=asreceived
channel => 1-6

 




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--
Andres
Technical Support
http://www.telesip.net

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Re: [asterisk-users] Asterisk build for Suse 10.1

[Anton Frolov]

here it is, mainly for suse

http://rpmseek.com/rpm-pl/asterisk.html?hl=com&cs=asterisk:PN:0:0:0:0

it's only one of the rpms (the "basic" one). You should make the search yourself
(try "asterisk") to locate all of them.

AF.


Robert A. Rawlinson wrote:
> Could you point me to where it is located? I had tried Suse and
> sourceforge.
> Bob Rawlinson
> 
> On Wed, 2007-01-10 at 20:18 +0100, Anton Frolov wrote:
>> There is certainly an rpm. Not sure about 1.4, but at least for 1.2.
>>
>> AF.
>>
>>
>>
>> Robert A. Rawlinson wrote:
>>> Yes you are correct. I do NOT plan to use it again. I have downloaded
>>> the latest version and plan to do an install. I was hoping there might
>>> be an rpm for it but does not seem to be. Thanks all.
>>> Bob Rawlinson
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Re: [asterisk-users] Asterisk build for Suse 10.1

[Anton Frolov]

well, I'm not rpm user anymore for several years already... Isn't it
http://www.rpmfind.com/ that is used to find the rpms?

AF.


Robert A. Rawlinson wrote:
> Could you point me to where it is located? I had tried Suse and
> sourceforge.
> Bob Rawlinson
> 
> On Wed, 2007-01-10 at 20:18 +0100, Anton Frolov wrote:
>> There is certainly an rpm. Not sure about 1.4, but at least for 1.2.
>>
>> AF.
>>
>>
>>
>> Robert A. Rawlinson wrote:
>>> Yes you are correct. I do NOT plan to use it again. I have downloaded
>>> the latest version and plan to do an install. I was hoping there might
>>> be an rpm for it but does not seem to be. Thanks all.
>>> Bob Rawlinson
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Re: [asterisk-users] Get dialed numbers in AGI

[Yuan LIU]

From: "Ralph Liebessohn" <[EMAIL PROTECTED]>

Hi Yuan and Anton,

Let's put here all AGI for test:

#!/usr/bin/php -q

...

$my_var="123";
fflush(STDERR);
fwrite(STDERR,"Just testing\"\"\n");
fflush(STDERR);
fwrite(STDOUT,"exec read
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
\n");
fwrite(STDOUT,"exec saydigits ${my_var} \n");
fflush(STDOUT);

$conn=odbc_connect('MSSQL', 'asterisk', '123456');
$query = odbc_exec($conn, "INSERT INTO usuario(nome) VALUES('$my_var')");
?>

If I not startup $my_var="123"; Saydigits receives a NULL as options. And 
so

nothing was inserted into db.


I did a quick test and it seems that everything passed to AGI is by value, 
and there is no apparent relationship between variable named used in two 
different AGI commands.


However, a small adaption of dial plan could accomplish what you wanted, 
that is, to read the variable in dial plan, then pass its value to AGI.  
Hope this helps.


Yuan Liu

I tried to use WAIT FOR DIGIT but it makes no sense, asterisk passed 
through

it directly like Joel Lansden  reported on
9/14/06.
Is there another function or way to test it or I must try in another
asterisk box?

--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn




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Re: [asterisk-users] Asterisk build for Suse 10.1

[Tzafrir Cohen]

On Wed, Jan 10, 2007 at 02:48:41PM -0500, Robert A. Rawlinson wrote:
> Strange! I had checked on both my DVD and on the Suse site and I have
> not been able to find it. Do you happen to know where it is located?
> Bob Rawlinson

I simply checked the list of source RPMs availble from the first suse
mirror I could find at opensuse.org .

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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RE: [asterisk-users] Re: Best inexpensive home office router forVoIP (QoS with maybe PoE)

[shadowym]

Regardless of the 1600's spec's which are outdated in many ways by todays
standards, ANY cheap 1600 or PIX etc. you buy on Ebay will likely have MANY
MANY hours on it.  Sure, they are built to last but they do not last
forever.  I would consider ANY of these boxes as somewhat unreliable for
high availability requirements. 

-Original Message-
From: Ed Rubright - mail lists [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, January 10, 2007 7:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re: Best inexpensive home office router
forVoIP (QoS with maybe PoE)

Mark Coccimiglio wrote:
> Marty,
>Where are you paying $1000 for a 1600 series Cisco?  I can get you 
> 20% off that price on any quantity (note: Sarcasam).  Its not the 
> 1990's anymore.  You can get them on eBay ($50-150) for only slightly 
> more then the Linksys.  The performance is rock solid.  Three-quarters 
> of the world have used them for decades.  I know of units running 2 
> and 3 YEARS between reboots.  The power company reboots my equipment 
> more then I do.  Ok it is true that Cisco does not support the models 
> anymore, but you can't buy a services contract for a linksys router 
> either.  It can sometimes be a little difficult to configure without 
> any technical knowledge but that is what most of us get paid for.  It 
> does impress the customer when you bring in the "grey" box labled 
> "Cisco".  As for performance just try to put 50 people behind a 
> linksys/netgear/dlink.  I've used 1605R supporting +100 users.  Not 
> even a blink.  Finally, untill everyone is using >10Mps FTTH the 
> "broad band" link is still the slowest part of the connection.  Not to 
> shabby for "antiquated" technology.
>
> Mark C
>
> Martin Joseph wrote:
>
>> On 2007-01-06 00:48:11 -0800, Mark Coccimiglio <[EMAIL PROTECTED]> said:
>>
>>> Mike
>>> I'm using a Cisco 1605R [running IOS 12.3(5a)] small office router 
>>> with "Fair-Weight" queueing enabled.  Works great.  The nice thing 
>>> about Fair-Weight queueing is that it dynamically adapts to lower 
>>> the priority of higher demand traffic (e.g. large downloads).  If 
>>> you want quality stick with quality stuff.
>>>
>>> Mark C
>>
>>
>> Reread the subject line please.  $1000 (US) isn't inexpensive by any 
>> stretch.
>>
>> Marty
>>
>>
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>
Mark,

Do these 1600 series Cisco routers you mention that you find on eBay for
$50-$150 support Layer3 routing?  I have a managed switch setup on my home
network with several VLANs defined. (work subnet, home subnet, VOIP 
subnet)   I currently have to use a Linux box to route between the 
VLANs.  I'd like to move to Gigabit routing, but I'd need to replace the
Linux box(more processor power and new NICs) and that gets expensive.

I'd much rather have a router or smart switch for that matter that does
Gigabit Layer3 routing all in one unit. 

Do you have any recommendationsthat wouldn't break the bank?

Thanks,
Ed


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Re: [asterisk-users] Directory too difficult?

[Bryan M. Johns]

Exactly.

ESU = Equipment Superior to Users

;-)

Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com


On Jan 10, 2007, at 12:47 PM, Alexander Lopez wrote:


More like a ID-10-T error…..







From: [EMAIL PROTECTED] [mailto:asterisk- 
[EMAIL PROTECTED] On Behalf Of Bryan M. Johns

Sent: Wednesday, January 10, 2007 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Directory too difficult?



I wish had some pearl of wisdom here, but I don't.  I will simply  
share my sympathy.




Sounds like an ESU situation to me.



Bryan M. Johns

Partner

Shelton | Johns Technology Group

office: 678:248:2637 x:1500

direct: 678:229:1809

mobile: 404.259.9216

iaxtel: 700:248:2637 x:1500

http://www.sheltonjohns.com






On Jan 10, 2007, at 11:10 AM, Colin Anderson wrote:




I have a group of users whos complaint about Asterisk is that the  
directory


application is too hard too use. (yeah, yeah, I know. For the record,

they're Calgarians) Now I'm in a pickle: I don't want to have to  
create a


custom directory for these guys. Anyone have any tips for making the

directory easier, maybe re-record the prompts so they are more  
verbose? We


go by first name.

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[asterisk-users] Possibility to catch DTMF when 2 users are in a conversation

[Antoine Fressancourt]

Hello,

I will expose my problem here. Please tell me if it is not the right  
place as I am really new to that list.


I use Asterisk as a SIP proxy. I have two users connected to it,  
Let's call them 1234 and 5678


In my dialplan I have two lines:

exten => 1234,1,Dial(SIP/1234)
exten => 5678,1,Dial(SIP/5678)

The SIP phones (X-lite) are configured to send DTMF's using RFC 2833  
mechanism.


I want to know if it is possible in Asterisk to catch a DTMF event  
sent by one of the phone to trigger an action, for example to play a  
sound/video clip to one of the phones.


Thank you very much in advance for your help,

Antoine
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Re: [asterisk-users] Asterisk and Avaya IP Office

[housi mueller]

In this case I would need to purchase an E1 card for the Avay PBX an an other 
for the *.  To save costs, I would like to intent the interconnection over the 
FXO port.
   
  Anyone has done this configuration so far?
  

Robert Boardman <[EMAIL PROTECTED]> wrote:
  Just done this for a client using an E1 Pri card in the avaya box and a 
sangoma a102, using qsig , works fine, I wouls recommend this to any 
oneits been up and stable for two months now

Regards
Robb

housi mueller wrote:
> The main goal is that any extension from the Avaya PBX can make long 
> distance calls using the asterisk server as VoIP gateway (using a SIP 
> Provider).
> It would be also great if from a remote IP Phone (in an other 
> location), a user could use the Asterisk server to dial in and the * 
> forward’s the call to an Avaya extension.
> The Avaya has an VCM card an IP Phones (5610) as extensions. First I 
> thought to connect the * to the Avaya through the ethernet interface 
> but then I was reading in forums that there are for Avaya third party 
> IP phone licence needed and that the communication with oh323 is not 
> stable.
> I thought also putting the Asterisk in front of the Avaya.
> Telco T1 -> Asterisk <- T1 -> Avaya PBX
> This could be a solution for later one. Right know for testing it 
> would be to expensive. That's why I thought about the Avaya analog 
> Asterisk FXO interconnection.
> Any suggestions..?
>
> */Thomas Kenyon /* wrote:
>
> housi mueller wrote:
> > I would like to connect an Asterik server to an Avaya IP Office
> IP406
> > and use the * as an VoIP Gateway.
> >
> > The IP Office has two Analog extensions available. I thought
> connecting
> > this analog extensions to 2 FXO ports in the * to interconnect
> the PBX’s.
> >
> What sort of interaction are you after? It may be a better idea to
> try
> to intercept the line card with asterisk, or if the IP406 has a
> VCM card
> then to talk to it through the ethernet interface.
>
> > Is this possible? Does any one have experience with such a
> configuration?
> >
> > Thanks in advance for all recommandations and suggestions..
> >
> > Housi Mueller>
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users


 
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Re: [asterisk-users] Asterisk build for Suse 10.1

[Robert A. Rawlinson]

Could you point me to where it is located? I had tried Suse and
sourceforge.
Bob Rawlinson

On Wed, 2007-01-10 at 20:18 +0100, Anton Frolov wrote:
> There is certainly an rpm. Not sure about 1.4, but at least for 1.2.
> 
> AF.
> 
> 
> 
> Robert A. Rawlinson wrote:
> > Yes you are correct. I do NOT plan to use it again. I have downloaded
> > the latest version and plan to do an install. I was hoping there might
> > be an rpm for it but does not seem to be. Thanks all.
> > Bob Rawlinson
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Re: [asterisk-users] Random dropped calls...

[Tzafrir Cohen]

Hi!

On Wed, Jan 10, 2007 at 01:14:59PM -0600, Carlos Chavez wrote:
>   I have a customer running Asterisk 1.2.13 with Zaptel 1.2.11 that is
> having calls dropped.  Sometimes you can stay on the phone for a long
> time and sometimes the call is dropped within a minute.
> 
>   There are 9 lines connected to 3 TDM04B cards.  The Panasonic Pbx we
> replaced did not have this problem.  There are 8 SIP phones and 16
> analog phones connected to two Astribank-8 units and everyone claims
> that their calls are dropped several times a day.
> 
>   Any suggestions?  Here is my zapata.conf:
> 
> language=es
> context=default
> ;rwink=300
> usecallerid=yes
> hidecallerid=no
> callwaiting=no
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echotraining=yes
> ;echocancelwhenbridged=yes
> rxgain=-1.0
> txgain=0.0
> busydetect=yes
> callprogress=no
> accountcode=Telmex
> amaflags=default
> signalling=fxs_ls
> group=1
> faxdetect=none
> callerid=asreceived
> channel => 1-6

Those are 6 channels of the 9?

What is the configuration of the other three?

What is the configuration of the 16 Astribank channels?

You don't set busycount. This uses the default value (3?). Can you try
setting it to a higher value?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Asterisk build for Suse 10.1

[Robert A. Rawlinson]

Strange! I had checked on both my DVD and on the Suse site and I have
not been able to find it. Do you happen to know where it is located?
Bob Rawlinson

On Wed, 2007-01-10 at 21:03 +0200, Tzafrir Cohen wrote:
> On Wed, Jan 10, 2007 at 01:25:31PM -0500, Robert A. Rawlinson wrote:
> > Yes you are correct. I do NOT plan to use it again. I have downloaded
> > the latest version and plan to do an install. I was hoping there might
> > be an rpm for it but does not seem to be. Thanks all.
> > Bob Rawlinson
> 
> suse 10.1 actually includes a package of Asterisk 1.2.5 . 10.2 includes
> 1.2.13 . I have no idea if security updates bothered updating 1.2.5 .
> 

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Re: [asterisk-users] Get dialed numbers in AGI

[Ralph Liebessohn]

On 1/10/07, Lee Jenkins <[EMAIL PROTECTED]> wrote:


Ralph Liebessohn wrote:
> Hi,
>
> I'm trying to write a AGI in PHP to get the numbers dialed (with
> read()), save it into a variable to insert it into a SQL server
> database. But I cannot see results into the variable, it always return
> NULL.
> Here is a piece of the AGI.
>
> fwrite(STDOUT,"exec Read my_var|/sound_to_play|5|||15 \n");
> fflush(STDOUT);
> $conn=odbc_connect('MSSQL', 'USER', 'PASS');
> $query = odbc_exec($conn, "INSERT INTO dialed(number)
VALUES('$my_var')");
>
> Even if I only show my_var value or try to use it inside asterisk, the
> value is NULL.
> There is another way to do it? Am I doing a mistake here?
> I'm using Asterisk 1.2.13.
>

I'm not a php guy, but aren't we missing the part that retrieves the
value saved into my_var from the call to READ?

// In this part you run the read command and asterisk
// stores the value into the channel variable "my_var"

fwrite(STDOUT,"exec Read my_var|/sound_to_play|5|||15 \n");

// In this part you are constructing your sql statement
// with a null value cause you didn't make a call to
// GET VARIABLE before constructing your sql.

$query = odbc_exec($conn, "INSERT INTO dialed(number) VALUES('$my_var')");

--

Warm Regards,

Lee




Hi Lee,

thanks for the tip. I tried other methods trying to get the variable value,
but no success.
Doing a GET VARIABLE my_var after READ the "get variable" returns the value
I dialed, but doesn't give the exact value to it. I got Resource ID #1
instead.
Using:
fwrite(STDOUT,"exec read
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
\n");
fwrite(STDOUT,"get variable my_var \n");
fflush(STDOUT);
$my_var=STDIN;
fwrite(STDOUT,"exec saydigits $my_var \n");

I got it:

AGI Rx << exec read
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
   -- AGI Script Executing Application: (read) Options:
(my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15)
   -- Accepting a maximum of 5 digits.
   -- Playing '//usr/share/asterisk/sounds/please-wait-connect-oncall-eng'
(language 'en')
   -- User entered '85214'
AGI Tx >> 200 result=0
AGI Rx << get variable my_var
AGI Tx >> 200 result=1 (85214)
AGI Rx << exec saydigits Resource id #1
   -- AGI Script Executing Application: (saydigits) Options: (Resource)
AGI Tx >> 200 result=0
AGI Rx << exec Resource id #1
   -- AGI Script Executing Application: (Resource) Options: (id)
Jan 10 17:31:33 WARNING[4867]: res_agi.c:1147 handle_exec: Could not find
application (Resource)
AGI Tx >> 200 result=-2


I also tried:
$my_var=fwrite(STDOUT,"get variable my_var \n");

But always I get 21 as value.
More tries?

--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn
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Re: [asterisk-users] RTP directly

[Eric \"ManxPower\" Wieling]

David Alcott wrote:


Is there a way to configure the Asterisk so that the RTP goes directly 
between the Endpoints as opposed to going through the asterisk?


That is the default if Asterisk believes it will work.  Things that 
might not make it work is tTwW options to Dial, protocol transation (one 
leg is SIP, the other is IAX2, transcoding, NAT, or many other things 
that make the two legs of the call not compatible with reinvites.

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Re: [asterisk-users] Proper use of the Local channel

[Eric \"ManxPower\" Wieling]

[EMAIL PROTECTED] wrote:

Is there any documentation you guys can point us to in order to learn more 
about the proper use of the Local channel? We don't currently use it. However, 
while evaluating other people's billing and management systems for Asterisk, we 
noticed they make extensive use of it.


Did you read localchannel.txt in the asterisk docs directory in the 
source tree?

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Re: [asterisk-users] Get dialed numbers in AGI

[Anton Frolov]

hi

I never really programmed in PHP, I use Perl for my purposes.
I found a good AGI library for Perl and is happy with it. It allows me to call
functions instead of parsing the input.
While looking for my library, I saw at least one for PHP. So why not to use it?

In Perl it looks like:
  my %agiArgs = $AGI->ReadParse();
  my $callerNum = $agiArgs{"callerid"}; // Got the caller id
  $retval = $AGI->exec('Dial', $CHANNEL."|".$CALL_OPTIONS); // Placing a call

it's so simple...
(and you have the error checking "built in"!)

I'm sure you could find such a library for PHP as well!

AF.


Ralph Liebessohn wrote:
> On 1/10/07, *Yuan LIU* <[EMAIL PROTECTED] >
> wrote:
> 
> Then there must be an error somewhere.  The variable READ() in Asterisk
> should be usable.  Should be able to use SayDigits() to play it back
> - or no
> value is read.
> 
> Yuan Liu
> 
> 
>  Hi Yuan and Anton,
> 
> Let's put here all AGI for test:
> 
> #!/usr/bin/php -q
>  ob_implicit_flush(false);
> error_reporting(0);
> $stdin = fopen( 'php://stdin', 'r' );
> 
> if (!defined('STDIN'))
> {
> define('STDIN',fopen('php://stdin','r'));
> }
> if (!defined('STDOUT'))
> {
> define('STDOUT',fopen('php://stdout','r'));
> }
> if (!defined('STDERR'))
> {
> define('STERR',fopen('php://stderr','r'));
> }
> 
> while(!feof($stdin))
> {
> $temp=trim(fgets(STDIN,4096));
> if (($temp=="") || ($temp="\n"))
> {
> break;
> }
> $s=split(":",$temp);
> $nome=str_subst("agi_","",$s[0]);
> $agi[$nome]=trim($s[1]);
> }
> 
> foreach($agi as $chave=>$valor)
> {
> fwrite(STDERR,"--$chave=$valor\n");
> fflush(STDERR);
> }
> $my_var="123";
> fflush(STDERR);
> fwrite(STDERR,"Just testing\"\"\n");
> fflush(STDERR);
> fwrite(STDOUT,"exec read
> my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
> \n");
> fwrite(STDOUT,"exec saydigits ${my_var} \n");
> fflush(STDOUT);
> 
> $conn=odbc_connect('MSSQL', 'asterisk', '123456');
> $query = odbc_exec($conn, "INSERT INTO usuario(nome) VALUES('$my_var')");
> ?>
> 
> If I not startup $my_var="123"; Saydigits receives a NULL as options.
> And so nothing was inserted into db.
> I tried to use WAIT FOR DIGIT but it makes no sense, asterisk passed
> through it directly like Joel Lansden 
> reported on 9/14/06.
> Is there another function or way to test it or I must try in another
> asterisk box?
> 
> -- 
> Ralph Liebessohn
> ICQ: 74835911
> Skype: liebessohn
> 
> 
> 
> 
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Re: [asterisk-users] Asterisk build for Suse 10.1

[Anton Frolov]

There is certainly an rpm. Not sure about 1.4, but at least for 1.2.

AF.



Robert A. Rawlinson wrote:
> Yes you are correct. I do NOT plan to use it again. I have downloaded
> the latest version and plan to do an install. I was hoping there might
> be an rpm for it but does not seem to be. Thanks all.
> Bob Rawlinson
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[asterisk-users] SIP invite and sip.conf relationship?

[Tony Mountifield]

I'm having a bit of trouble setting up my sip.conf entries to accept
calls from a particular provider, and the problem really is that I am
unclear exactly what parts of the INVITE are supposed to match what
parts of sip.conf.

I couldn't find this info on the wiki, so if someone here can shed
some light, I would be very grateful!

Here are the relevant lines from the INVITE (from sip debug):

<-- SIP read from 213.166.5.130:5060: 
INVITE sip:[EMAIL PROTECTED] SIP/2.0
From: "01234567890" ;tag=2F6B6198-D3D
To: 
Date: Wed, 10 Jan 2007 18:18:22 gmt
CSeq: 101 INVITE
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact: 

How do the items above, such as source address, INVITE URL, From, To, etc.,
relate to items in sip.conf in a type=user section, such as [sectionname],
user=username, host=hostname or host=dynamic, etc?

My provider gives me the option to set the invite URL, such as
sip:sip.mydomain.com or sip:[EMAIL PROTECTED], but does not use
a secret to authenticate. Does the myuser part get used at all?

Thanks for any insight.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] Random dropped calls...

[Carlos Chavez]

I have a customer running Asterisk 1.2.13 with Zaptel 1.2.11 that is
having calls dropped.  Sometimes you can stay on the phone for a long
time and sometimes the call is dropped within a minute.

There are 9 lines connected to 3 TDM04B cards.  The Panasonic Pbx we
replaced did not have this problem.  There are 8 SIP phones and 16
analog phones connected to two Astribank-8 units and everyone claims
that their calls are dropped several times a day.

Any suggestions?  Here is my zapata.conf:

language=es
context=default
;rwink=300
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echotraining=yes
;echocancelwhenbridged=yes
rxgain=-1.0
txgain=0.0
busydetect=yes
callprogress=no
accountcode=Telmex
amaflags=default
signalling=fxs_ls
group=1
faxdetect=none
callerid=asreceived
channel => 1-6

-- 
Telecomunicaciones Abiertas de Mexico S.A. de C.V.
Carlos Chàvez Prats
Director de Tecnologìa
+52-55-91169161 ext 2001


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Re: [asterisk-users] Asterisk build for Suse 10.1

[Tzafrir Cohen]

On Wed, Jan 10, 2007 at 01:25:31PM -0500, Robert A. Rawlinson wrote:
> Yes you are correct. I do NOT plan to use it again. I have downloaded
> the latest version and plan to do an install. I was hoping there might
> be an rpm for it but does not seem to be. Thanks all.
> Bob Rawlinson

suse 10.1 actually includes a package of Asterisk 1.2.5 . 10.2 includes
1.2.13 . I have no idea if security updates bothered updating 1.2.5 .

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Problem with zaptel drivers or card

[Joel Vandal]

Administrator a écrit :

It is a T1 and I am not sure what you mean by behaves like an E1. The
connection is a T1 with 23 b-channels and 1 d-channel. I think it just so
happens that the problem channel is 16 on the card. This worked fine for
over a year before the upgrade to the zaptel drivers.
  
I`ve got similar problem and look like the patch #7860 is responsable of 
this issue... like if this patch doesnt check if the line is an E1 or 
T1. I have reverse the patch on 1.2.12 and all work perfectly now.


--
Joel Vandal, CTO
ScopServ Inc.

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Re: [asterisk-users] Get dialed numbers in AGI

[Ralph Liebessohn]

On 1/10/07, Yuan LIU <[EMAIL PROTECTED]> wrote:


Then there must be an error somewhere.  The variable READ() in Asterisk
should be usable.  Should be able to use SayDigits() to play it back - or
no
value is read.

Yuan Liu



Hi Yuan and Anton,

Let's put here all AGI for test:

#!/usr/bin/php -q
$valor)
{
   fwrite(STDERR,"--$chave=$valor\n");
   fflush(STDERR);
}
$my_var="123";
fflush(STDERR);
fwrite(STDERR,"Just testing\"\"\n");
fflush(STDERR);
fwrite(STDOUT,"exec read
my_var|//usr/share/asterisk/sounds/please-wait-connect-oncall-eng|5|||15
\n");
fwrite(STDOUT,"exec saydigits ${my_var} \n");
fflush(STDOUT);

$conn=odbc_connect('MSSQL', 'asterisk', '123456');
$query = odbc_exec($conn, "INSERT INTO usuario(nome) VALUES('$my_var')");
?>

If I not startup $my_var="123"; Saydigits receives a NULL as options. And so
nothing was inserted into db.
I tried to use WAIT FOR DIGIT but it makes no sense, asterisk passed through
it directly like Joel Lansden  reported on
9/14/06.
Is there another function or way to test it or I must try in another
asterisk box?

--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn
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Re: [asterisk-users] Get dialed numbers in AGI

[Lee Jenkins]

Ralph Liebessohn wrote:

Hi,

I'm trying to write a AGI in PHP to get the numbers dialed (with 
read()), save it into a variable to insert it into a SQL server 
database. But I cannot see results into the variable, it always return 
NULL.

Here is a piece of the AGI.

fwrite(STDOUT,"exec Read my_var|/sound_to_play|5|||15 \n");
fflush(STDOUT);
$conn=odbc_connect('MSSQL', 'USER', 'PASS');
$query = odbc_exec($conn, "INSERT INTO dialed(number) VALUES('$my_var')");

Even if I only show my_var value or try to use it inside asterisk, the 
value is NULL.

There is another way to do it? Am I doing a mistake here?
I'm using Asterisk 1.2.13.



I'm not a php guy, but aren't we missing the part that retrieves the 
value saved into my_var from the call to READ?


// In this part you run the read command and asterisk
// stores the value into the channel variable "my_var"

fwrite(STDOUT,"exec Read my_var|/sound_to_play|5|||15 \n");

// In this part you are constructing your sql statement
// with a null value cause you didn't make a call to
// GET VARIABLE before constructing your sql.

$query = odbc_exec($conn, "INSERT INTO dialed(number) VALUES('$my_var')");

--

Warm Regards,

Lee

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Re: [asterisk-users] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)

[Jon Pounder]

Quoting Mark Coccimiglio <[EMAIL PROTECTED]>:




Mark,

Do these 1600 series Cisco routers you mention that you find on eBay 
for $50-$150 support Layer3 routing?  I have a managed switch setup 
on my home network with several VLANs defined. (work subnet, home 
subnet, VOIP subnet)   I currently have to use a Linux box to route 
between the VLANs.  I'd like to move to Gigabit routing, but I'd 
need to replace the Linux box(more processor power and new NICs) and 
that gets expensive.


I'd much rather have a router or smart switch for that matter that 
does Gigabit Layer3 routing all in one unit.

Do you have any recommendationsthat wouldn't break the bank?


Do you actually know how utilized the linux box is now ? its probably 
near zero

and all you need is a couple cards. Routing even with complex rules takes very
little cpu.







Thanks,
Ed



Ed,
  Layer3 routing is a fundamental function of a router which is 
supported by the Cisco 1600 series (1605R specifically) router.  
However VLAN definitations are not supported in the 1600 series.  You 
would need to moveup to the 1700 or 2500 series for that function.  
As for Gigabit support the 1600 and 1700 series do not support that 
high speed interface.  These router are designed around WAN style 
routing operating at ~1.5Mbps.  Gigabit routing is a rather cutting 
edge capablity that is only seen in newer hardware.  I would checkout 
a Cisco Catalyst 3500 series for something like that.  Be carefull 
and look closely some systems only support 2 ports on 1000baseT and 
the rest are 100BaseT.


Good luck and happy hunting,

Mark Coccimiglio


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Jon Pounder

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[asterisk-users] Zap calls

[Jay Moore]

I have 8 Zap channels, 25-32, all of which have their own line.

My zapata.conf file looks similar to:

group=1
context=context_1
signalling=fxs_ks
channel => 25

group=2
context=context_2
signalling=fxs_ks
channel => 26

and so forth for all 8 lines, where each channel has its own group and 
incoming context.


The first 4 channels are our primary trunk lines.  If we have to make an 
 outgoing call on a trunk line, how can I have it pick the first 
available line of the 4?


My first thought would be to have another group that includes the first 
4 channels, and then use that group in the Dial() command like so:


group=9
context=whatever
signally=fxs_ks
channel => 25-28

and
Dial(Zap/g9/${EXTEN},60)


Can I repeat channels like that or will it cause Asterisk to choke?  If 
I can't do it that way, can someone suggest a way to do it?


Thanks in advance,
Jay
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Re: [asterisk-users] dundi ENCREJ

[Alex Robar]

Hi Ramon,

Please post your peer details from dundi.conf so we can see what your setup
is.

Also, have you tried regenerating your keys? I wound up generating my keys
twice, they just didn't work the first time, I'm not sure why.

Alex

On 1/10/07, Ramon Schönborn <[EMAIL PROTECTED]> wrote:


hi list,

i have the same problem as mentioned here:

http://forums.digium.com/viewtopic.php?t=2678&view=next&sid=bd94cefd823b23156c5748843febb3ab

my asterisk version is 1.2.12.1

any ideas?






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--
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[EMAIL PROTECTED]
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Re: [asterisk-users] Asterisk build for Suse 10.1

[Robert A. Rawlinson]

Yes you are correct. I do NOT plan to use it again. I have downloaded
the latest version and plan to do an install. I was hoping there might
be an rpm for it but does not seem to be. Thanks all.
Bob Rawlinson

On Wed, 2007-01-10 at 18:25 +0100, Anton Frolov wrote:
> he is probably tried to install one of these "All in one Asterisk" CDs, that
> effectively formats the hard drive and installs everything from scratch,
> including the OS ;)
> 
> And, yes, it will happen again, if he re-runs this CD...
> 
> AF.
> 
> 
> Doug Crompton wrote:
> > Formated your hardisk... wow that is nasty, but I also cannot understand
> > how that could ever happen. There must be some other HW problem going on
> > or you got a hold of some really bad code.
> > 
> > What code (source or binary) and what were you doing when that happenned?
> > 
> > Doug
> > 
> > On Wed, 10 Jan 2007, Robert A. Rawlinson wrote:
> > 
> >> Thanks for the help. I was concerned because I tried once before and it
> >> formatted my hard disk. I wanted to be sure that did not happen again.\
> >> Bob Rawlinson
> >>
> >> On Tue, 2007-01-09 at 11:04 -0500, Robert A. Rawlinson wrote:
> >>> Has anyone heard of a build or instructions for installing Asterisk on a
> >>> Suse 10.1 system?
> >>> Bob Rawlinson
> >>>
> >>>
> >>> ___
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> > k
> > 
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Re: [asterisk-users] Send email notification

[Anselm Martin Hoffmeister]

Am Mittwoch, den 10.01.2007, 13:41 -0400 schrieb H Aranguren:
> Thanks for your answer Anselm,
> 
>But, why do you think that the problem is in the mail server, if I
> can send mails with esmtp, with the command /usr/sbin/sendmail without
> problem. But the Voicemail app never sends the notification.
> 
> I used ethereal and i couldn't see any message from asterisk box to my
> smtp server when i leave a voicemail

My experience with people setting up a mail server is that they tend to
forget small but important things - I often enough do myself, with all
that complexity. Like the possibility that the "sendmail" command, run
as user "asterisk", will not be allowed to send mail from any
e-mail-adress but "[EMAIL PROTECTED]" or so. If you use the sendmail -t
command as you wrote, then the first step of any e-mail to be sent will
be local and not appear in ethereal. Have you looked in the log files?
Are you _sure_ notifications are not sent?
When you replace "sendmail -t" with something like "cat > /tmp/1", will
that file appear? Might you have a PATH issue, like sendmail living
in /usr/bin instead of /usr/sbin/?

Just guessing in the dark, and naming things that would most probably
happen in a debug session if I sat in front of the machine.

Hth
Anselm

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Re: [asterisk-users] Send email notification

[Tzafrir Cohen]

On Wed, Jan 10, 2007 at 01:41:39PM -0400, H Aranguren wrote:
> Thanks for your answer Anselm,
> 
>   But, why do you think that the problem is in the mail server, if I
> can send mails with esmtp, with the command /usr/sbin/sendmail without
> problem. But the Voicemail app never sends the notification.

But you use a different command. Why do you need to override the default
sendmail command, BTW?

> 
> I used ethereal and i couldn't see any message from asterisk box to my
> smtp server when i leave a voicemail

What is /usr/sbin/sendmail? sendmail? postfix? any other MTA?

What do you see in its logs?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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RE: [asterisk-users] Problem with zaptel drivers or card

[Administrator]

It is a T1 and I am not sure what you mean by behaves like an E1. The
connection is a T1 with 23 b-channels and 1 d-channel. I think it just so
happens that the problem channel is 16 on the card. This worked fine for
over a year before the upgrade to the zaptel drivers.

 
James 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Wednesday, January 10, 2007 10:06 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Problem with zaptel drivers or card

On Wed, Jan 10, 2007 at 09:39:47AM -0700, Administrator wrote:
> Results From cat /proc/zaptel/*
> 
> Span 1: WCTDM/0 "Wildcard TDM2400P Board 1"
> IRQ misses: 24
> 
>1 WCTDM/0/0 FXOKS (In use)
>2 WCTDM/0/1 FXOKS (In use)
>3 WCTDM/0/2 FXOKS (In use)
>4 WCTDM/0/3 FXOKS (In use)
>5 WCTDM/0/4 FXOKS (In use)
>6 WCTDM/0/5 FXOKS (In use)
>7 WCTDM/0/6 FXOKS (In use)
>8 WCTDM/0/7 FXOKS (In use)
>9 WCTDM/0/8 FXOKS (In use)
>   10 WCTDM/0/9 FXOKS (In use)
>   11 WCTDM/0/10 FXOKS (In use)
>   12 WCTDM/0/11 FXOKS (In use)
>   13 WCTDM/0/12 FXOKS (In use)
>   14 WCTDM/0/13 FXOKS (In use)
>   15 WCTDM/0/14 FXOKS (In use)
>   16 WCTDM/0/15 FXOKS (In use)
>   17 WCTDM/0/16 FXOKS (In use)
>   18 WCTDM/0/17 FXOKS (In use)
>   19 WCTDM/0/18 FXOKS (In use)
>   20 WCTDM/0/19 FXOKS (In use)
>   21 WCTDM/0/20 FXOKS (In use)
>   22 WCTDM/0/21 FXOKS (In use)
>   23 WCTDM/0/22 FXOKS (In use)
>   24 WCTDM/0/23 FXOKS (In use)
> Span 2: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" B8ZS/ESF
> 
>   25 WCT1/0/1 Clear (In use)
>   26 WCT1/0/2 Clear (In use)
>   27 WCT1/0/3 Clear (In use)
>   28 WCT1/0/4 Clear (In use)
>   29 WCT1/0/5 Clear (In use)
>   30 WCT1/0/6 Clear (In use)
>   31 WCT1/0/7 Clear (In use)
>   32 WCT1/0/8 Clear (In use)
>   33 WCT1/0/9 Clear (In use)
>   34 WCT1/0/10 Clear (In use)
>   35 WCT1/0/11 Clear (In use)
>   36 WCT1/0/12 Clear (In use)
>   37 WCT1/0/13 Clear (In use)
>   38 WCT1/0/14 Clear (In use)
>   39 WCT1/0/15 Clear (In use)
>   40 WCT1/0/16
>   41 WCT1/0/17 Clear (In use)
>   42 WCT1/0/18 Clear (In use)
>   43 WCT1/0/19 Clear (In use)
>   44 WCT1/0/20 Clear (In use)
>   45 WCT1/0/21 Clear (In use)
>   46 WCT1/0/22 Clear (In use)
>   47 WCT1/0/23 Clear (In use)
>   48 WCT1/0/24 HDLCFCS (In use)

Is it supposed to be T1 or E1?

The card behaves as E1 but you attempt to configure it as T1.

> 
> zaptel.conf file:
> 
> fxoks=1-24
> span=2,1,0,esf,b8zs
> bchan=25-39
> #bchan=40
> bchan=41-47
> dchan=48
> loadzone=us
> defaultzone=us
> 
> James
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
> Sent: Tuesday, January 09, 2007 11:25 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Problem with zaptel drivers or card
> 
> On Tue, Jan 09, 2007 at 06:01:55PM -0700, Administrator wrote:
> > I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 1.4,
> and
> > Zaptel 1.4
> > 
> > The Digium cards installed are TDM2400 and TE110P.
> > 
> > Everything was working fine until I upgraded to zaptel 1.2.12 from 1.2.9
> > 
> > Now when I run ztcfg I get the following error message:
> > 
> > (CAS signalling on span 2 conflicts with Clear channel on channel 40)
> > 
> > --NOTE: signaling was spelled wrong in the error message--
> > 
> > I have since upgraded to 1.4 with the same problem.
> > 
> > Channel 40 is a standard bchan configuration and our provider sees no
> > problem with the channel.
> > 
> > When I disable the channel everything works fine.
> > 
> > My assumption is that something is wrong with the TE110P card.
> > 
> > Has anyone seen anything else like this?
> 
> What do you get from:
> 
> cat /proc/zaptel/*
> 
> What do you have on /etc/zaptel.conf  ?
> 
> -- 
>Tzafrir Cohen   
> icq#16849755jabber:[EMAIL PROTECTED]
> +972-50-7952406   mailto:[EMAIL PROTECTED]   
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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+972-50-7952406  

Re: [asterisk-users] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)

[Mark Coccimiglio]

Mark,

Do these 1600 series Cisco routers you mention that you find on eBay 
for $50-$150 support Layer3 routing?  I have a managed switch setup on 
my home network with several VLANs defined. (work subnet, home subnet, 
VOIP subnet)   I currently have to use a Linux box to route between 
the VLANs.  I'd like to move to Gigabit routing, but I'd need to 
replace the Linux box(more processor power and new NICs) and that gets 
expensive.


I'd much rather have a router or smart switch for that matter that 
does Gigabit Layer3 routing all in one unit.

Do you have any recommendationsthat wouldn't break the bank?

Thanks,
Ed



Ed,
  Layer3 routing is a fundamental function of a router which is 
supported by the Cisco 1600 series (1605R specifically) router.  However 
VLAN definitations are not supported in the 1600 series.  You would need 
to moveup to the 1700 or 2500 series for that function.  As for Gigabit 
support the 1600 and 1700 series do not support that high speed 
interface.  These router are designed around WAN style routing operating 
at ~1.5Mbps.  Gigabit routing is a rather cutting edge capablity that is 
only seen in newer hardware.  I would checkout a Cisco Catalyst 3500 
series for something like that.  Be carefull and look closely some 
systems only support 2 ports on 1000baseT and the rest are 100BaseT.


Good luck and happy hunting,

Mark Coccimiglio


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RE: [asterisk-users] Directory too difficult?

[Alexander Lopez]

More like a ID-10-T error.

 

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bryan M.
Johns
Sent: Wednesday, January 10, 2007 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Directory too difficult?

 

I wish had some pearl of wisdom here, but I don't.  I will simply share
my sympathy.

 

Sounds like an ESU situation to me.

 

Bryan M. Johns

Partner

Shelton | Johns Technology Group

office: 678:248:2637 x:1500

direct: 678:229:1809

mobile: 404.259.9216

iaxtel: 700:248:2637 x:1500

http://www.sheltonjohns.com  





 

On Jan 10, 2007, at 11:10 AM, Colin Anderson wrote:





I have a group of users whos complaint about Asterisk is that the
directory

application is too hard too use. (yeah, yeah, I know. For the record,

they're Calgarians) Now I'm in a pickle: I don't want to have to create
a

custom directory for these guys. Anyone have any tips for making the

directory easier, maybe re-record the prompts so they are more verbose?
We

go by first name. 

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[asterisk-users] dundi ENCREJ

[Ramon Schönborn]

hi list,

i have the same problem as mentioned here:
http://forums.digium.com/viewtopic.php?t=2678&view=next&sid=bd94cefd823b23156c5748843febb3ab

my asterisk version is 1.2.12.1

any ideas?






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[asterisk-users] Send email notification

[H Aranguren]

Thanks for your answer Anselm,

  But, why do you think that the problem is in the mail server, if I
can send mails with esmtp, with the command /usr/sbin/sendmail without
problem. But the Voicemail app never sends the notification.

I used ethereal and i couldn't see any message from asterisk box to my
smtp server when i leave a voicemail

Thanks



Am Mittwoch, den 10.01.2007, 11:46 -0400 schrieb H Aranguren:

Hi group,

I'm trying to configure the email notification when a user leave a
voicemail, but don't work (send email notification).

I configured esmtp in my linux box, if a try to use it with command
line, it works fine. (echo "Hello" | sendmail a at b.com -f b at c.com).


You could look wether a voicemail triggers something to happen inside
the mail system at all (logfiles...). In that case, chances are that the
mail cannot be parsed because of misconfiguration in the mail server /
restricted usage of the sendmail -t command or whatever.
In my setup (SMTP server listening on port 25 of the same machine) the
mailcmd is commented out, and It Just Works(tm). If you need mail system
specific help, there sure are lots of forums and info, but I cannot tell
where to connect to esmtp people. Exim is my favourite ;)



My voicemail.conf
[general]
format=wav49
attach=yes
serveremail=anonymous at abc.com
fromstring=Asterisk
mailcmd=/usr/sbin/sendmail -t
[my_home]
100 => ,number100,number100 at abc.com

Can you see the problem?. Do you know any documentarion on internet
where can i solve the problem?



BR
Anselm

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Re: [asterisk-users] Get dialed numbers in AGI

[Yuan LIU]

From: Anton Frolov <[EMAIL PROTECTED]>

you could use one of the AGI libraries...
then you can just call a function to get the number.

AF.

Ralph Liebessohn wrote:
> Hi,
>
> I'm trying to write a AGI in PHP to get the numbers dialed (with
> read()), save it into a variable to insert it into a SQL server
> database. But I cannot see results into the variable, it always return
> NULL.
> Here is a piece of the AGI.
>
> fwrite(STDOUT,"exec Read my_var|/sound_to_play|5|||15 \n");
> fflush(STDOUT);
> $conn=odbc_connect('MSSQL', 'USER', 'PASS');
> $query = odbc_exec($conn, "INSERT INTO dialed(number) 
VALUES('$my_var')");

>
> Even if I only show my_var value or try to use it inside asterisk, the
> value is NULL.


Then there must be an error somewhere.  The variable READ() in Asterisk 
should be usable.  Should be able to use SayDigits() to play it back - or no 
value is read.


Yuan Liu


> There is another way to do it? Am I doing a mistake here?
> I'm using Asterisk 1.2.13.
>
> Thank you all.
>
> --
> Ralph Liebessohn
> ICQ: 74835911
> Skype: liebessohn



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[asterisk-users] VIA EPIA DeadLock Issues

[Raymond McKay]

Greetings,

I've been having a large number of deadlock issues lately on chan_sip occurring 
only on VIA EPIA ML6000 boards.  I'm curious if anyone else is having similar 
issues.

My Config (have multiple systems all running the same hardware with the same 
problem)

VIA EPIA ML6000
1GB RAM
80GB HDD
Various Digium Cards (T1 and TDM cards)
Trixbox 1.2.2 (though running stock asterisk code)
Asterisk Versions 1.2.12 - 1.2.14 - with and without metermaid patch

Problem seems to happen more on systems that use parking lots.  The system will 
run for around 24 hours or so fine, and then mysteriously, without any errors 
leading up to it,  will stop being able to send calls to the chan_sip.  System 
from that point on reports the following in the logs.

Dec 13 12:07:04 DEBUG[16415] chan_zap.c: Took Zap/1-1 off hook
Dec 13 12:07:04 VERBOSE[16415] logger.c: -- Executing Wait("Zap/1-1", "1") 
in new stack
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 WARNING[2049] channel.c: Avoided initial deadlock for 
'0x9896848', 10 retries!
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 
'SIP/100-09883f80'
Dec 13 12:07:04 WARNING[2049] channel.c: Avoided initial deadlock for 
'0x9896848', 10 retries!

attempting to stop asterisk from the CLI causes the CLI to become unresponsive 
and a trace shows chan_sip goes into a mutex_wait state. 

Anybody seen this? Have a fix?

Raymond McKay
President
RAYNET Technologies LLC
http://www.raynettech.com
(860) 693-2226 x 31
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Re: [asterisk-users] Asterisk build for Suse 10.1

[Anton Frolov]

he is probably tried to install one of these "All in one Asterisk" CDs, that
effectively formats the hard drive and installs everything from scratch,
including the OS ;)

And, yes, it will happen again, if he re-runs this CD...

AF.


Doug Crompton wrote:
> Formated your hardisk... wow that is nasty, but I also cannot understand
> how that could ever happen. There must be some other HW problem going on
> or you got a hold of some really bad code.
> 
> What code (source or binary) and what were you doing when that happenned?
> 
> Doug
> 
> On Wed, 10 Jan 2007, Robert A. Rawlinson wrote:
> 
>> Thanks for the help. I was concerned because I tried once before and it
>> formatted my hard disk. I wanted to be sure that did not happen again.\
>> Bob Rawlinson
>>
>> On Tue, 2007-01-09 at 11:04 -0500, Robert A. Rawlinson wrote:
>>> Has anyone heard of a build or instructions for installing Asterisk on a
>>> Suse 10.1 system?
>>> Bob Rawlinson
>>>
>>>
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> k
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Re: [asterisk-users] Problem with zaptel drivers or card

[Tzafrir Cohen]

On Wed, Jan 10, 2007 at 09:39:47AM -0700, Administrator wrote:
> Results From cat /proc/zaptel/*
> 
> Span 1: WCTDM/0 "Wildcard TDM2400P Board 1"
> IRQ misses: 24
> 
>1 WCTDM/0/0 FXOKS (In use)
>2 WCTDM/0/1 FXOKS (In use)
>3 WCTDM/0/2 FXOKS (In use)
>4 WCTDM/0/3 FXOKS (In use)
>5 WCTDM/0/4 FXOKS (In use)
>6 WCTDM/0/5 FXOKS (In use)
>7 WCTDM/0/6 FXOKS (In use)
>8 WCTDM/0/7 FXOKS (In use)
>9 WCTDM/0/8 FXOKS (In use)
>   10 WCTDM/0/9 FXOKS (In use)
>   11 WCTDM/0/10 FXOKS (In use)
>   12 WCTDM/0/11 FXOKS (In use)
>   13 WCTDM/0/12 FXOKS (In use)
>   14 WCTDM/0/13 FXOKS (In use)
>   15 WCTDM/0/14 FXOKS (In use)
>   16 WCTDM/0/15 FXOKS (In use)
>   17 WCTDM/0/16 FXOKS (In use)
>   18 WCTDM/0/17 FXOKS (In use)
>   19 WCTDM/0/18 FXOKS (In use)
>   20 WCTDM/0/19 FXOKS (In use)
>   21 WCTDM/0/20 FXOKS (In use)
>   22 WCTDM/0/21 FXOKS (In use)
>   23 WCTDM/0/22 FXOKS (In use)
>   24 WCTDM/0/23 FXOKS (In use)
> Span 2: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" B8ZS/ESF
> 
>   25 WCT1/0/1 Clear (In use)
>   26 WCT1/0/2 Clear (In use)
>   27 WCT1/0/3 Clear (In use)
>   28 WCT1/0/4 Clear (In use)
>   29 WCT1/0/5 Clear (In use)
>   30 WCT1/0/6 Clear (In use)
>   31 WCT1/0/7 Clear (In use)
>   32 WCT1/0/8 Clear (In use)
>   33 WCT1/0/9 Clear (In use)
>   34 WCT1/0/10 Clear (In use)
>   35 WCT1/0/11 Clear (In use)
>   36 WCT1/0/12 Clear (In use)
>   37 WCT1/0/13 Clear (In use)
>   38 WCT1/0/14 Clear (In use)
>   39 WCT1/0/15 Clear (In use)
>   40 WCT1/0/16
>   41 WCT1/0/17 Clear (In use)
>   42 WCT1/0/18 Clear (In use)
>   43 WCT1/0/19 Clear (In use)
>   44 WCT1/0/20 Clear (In use)
>   45 WCT1/0/21 Clear (In use)
>   46 WCT1/0/22 Clear (In use)
>   47 WCT1/0/23 Clear (In use)
>   48 WCT1/0/24 HDLCFCS (In use)

Is it supposed to be T1 or E1?

The card behaves as E1 but you attempt to configure it as T1.

> 
> zaptel.conf file:
> 
> fxoks=1-24
> span=2,1,0,esf,b8zs
> bchan=25-39
> #bchan=40
> bchan=41-47
> dchan=48
> loadzone=us
> defaultzone=us
> 
> James
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
> Sent: Tuesday, January 09, 2007 11:25 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Problem with zaptel drivers or card
> 
> On Tue, Jan 09, 2007 at 06:01:55PM -0700, Administrator wrote:
> > I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 1.4,
> and
> > Zaptel 1.4
> > 
> > The Digium cards installed are TDM2400 and TE110P.
> > 
> > Everything was working fine until I upgraded to zaptel 1.2.12 from 1.2.9
> > 
> > Now when I run ztcfg I get the following error message:
> > 
> > (CAS signalling on span 2 conflicts with Clear channel on channel 40)
> > 
> > --NOTE: signaling was spelled wrong in the error message--
> > 
> > I have since upgraded to 1.4 with the same problem.
> > 
> > Channel 40 is a standard bchan configuration and our provider sees no
> > problem with the channel.
> > 
> > When I disable the channel everything works fine.
> > 
> > My assumption is that something is wrong with the TE110P card.
> > 
> > Has anyone seen anything else like this?
> 
> What do you get from:
> 
> cat /proc/zaptel/*
> 
> What do you have on /etc/zaptel.conf  ?
> 
> -- 
>Tzafrir Cohen   
> icq#16849755jabber:[EMAIL PROTECTED]
> +972-50-7952406   mailto:[EMAIL PROTECTED]   
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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-- 
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+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Proper use of the Local channel

[lists]

Is there any documentation you guys can point us to in order to learn more 
about the proper use of the Local channel? We don't currently use it. However, 
while evaluating other people's billing and management systems for Asterisk, we 
noticed they make extensive use of it.

Thanks,
Daniel

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RE: [asterisk-users] Which H323 module for asterisk

[Dan Austin]

Pavel wrote:
> I prefer h323 included in asterisk tree,
> I have caller id issues with ooh323 and nobody
> answer to bugreports oh323 from inaccessible 
> network is unmaintained/death project, incompatible
> with asterisk 1.4.
> PJ
Response to ooh323c bugs is very slow, and patches can
take some time to be applied if you manage to fix the
issue for yourself.

That said I prefer ooh323c, as it does not require
OpenH323 or PWlib.  I find building it easier.


> Michel wrote:
>> Hello,
>>
>> I need your advice about H323 and asterisk!  ;) 
>> Which one do you advice me to choose H323 
>> (only gateway mode)? ooh323? ooh323c?
Since you mention gateway mode, then ooh323c is 
worth testing.  The bugs that I am aware of are
mostly gatekeeper related (but not all).  Since
the channel doesn't have any external dependencies,
it is the easiest to test.  If it doesn't work for
your setup, there's a very good chance that 
chan_h323 included with Asterisk will and then you
can deal with getting the OpenH323 and PWlib
dependencies meet. (Not a major issue, but one I
have preferred to avoid)

>>
>> Which one is the best H323 module to use with 
>> asterisk? Which one did you choose and why?
>> What is your "return on experience"?
Bugs happen.  I've found that the code for 
chan_ooh323c is reasonably easy to read and
make patches for.  The current release seems
stable and I have it running on four light to
moderately loaded servers.

Dan
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Re: [asterisk-users] Directory too difficult?

[Bryan M. Johns]

I wish had some pearl of wisdom here, but I don't.  I will simply  
share my sympathy.


Sounds like an ESU situation to me.

Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com


On Jan 10, 2007, at 11:10 AM, Colin Anderson wrote:

I have a group of users whos complaint about Asterisk is that the  
directory

application is too hard too use. (yeah, yeah, I know. For the record,
they're Calgarians) Now I'm in a pickle: I don't want to have to  
create a

custom directory for these guys. Anyone have any tips for making the
directory easier, maybe re-record the prompts so they are more  
verbose? We

go by first name.
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Re: [asterisk-users] Zap 1.4 error line 0: Unable to open master device '/dev/zap/ctl'

[Chris Bullock]

Ok. I finally got past this. After doing all the relevant udev stuff, I ran
a make config from the zaptel sources, and got the service to install.

I'm still quiet an asterisk newbie, and defiantly a huge Linux newbie, so
thanks for the help.

-Chris

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[asterisk-users] app_system.c:105 system_exec_helper: Unable to execute '/sbin/zapscan.bin'

[Chris Bullock]

When I load the asterisk 1.4 gui and log into
"/asterisk/static/config/setup/install.html", it tells me "No Analog ports
has been detected on your system".

I have 2 Wildcard X100P cards that are properly installed. Ztcfg shows no
problems.

I also get the following message from the asterisk console "
app_system.c:105 system_exec_helper: Unable to execute '/sbin/zapscan.bin'"
when I log into the web interface.

Any ideas?

-Chris

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Re: [asterisk-users] Asterisk build for Suse 10.1

[Doug Crompton]

Formated your hardisk... wow that is nasty, but I also cannot understand
how that could ever happen. There must be some other HW problem going on
or you got a hold of some really bad code.

What code (source or binary) and what were you doing when that happenned?

Doug

On Wed, 10 Jan 2007, Robert A. Rawlinson wrote:

> Thanks for the help. I was concerned because I tried once before and it
> formatted my hard disk. I wanted to be sure that did not happen again.\
> Bob Rawlinson
>
> On Tue, 2007-01-09 at 11:04 -0500, Robert A. Rawlinson wrote:
> > Has anyone heard of a build or instructions for installing Asterisk on a
> > Suse 10.1 system?
> > Bob Rawlinson
> >
> >
> > ___
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
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k

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[asterisk-users] 1.4 and zap bugs

[Julian Lyndon-Smith]

We're currently running 1.4 r48326 - a little while before the full 1.4 
release.


We are having some problems (crashes) with attended transfers and some 
other stuff, and was going to move to the latest svn 1.4 as I beleive 
that the attended transfer bug has been fixed.


However, I note that #8763 (http://bugs.digium.com/view.php?id=8763) has 
some problems with the zap channels in the 1.4.0 release (which we 
*don't* have).


My question is, are the problems with zap also theoretically present in 
r48326 (it's just that we don't have them) or were they introduced after 
r48326 (and therefore we will have them if we upgrade).


Any thoughts / takers / advice ?

Julian.
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RE: [asterisk-users] Problem with zaptel drivers or card

[Administrator]

Results From cat /proc/zaptel/*

Span 1: WCTDM/0 "Wildcard TDM2400P Board 1"
IRQ misses: 24

   1 WCTDM/0/0 FXOKS (In use)
   2 WCTDM/0/1 FXOKS (In use)
   3 WCTDM/0/2 FXOKS (In use)
   4 WCTDM/0/3 FXOKS (In use)
   5 WCTDM/0/4 FXOKS (In use)
   6 WCTDM/0/5 FXOKS (In use)
   7 WCTDM/0/6 FXOKS (In use)
   8 WCTDM/0/7 FXOKS (In use)
   9 WCTDM/0/8 FXOKS (In use)
  10 WCTDM/0/9 FXOKS (In use)
  11 WCTDM/0/10 FXOKS (In use)
  12 WCTDM/0/11 FXOKS (In use)
  13 WCTDM/0/12 FXOKS (In use)
  14 WCTDM/0/13 FXOKS (In use)
  15 WCTDM/0/14 FXOKS (In use)
  16 WCTDM/0/15 FXOKS (In use)
  17 WCTDM/0/16 FXOKS (In use)
  18 WCTDM/0/17 FXOKS (In use)
  19 WCTDM/0/18 FXOKS (In use)
  20 WCTDM/0/19 FXOKS (In use)
  21 WCTDM/0/20 FXOKS (In use)
  22 WCTDM/0/21 FXOKS (In use)
  23 WCTDM/0/22 FXOKS (In use)
  24 WCTDM/0/23 FXOKS (In use)
Span 2: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" B8ZS/ESF

  25 WCT1/0/1 Clear (In use)
  26 WCT1/0/2 Clear (In use)
  27 WCT1/0/3 Clear (In use)
  28 WCT1/0/4 Clear (In use)
  29 WCT1/0/5 Clear (In use)
  30 WCT1/0/6 Clear (In use)
  31 WCT1/0/7 Clear (In use)
  32 WCT1/0/8 Clear (In use)
  33 WCT1/0/9 Clear (In use)
  34 WCT1/0/10 Clear (In use)
  35 WCT1/0/11 Clear (In use)
  36 WCT1/0/12 Clear (In use)
  37 WCT1/0/13 Clear (In use)
  38 WCT1/0/14 Clear (In use)
  39 WCT1/0/15 Clear (In use)
  40 WCT1/0/16
  41 WCT1/0/17 Clear (In use)
  42 WCT1/0/18 Clear (In use)
  43 WCT1/0/19 Clear (In use)
  44 WCT1/0/20 Clear (In use)
  45 WCT1/0/21 Clear (In use)
  46 WCT1/0/22 Clear (In use)
  47 WCT1/0/23 Clear (In use)
  48 WCT1/0/24 HDLCFCS (In use)

zaptel.conf file:

fxoks=1-24
span=2,1,0,esf,b8zs
bchan=25-39
#bchan=40
bchan=41-47
dchan=48
loadzone=us
defaultzone=us

James

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: Tuesday, January 09, 2007 11:25 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Problem with zaptel drivers or card

On Tue, Jan 09, 2007 at 06:01:55PM -0700, Administrator wrote:
> I have an Asterisk box running Fedora Core 4, Asterisk 1.4, Lippri, 1.4,
and
> Zaptel 1.4
> 
> The Digium cards installed are TDM2400 and TE110P.
> 
> Everything was working fine until I upgraded to zaptel 1.2.12 from 1.2.9
> 
> Now when I run ztcfg I get the following error message:
> 
> (CAS signalling on span 2 conflicts with Clear channel on channel 40)
> 
> --NOTE: signaling was spelled wrong in the error message--
> 
> I have since upgraded to 1.4 with the same problem.
> 
> Channel 40 is a standard bchan configuration and our provider sees no
> problem with the channel.
> 
> When I disable the channel everything works fine.
> 
> My assumption is that something is wrong with the TE110P card.
> 
> Has anyone seen anything else like this?

What do you get from:

cat /proc/zaptel/*

What do you have on /etc/zaptel.conf  ?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Get dialed numbers in AGI

[Anton Frolov]

you could use one of the AGI libraries...
then you can just call a function to get the number.

AF.


Ralph Liebessohn wrote:
> Hi,
> 
> I'm trying to write a AGI in PHP to get the numbers dialed (with
> read()), save it into a variable to insert it into a SQL server
> database. But I cannot see results into the variable, it always return
> NULL.
> Here is a piece of the AGI.
> 
> fwrite(STDOUT,"exec Read my_var|/sound_to_play|5|||15 \n");
> fflush(STDOUT);
> $conn=odbc_connect('MSSQL', 'USER', 'PASS');
> $query = odbc_exec($conn, "INSERT INTO dialed(number) VALUES('$my_var')");
> 
> Even if I only show my_var value or try to use it inside asterisk, the
> value is NULL.
> There is another way to do it? Am I doing a mistake here?
> I'm using Asterisk 1.2.13.
> 
> Thank you all.
> 
> -- 
> Ralph Liebessohn
> ICQ: 74835911
> Skype: liebessohn
> 
> 
> 
> 
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Re: [asterisk-users] Send email notification

[Anselm Martin Hoffmeister]

Am Mittwoch, den 10.01.2007, 11:46 -0400 schrieb H Aranguren:
> Hi group,
> 
> I'm trying to configure the email notification when a user leave a
> voicemail, but don't work (send email notification).
> 
> I configured esmtp in my linux box, if a try to use it with command
> line, it works fine. (echo "Hello" | sendmail [EMAIL PROTECTED] -f [EMAIL 
> PROTECTED]).

You could look wether a voicemail triggers something to happen inside
the mail system at all (logfiles...). In that case, chances are that the
mail cannot be parsed because of misconfiguration in the mail server /
restricted usage of the sendmail -t command or whatever.
In my setup (SMTP server listening on port 25 of the same machine) the
mailcmd is commented out, and It Just Works(tm). If you need mail system
specific help, there sure are lots of forums and info, but I cannot tell
where to connect to esmtp people. Exim is my favourite ;)

> 
> My voicemail.conf
> [general]
> format=wav49
> attach=yes
> [EMAIL PROTECTED]
> fromstring=Asterisk
> mailcmd=/usr/sbin/sendmail -t
> [my_home]
> 100 => ,number100,[EMAIL PROTECTED]
> 
> Can you see the problem?. Do you know any documentarion on internet
> where can i solve the problem?

BR
Anselm

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[asterisk-users] Service Level Compliance

[lists]

Hello all,

We have a slight issue to resolve. We have a client who we are drafting an SLA 
for the delivery of telephony services using Asterisk. Nothing extraordinary. 
However, we do need a way to measure our service availability.

We currently use Nagios and Cacti to monitor server availability as well as 
asterisk and mysql responsiveness, and last, "ping" availability to our 
origination VoIP providers. In an ideal world, this should be fine. However, 
there are a few cases we have noticed this setup not to be enough.

Our particular setup is origination traffic comes into Asterisk box A, where 
the call goes through some AGI-based IVR. After navigating thru the IVR, the 
call is transfered to Asterisk box B, where the call is put in a queue and 
distributed to SIP-based agents.

The issues we would like to resolve are the following:

1) We can ping our originating SIP providers. However, that doesn't guaratee us 
that we can receive calls from them. In several occasions, some of our SIP 
providers have had routing (SIP) problems and when we dial any of the DIDs, 
they would not even hit our box. The call would simply die somewhere in their 
network or their providers' networks. How can we proactively confirm that they 
are actually routing calls to us? We thought we could probably dial out through 
any of our other providers so the call comes in via a different provider and 
maybe hit an AGI script. This script could update a MySQL table with a 
timestamp of the last successful "test". We could then take the data from that 
table and bring it to Nagios and/or Cacti. Is there a better approach?

2) We can test Asterisk responsiveness by doing something like 'asterisk -rx 
"show uptime"' and parse the results. We can also connect to MySQL and execute 
a test query. However, how can we verify that Asterisk is actually talking to 
MySQL and that it's connection hasn't died?

3) As stated above, we can test the responsiveness of asterisk. However, we 
have noticed in, at least, one occasion, that even though asterisk seems to be 
responsive, it would not accept or place any calls. Somehow it's "call" engine 
was locked and we had to restart asterisk. How can we verify that asterisk is 
actually capable of receiving and placing calls?

4) We have no Digium boards and all kernels are 2.6 or above, so we end us 
using ztdummy, if needed. The client's agents are in a different country and 
the average lantency is around 250ms. Most of the time, call quality is good. 
However, there are a few situations where people complaint about echo. Is there 
a way to measure or "improve" this? I know it has been a topic discussed at 
lenght and if we could probably script a way to measure some sort of a MOS 
value, that would be great. Any ideas?

5) Anything else that you could think of we could measure to make sure all 
components are working?

You input is greatly appreciated it. I promise that whatever solution is best 
recommended and scriptable, we will post our development and working solutions 
for the community to benefit from.

Thanks again,
Daniel

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Re: [asterisk-users] how to realize chief - secretary (or Manager - Assistant) setup with Asterisk?

[Anselm Martin Hoffmeister]

Am Mittwoch, den 10.01.2007, 16:37 +0100 schrieb Michael Hamann:
> Hello,
> 
> we are running a Asterisk (1.2) installation with about 80 snom phones
> (300,320,360).
> 
> Now have the demand for a special manager - assistant setup for a few
> extensions.
> 
> Since Shared Line Appearance is not available in 1.2 I´m wondering how
> to realize this...
> 
> What we need is that the manager can decide whether he wants to get
> calls or not. If not he must have the possibility to redirect all
> incoming calls to his secretary. The secretary itself answers all calls
> and decides if the call is important enough to disturb the manager. If
> so she/he transfers the call to the manager. So the secretary can filter
> the calls for the manager...
> 
> The only way I can imagine so far is via a redirect by AstDB on the
> manager extension. The managers phone has two different lines - the
> official and a secret one only the secretary uses...
> 
> Or are there any other solutions?
> 
> Any hint will be appreciated ...

Hello Michael,

as I see it, the most obvious setup would be

- have SIP accounts, e.g. sip123 for the secretary phone, sip456 and
sip789 for the manager phone.
- the "official"/"public" extension number for the manager might be
"4321", so

exten => 4321,1,Dial(SIP/sip123&SIP/sip456)

would ring both the secretary phone and the manager phone on the
"public" id (which most probably can have a separate ringtone than the
"private" id). You would also want a "private" extension like

exten => 4901,1,Dial(SIP/sip789)

for the secretary to reach the manager. A few thoughts:
- The Callerid setting for both secretary and chief should be "4321", no
matter which line the chief chooses to call out through.
- Do not choose an obvious private number, like 4321 and 4322
- You could even choose a "real long" number, that only is available
from internal phones, and put it to a speed dial button on the secretary
phone

If you want the manager to be able to selectively not be disturbed by
"public number" calls, but only by his secretary, some AstDB logic could
come into the game. This can be highly dynamic, or you just configure a
few extensions by hand to do exactly this:

exten => 770/4321,1,Set(DB(list/4321)=SIP/sip123&SIP/sip456)
exten => 770/4321,2,Playback(feature-donotdisturb-off)
exten => 771/4321,1,Set(DB(list/4321)=SIP/sip123)
exten => 771/4321,2,Playback(feature-donotdisturb-on)
exten => 4321,1,Dial(${DB(list/4321)})

So either the chief or the secretary could activate do-not-disturb by
dialing 771, and deactivate with 770. Just examples; choose those codes
from a range that is not in use as extensions; for my personal setup,
the 2*/3*/4*/5*/6* internal numbering for SIP devices, OOH devices, IAX
devices etc.pp., 8* being applications (like 888 the talking clock), 9*
experimental and 0* PSTN calls (how 80's! :-). A somehow similar
function (divert to VoiceMail delay in seconds can be set from any
phone, between 0 and 60 seconds) is available here as 811x.
Choose whatever suits you best.

Of course one could imagine also that the manager phone number NOT rings
the secretary while the manager is there and ready to take calls - just
edit the 770/771 lines (or add 772 for that function) - in that case,
the secretary could make use of an extension number for him/herself, but
her phone also has several lines, so why not.

HTH&BR
Anselm

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[asterisk-users] Directory too difficult?

[Colin Anderson]

I have a group of users whos complaint about Asterisk is that the directory
application is too hard too use. (yeah, yeah, I know. For the record,
they're Calgarians) Now I'm in a pickle: I don't want to have to create a
custom directory for these guys. Anyone have any tips for making the
directory easier, maybe re-record the prompts so they are more verbose? We
go by first name. 
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Re: [asterisk-users] Asterisk build for Suse 10.1

[Robert A. Rawlinson]

Thanks for the help. I was concerned because I tried once before and it
formatted my hard disk. I wanted to be sure that did not happen again.\
Bob Rawlinson

On Tue, 2007-01-09 at 11:04 -0500, Robert A. Rawlinson wrote:
> Has anyone heard of a build or instructions for installing Asterisk on a
> Suse 10.1 system?
> Bob Rawlinson
> 
> 
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Re: [asterisk-users] Adding 4000 Lines to asteriskdb via asterisk -rx ?

[Christoph Adomeit]

> Option A: Use the manager interface.
> 
Tzafrir , Thanks,

the idea to use the manager interface is wonderful. It is really fast
and no data gets lost. I don't think 4000 Rows are a noticeable 
amaount of data for a db1 database.

I coded this:
#!/usr/bin/perl

use Asterisk::Manager;

my $astman = new Asterisk::Manager;

$astman->user('admin');
$astman->secret('bla');
$astman->host('localhost');
$astman->connect || die "Could not connect to " . $astman->host . "!\n";

foreach $num(1..5000) {
  $astman->command("database put callerids willi$num $num");
}

$astman->disconnect;



-- 
Two hours of trial and error can save ten minutes of manual reading.
GATWORKS GmbH
[EMAIL PROTECTED] Internetloesungen vom Feinsten
Fon. +49 2166 9149-32  Fax. +49 2166 9149-10
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Re: [asterisk-users] how to realize chief - secretary (or Manager - Assistant) setup with Asterisk?

[Henry.L.Coleman]

Hi Michael, in practice I think that the managers extension should default
to the assistant who can screen the call or call forward it.
Call Forward - always or Call Forward - no answer would give you the
flexability required.


Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada


> Hello,
>
> we are running a Asterisk (1.2) installation with about 80 snom phones
> (300,320,360).
>
> Now have the demand for a special manager - assistant setup for a few
> extensions.
>
> Since Shared Line Appearance is not available in 1.2 I´m wondering how
> to realize this...
>
> What we need is that the manager can decide whether he wants to get
> calls or not. If not he must have the possibility to redirect all
> incoming calls to his secretary. The secretary itself answers all calls
> and decides if the call is important enough to disturb the manager. If
> so she/he transfers the call to the manager. So the secretary can filter
> the calls for the manager...
>
> The only way I can imagine so far is via a redirect by AstDB on the
> manager extension. The managers phone has two different lines - the
> official and a secret one only the secretary uses...
>
> Or are there any other solutions?
>
> Any hint will be appreciated ...
>
> Michael
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

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[asterisk-users] Get dialed numbers in AGI

[Ralph Liebessohn]

Hi,

I'm trying to write a AGI in PHP to get the numbers dialed (with read()),
save it into a variable to insert it into a SQL server database. But I
cannot see results into the variable, it always return NULL.
Here is a piece of the AGI.

fwrite(STDOUT,"exec Read my_var|/sound_to_play|5|||15 \n");
fflush(STDOUT);
$conn=odbc_connect('MSSQL', 'USER', 'PASS');
$query = odbc_exec($conn, "INSERT INTO dialed(number) VALUES('$my_var')");

Even if I only show my_var value or try to use it inside asterisk, the value
is NULL.
There is another way to do it? Am I doing a mistake here?
I'm using Asterisk 1.2.13.

Thank you all.

--
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn
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[asterisk-users] RTP directly

[David Alcott]

Is there a way to configure the Asterisk so that the RTP goes directly 
between the Endpoints as opposed to going through the asterisk?


-Dave


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[asterisk-users] Send email notification

[H Aranguren]

Hi group,

I'm trying to configure the email notification when a user leave a
voicemail, but don't work (send email notification).

I configured esmtp in my linux box, if a try to use it with command
line, it works fine. (echo "Hello" | sendmail [EMAIL PROTECTED] -f [EMAIL 
PROTECTED]).


My voicemail.conf
[general]
format=wav49
attach=yes
[EMAIL PROTECTED]
fromstring=Asterisk
mailcmd=/usr/sbin/sendmail -t
[my_home]
100 => ,number100,[EMAIL PROTECTED]

My sip.conf
[100]
type=friend
secret=pass
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=internal
[EMAIL PROTECTED]


Can you see the problem?. Do you know any documentarion on internet
where can i solve the problem?

Regards,
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Re: [asterisk-users] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)

[Gary Richardson]

I'd wager to say yes, it does support layer 3 routing :) That's a bit of a
redundant term (though you can route above layer 3). Depending on how many
interfaces you have on your router, you may be sending multiple vlans over a
trunk port (I'm pretty sure the 1600 series support trunk ports -- you may
want to google 'router on a stick').

Most of the layer 3 gigabit switches will still be very expensive, though
Catalyst 3500's may be getting 'cheaper' -- most of the 3500 and 3700 series
switch have multi-gigabit backplanes (usually 16 - 32 gigabits) and can
usually route packets are wire speed, or very close to it. If you are
looking for a gigabit port or two for uplink, I believe they even made a
2900G, though that won't have PoE. And now that I think about it, probably
doesn't support layer 3 routing :(

That's the Cisco world, I'm sure you can find other vendors that have
hardware for much cheaper, though this is an advantage to using the same
networking equipment most other people are using. Also, most of this is
overkill for a handful of network devices.


On 1/10/07, Ed Rubright - mail lists <[EMAIL PROTECTED]> wrote:



Do these 1600 series Cisco routers you mention that you find on eBay for
$50-$150 support Layer3 routing?  I have a managed switch setup on my
home network with several VLANs defined. (work subnet, home subnet, VOIP
subnet)   I currently have to use a Linux box to route between the
VLANs.  I'd like to move to Gigabit routing, but I'd need to replace the
Linux box(more processor power and new NICs) and that gets expensive.

I'd much rather have a router or smart switch for that matter that does
Gigabit Layer3 routing all in one unit.

Do you have any recommendationsthat wouldn't break the bank?

Thanks,
Ed
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