Re: [asterisk-users] Asterisk 1.2.30.3, 1.4.23-rc2, 1.6.0.2, 1.6.1-beta3, and Asterisk-Addons 1.6.0.1, 1.6.1-rc2 released

[Giedrius Augys]

2008/12/3 Tilghman Lesher 

> On Tuesday 02 December 2008 12:22:16 Dave Fullerton wrote:
> > Is anyone else having difficulty compiling 1.6.0.2?
>
> I'll get a new release candidate out either this afternoon or tomorrow;
> I'm currently working on ensuring that 1.6.0.3 will not be a regression
> from
> 1.4.23.
>
> --
> Tilghman
>
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Hi,

 I'm using centos 5.2 and I've successfully compiled asterisk 1.6.0.1 , but
when I want to compile 1.6.0.2. I get :
[CC] manager.c -> manager.o
manager.c: In function 'action_getvar':
manager.c:1732: error: 'SENTINEL' undeclared (first use in this function)
manager.c:1732: error: (Each undeclared identifier is reported only once
manager.c:1732: error: for each function it appears in.)
make[1]: *** [manager.o] Error 1
make: *** [main] Error 2

Have you solved?

-- 
Pagarbiai  / Best Regards,
Giedrius Augys
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Re: [asterisk-users] Directory exists when * is pressed....but where?

[Fred Posner]

On Dec 23, 2008, at 4:49 PM, Mike wrote:

> Thanks, to you and Mark, for the quick reply.  I used to rely on the  
> Wiki
> but it seems I shouldn't
>
> Mike
>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
>> boun...@lists.digium.com] On Behalf Of Tilghman Lesher
>> Sent: Tuesday, December 23, 2008 16:02
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] Directory exists when * is  
>> pressedbut
>> where?
>>
>> On Tuesday 23 December 2008 14:49:52 Mike wrote:
>>> I have been trying to figure out how the * works when in the  
>>> Directory
>>> (dial-by-name).  When I press * (which is supposed to exit the
> directory)
>> I
>>> end up somewhere which I never specified.  It seems like Asterisk  
>>> just
>>> picked that place to go, because I never specified it.
>>
>> When you press '*', it enters the 'a' extension in the dial context
> (second
>> argument to Directory, or first, if the second is not specified).   
>> If the
>> 'a'
>> extension does not exist, then Directory exits normally and the next
>> priority
>> in the current extension is executed.
>>
>> --
>> Tilghman
>>
>>

Anyone have issues with this and 1.6.0.rc3?

Behavior is repetition of menu on * or 0, not using the o or a  
variables.



Fred Posner




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[asterisk-users] DAHDI error

[Jerry Geis]

[Dec 23 17:58:49] ERROR[3091]: chan_dahdi.c:8413 dahdi_pri_error: XXX 
Missing handling for mandatory IE 12 (cs0, Connected Number) XXX


I am seeing the above error on DAHDI 2.1.0, asterisk 1.4.22 and libpri 1.4.7
I am using a TE120P card.

I am also getting this VERY frequently:
  -- Channel 0/1, span 1 got hangup request, cause 16
  -- Hungup 'DAHDI/0-1'

Versus a normal hangup:
  == Spawn extension (smvoice-dialout, smvoice_callprogress, 4) exited 
non-zero on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'

system.conf:
loadzone=us
defaultzone=us

span=1,1,0,esf,b8zs
bchan=1-3
dchan=24

chan_dahdi.conf:
[channels]

switchtype=national
signalling=pri_cpe
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
callerid=asreceived
context=smvoice-incoming
group=1
channel => 1-3



I was using zaptel 1.4.12.1 before using DAHDI and was getting the same 
thing.
Sometimes is appears to work fine - other times I get these SPURIUOS 
hangups.

Jerry

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Re: [asterisk-users] AEL Variable Warning Messages

[Brent Davidson]

Tzafrir Cohen wrote:



What error message from where?

With Zaptel the echo canceller "settings" are global (that is: one
hard-coded echo canceller). With DAHDI there are echo canceller modules
and you can (and actually need to) set them per-channel.

  
It might have something to do with the 
RCBFX drivers, I'm not sure.  I found your page and followed your 
instructions.  Everything appeared to work until I checked with 
dahdi_cfg -vv.  That's where I got the message.  Don't have my notes 
here so I don't have the actual error message right now.

I don't remember the actual error name, but it showed up when I did 
dahdi_cfg -vv.  It was something like "DAHDI_ATTACH_ECHO_CANCELLER 
Failed for Channel 1.  Unsupported command (22)".


I was trying to see if maybe it was logged to my  syslog but this is all 
I find in my /var/log/messages:


Dec 22 17:01:43 localhost modprobe: FATAL: Error inserting 
dahdi_echocan_oslec (/lib/modules/2.6.23.8x86_64/dahdi/dahdi_echocan_osle

c.ko): Unknown symbol in module, or unknown parameter (see dmesg)

Dec 22 17:01:43 localhost kernel: rcbfx 1: Spotted a Rhino: Rhino 
RCB4FXO (4 channels)
Dec 22 17:01:43 localhost kernel: dahdi_echocan_oslec: Unknown symbol 
oslec_create
Dec 22 17:01:43 localhost kernel: dahdi_echocan_oslec: Unknown symbol 
oslec_update
Dec 22 17:01:43 localhost kernel: dahdi_echocan_oslec: Unknown symbol 
oslec_free



Also, when using the Dahdi/Oslec/RCBFX combination I was getting tons of 
blocks like this in my syslog:


Dec 22 16:54:58 localhost kernel: 80c = 2c7e5000
Dec 22 16:54:58 localhost kernel: 810 = 240
Dec 22 16:54:59 localhost kernel: 814 = 0
Dec 22 16:55:00 localhost kernel: 818 = 0
Dec 22 16:55:00 localhost kernel: 81c = 0
Dec 22 16:55:00 localhost kernel: 820 = 
Dec 22 16:55:01 localhost kernel: 824 = 
Dec 22 16:55:01 localhost kernel: 828 = 
Dec 22 16:55:02 localhost kernel: 82c = 0
Dec 22 16:55:02 localhost kernel: 830 = 
Dec 22 16:55:02 localhost kernel: 834 = 
Dec 22 16:55:02 localhost kernel: 838 = 
Dec 22 16:55:03 localhost kernel: 83c = 0
Dec 22 16:55:03 localhost kernel: 840 = 3
Dec 22 16:55:04 localhost kernel: 844 = f
Dec 22 16:55:04 localhost kernel: 848 = 
Dec 22 16:55:04 localhost kernel: 84c = 0
Dec 22 16:55:04 localhost kernel: 850 = 0
Dec 22 16:55:05 localhost kernel: 854 = 10f
Dec 22 16:55:05 localhost kernel: 858 = 14e00ff
Dec 22 16:55:12 localhost kernel: 85c = 3d434310
Dec 22 16:55:13 localhost kernel: 860 = 0
Dec 22 16:55:13 localhost kernel: 864 = 0
Dec 22 16:55:18 localhost kernel: 868 = 229e229e
Dec 22 16:55:18 localhost kernel: 86c = 0
Dec 22 16:55:19 localhost kernel: 870 = 5
Dec 22 16:55:19 localhost kernel: 874 = 5
Dec 22 16:55:20 localhost kernel: 878 = 
Dec 22 16:55:20 localhost kernel: 87c = 0
Dec 22 16:55:21 localhost kernel: 880 = 0
Dec 22 16:55:21 localhost kernel: 884 = 0
Dec 22 16:55:21 localhost kernel: 888 = 0
Dec 22 16:55:22 localhost kernel: 88c = 0
Dec 22 16:55:22 localhost kernel: 890 = 0
Dec 22 16:55:22 localhost kernel: 894 = 0
Dec 22 16:55:24 localhost kernel: 898 = 0
Dec 22 16:55:26 localhost kernel: 89c = 0
Dec 22 16:55:28 localhost kernel: 8a0 = 0
Dec 22 16:55:28 localhost kernel: 8a4 = 0
Dec 22 16:55:28 localhost kernel: 8a8 = 0
Dec 22 16:55:28 localhost kernel: 8ac = 0
Dec 22 16:55:28 localhost kernel: 8b0 = 0
Dec 22 16:55:28 localhost kernel: 8b4 = 0
Dec 22 16:55:28 localhost kernel: 8b8 = 0
Dec 22 16:55:28 localhost kernel: 8bc = 0
Dec 22 16:55:28 localhost kernel: 8c0 = 0
Dec 22 16:55:28 localhost kernel: 8c4 = 0
Dec 22 16:55:28 localhost kernel: 8c8 = 0
Dec 22 16:55:28 localhost kernel: 8cc = 0
Dec 22 16:55:28 localhost kernel: 8d0 = 0
Dec 22 16:55:28 localhost kernel: 8d4 = 0
Dec 22 16:55:28 localhost kernel: 8d8 = 0
Dec 22 16:55:28 localhost kernel: 8dc = 0
Dec 22 16:55:28 localhost kernel: 8e0 = 0
Dec 22 16:55:28 localhost kernel: 8e4 = 0
Dec 22 16:55:28 localhost kernel: 8e8 = 0
Dec 22 16:55:28 localhost kernel: 8ec = 0
Dec 22 16:55:29 localhost kernel: 8f0 = 0
Dec 22 16:55:29 localhost kernel: 8f4 = 0
Dec 22 16:55:29 localhost kernel: 8f8 = 0
Dec 22 16:55:29 localhost kernel: 8fc = 0
Dec 22 16:55:29 localhost kernel: 900 = 0
Dec 22 16:55:29 localhost kernel: 904 = 0
Dec 22 16:55:29 localhost kernel: 908 = 0
Dec 22 16:55:29 localhost kernel: 90c = f0f0f0f
Dec 22 16:55:29 localhost kernel: 910 = f0f0f0f

Switching back to Zaptel solved all of the problems.
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Re: [asterisk-users] Dailplan code for holiday detection?

[Dan Austin]

Tilghman wrote:
> Astdb is a nice idea.  Something along the lines of:

> GotoIf(0${DB(holiday/${STRFTIME(,,%Y-%m-%d)})}?holiday,s,1)

> would work.  Holidays are evaluated as "01", which is true.
> Anything not in the database would be evaluated as "0", which
> is false.  This will work both for holidays where the date
> changes every year (e.g. Thanksgiving, Labor Day), as well as
> holidays where it doesn't (e.g. Christmas, Independence Day).

On one hand I am embarrassed that it is that simple, on the other
I am thrilled that it is that simple.

After the Holidays I guess I need to put together a cheesy
web page to allow for adding the dates to Astdb, but for now
this is awesome and much appreciated.

Thanks,
Dan

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Re: [asterisk-users] AEL Variable Warning Messages

[Tzafrir Cohen]

On Tue, Dec 23, 2008 at 04:01:24PM -0600, Brent Davidson wrote:
> Tzafrir Cohen wrote:
> >On Tue, Dec 23, 2008 at 03:09:51PM -0600, Brent Davidson wrote:
> >
> >  
> >>Unfortunately 1.4.22 no 
> >>longer has Zaptel.  :(
> >>
> >
> >Asterisk 1.4.22 builds with both Zaptel and DAHDI.
> >
> >  
> I spent several hours trying to make it work yesterday and it just 
> wouldn't.  I kept getting an error message that it was unable to bind 
> the echo canceler to channel 1.  

What error message from where?

With Zaptel the echo canceller "settings" are global (that is: one
hard-coded echo canceller). With DAHDI there are echo canceller modules
and you can (and actually need to) set them per-channel.

> It might have something to do with the 
> RCBFX drivers, I'm not sure.  I found your page and followed your 
> instructions.  Everything appeared to work until I checked with 
> dahdi_cfg -vv.  That's where I got the message.  Don't have my notes 
> here so I don't have the actual error message right now.
> 
> 
> 
> -- 
> Brent Davidson
> I.T. Manager
> Texas Country Title Company
> 112 W 2nd / P.O. Box 663
> Cameron, TX 76520
> 254-605-0140 ex. 21
> 

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-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Dailplan code for holiday detection?

[Daniel Hazelbaker]

We chose to use a mySQL database to store the holiday information.   
When a call is answered we query the database to see if there is a  
"holiday" greeting recorded, if so we play the indicated greeting,  
otherwise play the "default" menu greeting. (We do our dialplans in AEL)


context checkHoliday {
 s =>
   {
begin:

 MYSQL(Connect temp communicator username password  
asterisk);
 MYSQL(Query resultid ${temp} SELECT greeting FROM  
menuGreetings WHERE startTime<=FROM_UNIXTIME(${EPOCH}) AND  
endTime>=FROM_UNIXTIME(${EPOCH}) LIMIT 1);
 MYSQL(Fetch foundRow ${resultid} sqlGreeting);
 MYSQL(Clear ${resultid});
 MYSQL(Disconnect ${temp});

 if ("${foundRow}"=="1")
 {
 Background(custom/mainMenu/${sqlGreeting});
 goto mainMenu,s,begin;
 }
 else
 {
 goto checkTime,s,begin;
 }
 }
 includes
   {
mainMenu;
 tempGreeting;
 voicemail;
 publicExt;
 }
};


The 'checkTime' context simply checks if we are open or closed and  
plays the appropriate greeting (if no holiday greeting is found).

Daniel

On Dec 23, 2008, at 1:14 PM, Scott L. Lykens wrote:

> Not the most elegant but since I have a generic context for my IVRs I
> simple check the date there.
>
> exten => s,n,GotoIfTime(*|*|1|jan?closed-holiday|1)
> exten => s,n,GotoIfTime(*|*|10|apr?closed-holiday|1)
> exten => s,n,GotoIfTime(*|*|25|may?closed-holiday|1)
> exten => s,n,GotoIfTime(*|*|3|jul?closed-holiday|1)
> exten => s,n,GotoIfTime(*|*|7|sep?closed-holiday|1)
> exten => s,n,GotoIfTime(*|*|26|nov?closed-holiday|1)
> exten => s,n,GotoIfTime(*|*|27|nov?closed-holiday|1)
> exten => s,n,GotoIfTime(*|*|25|dec?closed-holiday|1)
> exten => s,n,GotoIfTime(*|*|26|dec?closed-holiday|1)
>
> exten =>
> closed-holiday,1,Background(ivr-closed-holiday-${AUTOATTENDANT}||)
> exten => closed-holiday,n,Hangup
>
> This is next year's holidays for us but with this year's Christmas  
> days
> in it.
>
> sl
>
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Re: [asterisk-users] AEL Variable Warning Messages

[Brent Davidson]

Jeff LaCoursiere wrote:

On Tue, 23 Dec 2008, Brent Davidson wrote:

  

Dave Fullerton wrote:


I had gotten similar messages when I forgot to put quotes around
channels like that (took me forever to realize that one). Since you have
them I would say this is a bug. What version of asterisk are you running?

-Dave
  

I'm running 1.4.21.2 and I can't upgrade until Oslec works reliably with
DAHDI and Rhino RCBFX card.  I tried doing a new install with 1.4.22
yesterday and couldn't get Oslec to work correctly with the Rhino card
when running with DAHDI instead of zaptel.  Unfortunately 1.4.22 no
longer has Zaptel.  :(




Why do you need oslec to work with the rhino card - it has hardware echo 
cancellation built in doesn't it?


j
  


The Rhino card is supposed to have hardware echo cancellation.  That's 
one of the main reasons I switched to that card from the X-100p's I was 
using.  Unfortunately, either I don't know how to turn on the hardware 
echo cancellation or it just doesn't work.  I have 5  separate location 
where I'm using that card and if I turn off Oslec at any of them the 
echo is so bad that the systems is virtually unusable.  With Oslec 
enabled, however, there is no echo at all.



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Re: [asterisk-users] Directory exists when * is pressed....but where?

[Mike]

Thanks, to you and Mark, for the quick reply.  I used to rely on the Wiki
but it seems I shouldn't

Mike

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Tilghman Lesher
> Sent: Tuesday, December 23, 2008 16:02
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Directory exists when * is pressedbut
> where?
> 
> On Tuesday 23 December 2008 14:49:52 Mike wrote:
> > I have been trying to figure out how the * works when in the Directory
> > (dial-by-name).  When I press * (which is supposed to exit the
directory)
> I
> > end up somewhere which I never specified.  It seems like Asterisk just
> > picked that place to go, because I never specified it.
> 
> When you press '*', it enters the 'a' extension in the dial context
(second
> argument to Directory, or first, if the second is not specified).  If the
> 'a'
> extension does not exist, then Directory exits normally and the next
> priority
> in the current extension is executed.
> 
> --
> Tilghman
> 
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Re: [asterisk-users] AEL Variable Warning Messages

[Brent Davidson]

Tzafrir Cohen wrote:

On Tue, Dec 23, 2008 at 03:09:51PM -0600, Brent Davidson wrote:

  
Unfortunately 1.4.22 no 
longer has Zaptel.  :(



Asterisk 1.4.22 builds with both Zaptel and DAHDI.

  
I spent several hours trying to make it work yesterday and it just 
wouldn't.  I kept getting an error message that it was unable to bind 
the echo canceler to channel 1.  It might have something to do with the 
RCBFX drivers, I'm not sure.  I found your page and followed your 
instructions.  Everything appeared to work until I checked with 
dahdi_cfg -vv.  That's where I got the message.  Don't have my notes 
here so I don't have the actual error message right now.




--
Brent Davidson
I.T. Manager
Texas Country Title Company
112 W 2nd / P.O. Box 663
Cameron, TX 76520
254-605-0140 ex. 21

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Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual

[Atis Lezdins]

On Tue, Dec 23, 2008 at 11:17 PM, Kristian Kielhofner
 wrote:
>  This is true, however, I wasn't very excited about any other debug
> messages that might get printed with "debug 1".  I knew I only needed
> the endpoint RTP address, so I just removed the if.  Of course you
> could always just run with "debug 1" instead of the patch too.
>
> Again, this modification isn't strictly required.  I just did if for
> SIP providers that give unpredictable media endpoint IP addresses...
> :)

Debug 1 isn't that much. Just grep for line you're using and
everything should work fast and fine. Sometimes i even log our
production servers for weeks with debug 1. So i would suggest
submiting this modification to digium bugtracker, if it really helps
tracking ip's.

Thanks again,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] AEL Variable Warning Messages

[Tzafrir Cohen]

On Tue, Dec 23, 2008 at 03:09:51PM -0600, Brent Davidson wrote:

> Unfortunately 1.4.22 no 
> longer has Zaptel.  :(

Asterisk 1.4.22 builds with both Zaptel and DAHDI.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual

[Kristian Kielhofner]

On Tue, Dec 23, 2008 at 4:02 PM, Atis Lezdins  wrote:
>
> Hi,
>
> This is good idea, and i will probably try it out someday next year
> (too busy completing my business requirements :)

  Luckily next year is just over a week away.  We won't have to wait
that long ;).

> I took a look at asterisk patch, and it seems quite simple. I just
> don't see the point of removing "if(debug)". You could easily get this
> additional logging into Asterisk trunk (if preserving RTP info in
> debug level), and starting asterisk with "debug 1". So, then it would
> be easier to install recqual. Also, being able to run on unmodified
> version of Asterisk, it would be good to allow keeping current
> dialplan and just route test calls trough it. So, people would be able
> to keep track of their billing, etc for those test calls.

  This is true, however, I wasn't very excited about any other debug
messages that might get printed with "debug 1".  I knew I only needed
the endpoint RTP address, so I just removed the if.  Of course you
could always just run with "debug 1" instead of the patch too.

  Again, this modification isn't strictly required.  I just did if for
SIP providers that give unpredictable media endpoint IP addresses...
:)

> Also, thanks for showing us magics of ecasound. I have similar project
> (pbx-test-framework) that allows IVR/Queue/etc testing in automated
> mode. Recording everything and checking voice quuailty would be great
> addition :)

  Ecasound is very, very cool.  Recqual is only scratching the surface
of what it can do!

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

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Re: [asterisk-users] Dailplan code for holiday detection?

[Scott L. Lykens]

Not the most elegant but since I have a generic context for my IVRs I
simple check the date there.

exten => s,n,GotoIfTime(*|*|1|jan?closed-holiday|1)
exten => s,n,GotoIfTime(*|*|10|apr?closed-holiday|1)
exten => s,n,GotoIfTime(*|*|25|may?closed-holiday|1)
exten => s,n,GotoIfTime(*|*|3|jul?closed-holiday|1)
exten => s,n,GotoIfTime(*|*|7|sep?closed-holiday|1)
exten => s,n,GotoIfTime(*|*|26|nov?closed-holiday|1)
exten => s,n,GotoIfTime(*|*|27|nov?closed-holiday|1)
exten => s,n,GotoIfTime(*|*|25|dec?closed-holiday|1)
exten => s,n,GotoIfTime(*|*|26|dec?closed-holiday|1)

exten =>
closed-holiday,1,Background(ivr-closed-holiday-${AUTOATTENDANT}||)
exten => closed-holiday,n,Hangup

This is next year's holidays for us but with this year's Christmas days
in it.

sl

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Re: [asterisk-users] AEL Variable Warning Messages

[Jeff LaCoursiere]

On Tue, 23 Dec 2008, Brent Davidson wrote:

> Dave Fullerton wrote:
>> I had gotten similar messages when I forgot to put quotes around
>> channels like that (took me forever to realize that one). Since you have
>> them I would say this is a bug. What version of asterisk are you running?
>>
>> -Dave
> I'm running 1.4.21.2 and I can't upgrade until Oslec works reliably with
> DAHDI and Rhino RCBFX card.  I tried doing a new install with 1.4.22
> yesterday and couldn't get Oslec to work correctly with the Rhino card
> when running with DAHDI instead of zaptel.  Unfortunately 1.4.22 no
> longer has Zaptel.  :(
>

Why do you need oslec to work with the rhino card - it has hardware echo 
cancellation built in doesn't it?

j

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Re: [asterisk-users] AEL Variable Warning Messages

[Brent Davidson]

Dave Fullerton wrote:
> I had gotten similar messages when I forgot to put quotes around 
> channels like that (took me forever to realize that one). Since you have 
> them I would say this is a bug. What version of asterisk are you running?
>
> -Dave
I'm running 1.4.21.2 and I can't upgrade until Oslec works reliably with 
DAHDI and Rhino RCBFX card.  I tried doing a new install with 1.4.22 
yesterday and couldn't get Oslec to work correctly with the Rhino card 
when running with DAHDI instead of zaptel.  Unfortunately 1.4.22 no 
longer has Zaptel.  :(

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Re: [asterisk-users] Directory exists when * is pressed....but where?

[Mark Michelson]

Mike wrote:
> I have been trying to figure out how the * works when in the Directory 
> (dial-by-name).  When I press * (which is supposed to exit the 
> directory) I end up somewhere which I never specified.  It seems like 
> Asterisk just picked that place to go, because I never specified it.
> 
>  
> 
> The wiki is no help on that one…
> 
>  
> 
>  
> 
> Mike
> 

If you look at the help text for the Directory application using the Asterisk 
CLI (core show application directory), it specifies that pressing the '*' key 
will send you to the 'a' extension if it exists.

Mark Michelson

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Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual

[Atis Lezdins]

On Mon, Dec 22, 2008 at 5:37 PM, Kristian Kielhofner
 wrote:
> Hey everyone,
>
>   A while back I worked on a project to measure call quality.  I've
> finally gotten around to releasing it and I'm calling it recqual (Real
> Call Quality).  There isn't much to it and it should be considered
> alpha quality.  I'm hoping some of the bright minds on the list can
> help me out with it.  I'll include the intro text from the README in
> the tarball:
>
> 
> Recqual is collection of scripts using Asterisk and other
> Linux utilities to measure call quality on an automated
> basis.
>
> -How it Works-
> Recqual was designed to detect audio quality problems in a call
> path that may not be visible from the technology being used
> locally.  Whether it's SIP, ZAP, or IAX the fact is there are
> many potential sources of call quality problems in just about
> any call being made.  Often times a SIP "provider" may resell
> services, for example.  While the delivery of IP packets to/from
> this provider may look excellent there may be other problems
> upstream that an analysis of the IP packets, path, etc may not
> be able to detect.
>
> In scenarios such as this the only way to identify call quality
> problems is to analyze the audio itself.  Regardless of method
> or transport being used, the goal of any telephony system is to
> deliver reliable, consistent call quality.
>
> Recqual is designed to allow you to place a large number of
> automated calls (using Asterisk) using different call scenario
> files.
>
> The key here is consistency.  When Asterisk places the outbound
> call (and answers the inbound call) it will generate a set of
> tones while recording the return audio path.  Once the run has
> finished Ecasound will run with various filters and noise gates
> to detect certain amounts of distortion, signal loss, etc.
>
> Calls either pass or fail based on how much variation there is
> in the audio once it has been returned.  Of course you can pass
> audio through any combination of networks - including the PSTN.
>
> Almost any "call quality" problem(s) can be detected with this
> method.  Whether it's one way calls, echo, dropped packets,
> distortion, etc ecasound should be able to isolate the problem
> calls.  If not you can just tweak the script ;).
>
> Only calls that fail are saved.  These files can be imported
> into your favorite audio processing utility and/or run through
> Ecasound again if you'd like to tweak the process script to
> detect them automatically.
>
> Recqual has been designed (and optimized) to work with SIP
> channels.  For example, it has the ability to correlate problem
> calls with specific RTP endpoint IP addresses.  However, due to
> the protocol independent nature of Asterisk you can use just
> about any channel type with a few simple changes.
> ---
>
>  So there you have it.  I've used this with a great deal of success
> but I think there is still a lot to be done.  More on my blog here:
>
> http://blog.krisk.org
>
>  Thoughts?
>

Hi,

This is good idea, and i will probably try it out someday next year
(too busy completing my business requirements :)

I took a look at asterisk patch, and it seems quite simple. I just
don't see the point of removing "if(debug)". You could easily get this
additional logging into Asterisk trunk (if preserving RTP info in
debug level), and starting asterisk with "debug 1". So, then it would
be easier to install recqual. Also, being able to run on unmodified
version of Asterisk, it would be good to allow keeping current
dialplan and just route test calls trough it. So, people would be able
to keep track of their billing, etc for those test calls.

Also, thanks for showing us magics of ecasound. I have similar project
(pbx-test-framework) that allows IVR/Queue/etc testing in automated
mode. Recording everything and checking voice quuailty would be great
addition :)

Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
a...@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] Directory exists when * is pressed....but where?

[Tilghman Lesher]

On Tuesday 23 December 2008 14:49:52 Mike wrote:
> I have been trying to figure out how the * works when in the Directory
> (dial-by-name).  When I press * (which is supposed to exit the directory) I
> end up somewhere which I never specified.  It seems like Asterisk just
> picked that place to go, because I never specified it.

When you press '*', it enters the 'a' extension in the dial context (second
argument to Directory, or first, if the second is not specified).  If the 'a'
extension does not exist, then Directory exits normally and the next priority
in the current extension is executed.

-- 
Tilghman

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[asterisk-users] Directory exists when * is pressed....but where?

[Mike]

I have been trying to figure out how the * works when in the Directory
(dial-by-name).  When I press * (which is supposed to exit the directory) I
end up somewhere which I never specified.  It seems like Asterisk just
picked that place to go, because I never specified it.

 

The wiki is no help on that one…

 

 

Mike

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Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual

[Kristian Kielhofner]

On Tue, Dec 23, 2008 at 2:59 PM, Mindaugas Kezys  wrote:
>
> Looks very interesting. After reading all available info I have two
> questions before testing:
>
> 1. Who/what answers the calls at the other end? I guess real live traffic
> should be sent through this Asterisk server?
> 2. How many calls you had made to to diagnose your problems?
>
> Thank you.
>
> Regards,
> Mindaugas Kezys
> http://www.kolmisoft.com
> VoIP Billing and Routing Solutions
>

Mindaugas,

  Thanks.

  1)  The same Asterisk system places the call and answers the call.
You can use one technology to place the call and another to answer
(but you don't have to).  One connection is the control, the other is
the variable.  For instance, in my experiments I was testing SIP
providers against one another and each other.  I placed outbound calls
to the SIP provider to a DID I had on a local PRI that came back into
the same system running recqual.  Recqual then (as designed) played
audio in both directions while recording, like this:

outbound call (to SIP provider, dialing DID on PRI below):
<- (audio in from inbound leg)
--> (audio out to outbound leg - not recorded, should always be perfect)

Inbound call (from PRI):
<- (audio in from outbound leg)
--> (audio out to inbound leg - not recorded, should always be perfect)

  I know what the audio on both legs should look like so all I need to
do is analyze it once it is returned to me.  That way I can see how it
changed from what Asterisk was supposed to be generating in the first
place.  This is how recqual is able to detect such a wide range of
quality problems.

  2)  Depends on what you are testing for.  For example, if you know
you have a bad pair in a bundle of analog lines, you could just call
all of the numbers a couple of times.  Whichever number/pair has the
worst audio (fails the test or looks the worst) is the bad pair.  In
the case of my SIP provider, they used outbound load balancing to send
traffic to multiple media gateways and multiple trunk groups all over
the country.  Because we had no control over these routing decisions
we had to make periodic calls for at least a day to detect the
majority of the bad hosts/trunks/etc.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

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Re: [asterisk-users] AEL Variable Warning Messages

[Dave Fullerton]

Brent Davidson wrote:
> Philipp Kempgen wrote:
>> Brent Davidson schrieb:
>>
>>  
>>> macro outside-dial ( num ) {
>>>   if (${DB_EXISTS(Office/${CALLERID(num)})}) {
>>> TRUNK="Zap/r2";
>>>   } else {
>>> TRUNK="Zap/r1";
>>>   }
>>>   Dial(${TRUNK}/${num},,Ttok);
>>> }
>>> 
>>
>>  
>>> [Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item: 
>>> Warning: file /etc/asterisk/extensions.ael, line 93-93: expression 
>>> "Zap/r2" has operators, but no variables. Interesting...
>>> [Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item: 
>>> Warning: file /etc/asterisk/extensions.ael, line 95-95: expression 
>>> "Zap/r1" has operators, but no variables. Interesting...
>>> 
>>
>> I'd suggest
>> Set(TRUNK=Zap/r2);
>> resp.
>> Set(TRUNK=Zap/r1);
>>
>>
>>Philipp Kempgen
>>
>>   
> 
> According to the AEL Documentation I should be able to set variables 
> without using the "Set" command.  They even give the following example:
> 
> context foo {
>555 => {
> x=5;
> y=blah;
> divexample=10/2
> NoOp(x is ${x} and y is ${y} !);
>};
> };
> 
> I wonder if maybe AEL is ignoring the double quotes and treating the 
> Zap/r2 as if it were division???  Should I file a bug report on this?
> 

I had gotten similar messages when I forgot to put quotes around 
channels like that (took me forever to realize that one). Since you have 
them I would say this is a bug. What version of asterisk are you running?

-Dave

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Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual

[Mindaugas Kezys]

On Mon, Dec 22, 2008 at 10:37:01AM -0500, Kristian Kielhofner wrote:
> Hey everyone,
> 
>A while back I worked on a project to measure call quality.  I've
> finally gotten around to releasing it and I'm calling it recqual (Real
> Call Quality).  There isn't much to it and it should be considered
> alpha quality.  I'm hoping some of the bright minds on the list can
> help me out with it.  I'll include the intro text from the README in
> the tarball:


Looks very interesting. After reading all available info I have two
questions before testing:

1. Who/what answers the calls at the other end? I guess real live traffic
should be sent through this Asterisk server?
2. How many calls you had made to to diagnose your problems?

Thank you.

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions


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Re: [asterisk-users] AEL Variable Warning Messages

[Brent Davidson]

Philipp Kempgen wrote:

Brent Davidson schrieb:

  

macro outside-dial ( num ) {
  if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK="Zap/r2";
  } else {
TRUNK="Zap/r1";
  }
  Dial(${TRUNK}/${num},,Ttok);
}



  
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item: 
Warning: file /etc/asterisk/extensions.ael, line 93-93: expression 
"Zap/r2" has operators, but no variables. Interesting...
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item: 
Warning: file /etc/asterisk/extensions.ael, line 95-95: expression 
"Zap/r1" has operators, but no variables. Interesting...



I'd suggest
Set(TRUNK=Zap/r2);
resp.
Set(TRUNK=Zap/r1);


   Philipp Kempgen

  


According to the AEL Documentation I should be able to set variables 
without using the "Set" command.  They even give the following example:


context foo {
   555 => {
x=5;
y=blah;
divexample=10/2
NoOp(x is ${x} and y is ${y} !);
   };
};

I wonder if maybe AEL is ignoring the double quotes and treating the 
Zap/r2 as if it were division???  Should I file a bug report on this?



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Re: [asterisk-users] Pattern Matching

[Brent Davidson]

Philipp Kempgen wrote:

Brent Davidson schrieb:
  
On my asterisk system, if an incoming call only has a number for the 
caller ID and no name, the system is using the channel name as in the 
Callerid Name field.  I would like to use some sort of pattern match 
test to test for the presence of "Zap/" in the ${CALLERID(name)} 
variable and if it is present, replace it with "Unknown".  I'm using the 
ael format for my dialplan and have been looking for a way to do this, 
but haven't found anything yet.  Is there a way to do this inside the 
dialplan



if ("${CALLERID(name):0:4}" = "Zap/") {
Set(CALLERID(name)=Unknown);
}

Not sure why you would want to put the channel name into the
caller ID name in the first place.


   Philipp Kempgen

  


Thanks all.  As far as why the channel name is in the caller ID, I don't 
know.  I'm certainly not doing it intentionally.  I don't have any code 
in the dialplan that even touches the CallerID, so I guess Asterisk is 
doing somehow when the Name part of the CallerID is unknown...  Either 
that or my Snom 300 phones are picking the wrong info to use for CallerID.
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Re: [asterisk-users] AEL Variable Warning Messages

[Philipp Kempgen]

Brent Davidson schrieb:

> macro outside-dial ( num ) {
>   if (${DB_EXISTS(Office/${CALLERID(num)})}) {
> TRUNK="Zap/r2";
>   } else {
> TRUNK="Zap/r1";
>   }
>   Dial(${TRUNK}/${num},,Ttok);
> }

> [Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item: 
> Warning: file /etc/asterisk/extensions.ael, line 93-93: expression 
> "Zap/r2" has operators, but no variables. Interesting...
> [Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item: 
> Warning: file /etc/asterisk/extensions.ael, line 95-95: expression 
> "Zap/r1" has operators, but no variables. Interesting...

I'd suggest
Set(TRUNK=Zap/r2);
resp.
Set(TRUNK=Zap/r1);


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] Dailplan code for holiday detection?

[Tilghman Lesher]

On Tuesday 23 December 2008 12:11:41 Dan Austin wrote:
> This has been on my ToDo list far too long.
>
> I have a small call-center setup, with basic
> time of day/day of week validation before putting
> callers in the queues.
>
> With the holidays upon us, I need to add check to
> see if 'today' is a holiday so I do not put callers
> in unmanned queues.  Due to how the agents work, I have
> to allow joinwhenempty.
>
> Does anyone have a snippet of dialplan code, perhaps using
> Astdb, to check it 'today' is a listed holiday?

Astdb is a nice idea.  Something along the lines of:

GotoIf(0${DB(holiday/${STRFTIME(,,%Y-%m-%d)})}?holiday,s,1)

would work.  Holidays are evaluated as "01", which is true.  Anything not
in the database would be evaluated as "0", which is false.  This will work
both for holidays where the date changes every year (e.g. Thanksgiving,
Labor Day), as well as holidays where it doesn't (e.g. Christmas, Independence
Day).

-- 
Tilghman

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Re: [asterisk-users] Pattern Matching

[Philipp Kempgen]

Brent Davidson schrieb:
> On my asterisk system, if an incoming call only has a number for the 
> caller ID and no name, the system is using the channel name as in the 
> Callerid Name field.  I would like to use some sort of pattern match 
> test to test for the presence of "Zap/" in the ${CALLERID(name)} 
> variable and if it is present, replace it with "Unknown".  I'm using the 
> ael format for my dialplan and have been looking for a way to do this, 
> but haven't found anything yet.  Is there a way to do this inside the 
> dialplan

if ("${CALLERID(name):0:4}" = "Zap/") {
Set(CALLERID(name)=Unknown);
}

Not sure why you would want to put the channel name into the
caller ID name in the first place.


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] Pattern Matching

[Tilghman Lesher]

On Tuesday 23 December 2008 11:47:08 Brent Davidson wrote:
> On my asterisk system, if an incoming call only has a number for the
> caller ID and no name, the system is using the channel name as in the
> Callerid Name field.  I would like to use some sort of pattern match
> test to test for the presence of "Zap/" in the ${CALLERID(name)}
> variable and if it is present, replace it with "Unknown".  I'm using the
> ael format for my dialplan and have been looking for a way to do this,
> but haven't found anything yet.  Is there a way to do this inside the
> dialplan or do I have to pass it out to an AGI script?

Set(CALLERID(name)=${IF($[${CUT(CALLERID(name),/,1)}=Zap]?Unknown:
${CALLERID(name)})})


-- 
Tilghman

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[asterisk-users] AEL Variable Warning Messages

[Brent Davidson]

I have two offices sharing a phone system.  They also share a common 
internal context because all of the employees of the second office also 
work for the first office.  Each office has 4 outside lines and I have 
defined 2 channel groups in my zapata.conf.  The second office needs all 
of their outgoing calls to go out over their lines so the people they 
call will have the correct callerID.  I created an asterisk database and 
with entries in the database for all extensions in the second office and 
defined the following macro:

globals {
  CONSOLE="Console/dsp";
  TRUNK="Zap/r1";
  TCTC_Operator=15;
  Law_Operator=12;
};

macro outside-dial ( num ) {
  if (${DB_EXISTS(Office/${CALLERID(num)})}) {
TRUNK="Zap/r2";
  } else {
TRUNK="Zap/r1";
  }
  Dial(${TRUNK}/${num},,Ttok);
}

It's working and correctly routing outside calls, but I get the 
following messages when I reload the extensions.ael file:

[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item: 
Warning: file /etc/asterisk/extensions.ael, line 93-93: expression 
"Zap/r2" has operators, but no variables. Interesting...
[Dec 23 12:16:22] WARNING[2994]: pbx_ael.c:2500 check_pval_item: 
Warning: file /etc/asterisk/extensions.ael, line 95-95: expression 
"Zap/r1" has operators, but no variables. Interesting...

Any idea what is causing the warnings?

Thanks,
Brent

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[asterisk-users] Dailplan code for holiday detection?

[Dan Austin]

This has been on my ToDo list far too long.

I have a small call-center setup, with basic
time of day/day of week validation before putting
callers in the queues.

With the holidays upon us, I need to add check to
see if 'today' is a holiday so I do not put callers
in unmanned queues.  Due to how the agents work, I have
to allow joinwhenempty.

Does anyone have a snippet of dialplan code, perhaps using
Astdb, to check it 'today' is a listed holiday?

Thanks,
Dan

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[asterisk-users] Pattern Matching

[Brent Davidson]

On my asterisk system, if an incoming call only has a number for the 
caller ID and no name, the system is using the channel name as in the 
Callerid Name field.  I would like to use some sort of pattern match 
test to test for the presence of "Zap/" in the ${CALLERID(name)} 
variable and if it is present, replace it with "Unknown".  I'm using the 
ael format for my dialplan and have been looking for a way to do this, 
but haven't found anything yet.  Is there a way to do this inside the 
dialplan or do I have to pass it out to an AGI script?

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Re: [asterisk-users] why does users.conf generate SIP peer and SIP user?

[Kristian Kielhofner]

On Tue, Dec 23, 2008 at 4:40 AM, Steve Totaro
 wrote:
> "It's all ball bearings these days"
>

What is the deal with Fletch quotes these days?  Don't get me wrong, I
appreciate them but I'm starting to wonder where this is all coming
from.

I *think* it's because Fletch has been on HBO lately.  Am I correct?

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

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Re: [asterisk-users] IMAP Voicemail and Directory not working?

[Tzafrir Cohen]

On Tue, Dec 23, 2008 at 10:13:12AM -0500, Noah Miller wrote:
> Hi Tzafrir -
> 
> >> I'm wondering if anybody has IMAP Voicemail AND the directory working
> >> together.  I haven't had any success.  IMAP voicemail works fine, but
> >> when it's active, the Directory does not work.  The problem seems to
> >> be with libc-client.  Specifically, asterisk is not able to access the
> >> mm_dlog function.
> >>
> >> I've tried with Asterisk 1.4.22+ and 1.6.0+ using CentOS 5.2, Ubuntu
> >> 8.10 and Fedora 9.  In each case, I used the native package manager to
> >> install libc-client, and in each case, after asterisk is compiled and
> >> voicemail users are configured, I get an error in the log that says
> >> this:
> >
> > On Ubuntu and Debian (Lenny/Sid) -
> >
> >  apt-get source asterisk
> >  # as root / using sudo:
> >  apt-get build-dep asterisk
> >  cd asterisk-1
> >  ASTERISK_NO_DOCS=yes fakeroot debian/rules build
> >
> > Does it build? If so, you have a similar version of Asterisk that builds
> > with IMAP support.
> 
> I finally got this to work.  For some reason, none of the packaged
> versions of libc-client from any of the distributions I tried support
> mm_dlog, which is required by the Directory app.  I ended up compiling
> from uw-imap's source on Ubuntu, and that worked right away.  

http://packages.ubuntu.com/intrepid/asterisk
http://packages.ubuntu.com/intrepid/amd64/asterisk/filelist

app_voicemail_imap.so is included . 

I built several newer versions on my Lenny system and never encountered
such an issue.

Hmm but in my 1.6.0 copy app_voicemail_imap.so doesn't have mm_dlog
as well.

It came from http://svn.digium.com/svn/asterisk/branches/1...@134223
("Merging the imap_consistency branch.")

-- 
   Tzafrir Cohen
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Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual

[Kristian Kielhofner]

On Tue, Dec 23, 2008 at 6:31 AM, Tzafrir Cohen  wrote:
> On Mon, Dec 22, 2008 at 10:37:01AM -0500, Kristian Kielhofner wrote:
>> Hey everyone,
>>
>>A while back I worked on a project to measure call quality.  I've
>> finally gotten around to releasing it and I'm calling it recqual (Real
>> Call Quality).  There isn't much to it and it should be considered
>> alpha quality.  I'm hoping some of the bright minds on the list can
>> help me out with it.  I'll include the intro text from the README in
>> the tarball:
>
> You seem to have ommited the relevant links:
>
> http://blog.krisk.org/2008/12/introducing-recqual.html
> http://admin.star2star.com/recqual/
>
> Looks interesting...
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

Tzafrir,

  I included a link to my blog in the original post but it probably
wasn't as clear as it could have been.

  Looks interesting?  Thanks!  I'd love to see it deployed more, I've
only used it in a few specific scenarios.

-- 
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com

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Re: [asterisk-users] No Audio

[michel freiha]

Dear Sir,

I used several other Softphones like Skype and they are facing the same
problem...It seems that the issue is global du to an undersea cable cut
Regards

On Mon, Dec 22, 2008 at 9:07 PM, michel freiha  wrote:

> Hi all,
> Sometimes when making a PC to PSTN call through asterisk, I got no audio in
> both sides...tracing by wireshark, I can find that RTP packets are hitting
> my PC but no audio...Can someone guess what could be that issue?
>
> Maybe it's a latency issue?
>
> Regards
>
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Re: [asterisk-users] IMAP Voicemail and Directory not working?

[Noah Miller]

Hi Tzafrir -

>> I'm wondering if anybody has IMAP Voicemail AND the directory working
>> together.  I haven't had any success.  IMAP voicemail works fine, but
>> when it's active, the Directory does not work.  The problem seems to
>> be with libc-client.  Specifically, asterisk is not able to access the
>> mm_dlog function.
>>
>> I've tried with Asterisk 1.4.22+ and 1.6.0+ using CentOS 5.2, Ubuntu
>> 8.10 and Fedora 9.  In each case, I used the native package manager to
>> install libc-client, and in each case, after asterisk is compiled and
>> voicemail users are configured, I get an error in the log that says
>> this:
>
> On Ubuntu and Debian (Lenny/Sid) -
>
>  apt-get source asterisk
>  # as root / using sudo:
>  apt-get build-dep asterisk
>  cd asterisk-1
>  ASTERISK_NO_DOCS=yes fakeroot debian/rules build
>
> Does it build? If so, you have a similar version of Asterisk that builds
> with IMAP support.

I finally got this to work.  For some reason, none of the packaged
versions of libc-client from any of the distributions I tried support
mm_dlog, which is required by the Directory app.  I ended up compiling
from uw-imap's source on Ubuntu, and that worked right away.  On the
Red Hat varieties, compiling from source worked, but I had to specify
-fPIC and a few other compiler flags when building UW's c-client.

For the record, if anybody needs to do this on a redhat platform:

1. Download imap-2007e (or latest version) from
ftp://ftp.cac.washington.edu/imap/
2. Unpack and compile with a make command like:

 make  SSLTYPE=none EXTRACFLAGS="-DIGNORE_LOCK_EACCES_ERRORS=1 \
 -I/usr/include/openssl -fPIC -fno-strict-aliasing -Wall
-Wno-pointer-sign -Wno-parentheses"

  (See the Makefile for a list of platforms - I used 'lr5' for CentOS 5.2)

3. In the asterisk source, run the configure script with the imap flag:

./configure --with-imap=/path/to/imap-source

  (use the base directory of the imap source - e.g. /usr/src/imap-2007e )

4. Run "make menuselect" for asterisk and select "IMAP_STORAGE" from
the Voicemail Build Options.


Of course, you'll also need an appropriately configured IMAP server
(for CentOS, I recommend their default choice of Dovecot).


- Noah

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Re: [asterisk-users] Ghost in the Channel-Banks

[Jerry Jones]

On Dec 22, 2008, at 10:38 PM, Martin Lima wrote:

> On Thursday 18 December 2008, Justin Phelps wrote:
>> I've been struggling with an ongoing problem the last month.
>>
>> Here is the layout of the wiring:
>> T1 from ISP > DiTech Echo Cancel device > Voice Channel-Bank
>>
>> (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1  
>> server
>> zap card > fax channel bank
>>
>> (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1  
>> server
>> zap card > asterisk2 server
>>
>> Now, let me explain the symptoms.
>>
>> d-channel errors on the asterisk1 server on span1 (which is the line
>> coming from the echo cancel from the ISP). asterisk2 server isn't  
>> being
>> used as far as I can tell. I've got a red alarm on the port on  
>> asterisk1
>> that asterisk2 is plugged into.
>
> I would bet your "asterisk2" server was meant for some kind of  
> transition to a
> different setup. Is there at least some dialplan inside? sip.conf?  
> iax,
> voicemail etc...?
Something does not fit here. If you have the T1 from the ISP going to  
the echo can, then it cannot go to more than one device. It is not a  
MUX as far as I remember, have used to great effect in past. Each T1  
in matches up to a T1 out.
>
>
>> faxes through the fax channel bank are working most of the time.  
>> There
>> seems to be problems with multipage faxes. Not isolated to a  
>> particular
>> port on the channel bank. As I understand it, echo cancel for faxes  
>> is a
>> bad thing, so I don't understand why the previous IT Admin here setup
>> the system as such.
> Some echo cancellers detect a fax transmission and turn echo  
> cancellation off.
> In your case if I understand your setup correctly such a device is  
> ambiguous
> at the best... Possibly it used to be another set of voice lines  
> converted to
> fax without changes in configuration?
> I dont really understand the role of echo canceller on E1/T1...

Guarantee the Ditech deteects fax and cancels echo can. However any IP  
in FAX path is problematic for most FAX. If possible turn off ECM. If  
symptoms are mainly on a single fax machine make sure you have enough  
memory to buffer entire fax. Some systems let you print as they come  
in, but I have also seen some which are still trying to buffer while  
printing and run over. This showed up with a mortgage company doing 50  
page legal forms. First 20 or so were fine then started bombing.
Best method I have found to troubleshooting FAX is to use a machine  
which generates a T30 trace output upon completion.

>
>
>> Some voice phones on the voice channel banks were not recognizing  
>> tones
> Why the phone should recognize tones? It just generated them while  
> dialing.
>
>> when dialing. That seems to have been resolved after power cycling  
>> the
>> channel bank a few times, and restarting asterisk2 (odd that there
>> doesn't seem to be anything active on asterisk2)
> Looks like some strange ground loops in your wiring, power issue or  
> something
> similar.
>
>> I've been working with the ISP on the d-channel stuff, and things  
>> seem
>> to get a little better as they reset equipment, but the d-channel  
>> errors
>> have not gone away.
> Some as above...
> BTW they will always tell you they "reset something" no matter what  
> they have
> really done... :-)

Again if this is a PRI, how is signalling being done for 2 T1  
connections when only one D Channel. I suspect a partial PRI?
>
>
>> I really need advice on these problems. From what I've said, where do
>> you think the problem lies? in the channel banks? in the echo  
>> canceller?
>> in the asterisk2 or asterisk1 server? With the ISP?
> Hard to guess without deeper knowledge of your setup. Intermittent  
> errors and
> hardware lockups are often caused by power conditions, potential  
> differences
> and spikes in the powerline. (two pieces of equipment connected  
> together but
> plugged into two different power outlets coming from two opposite  
> ends of the
> building can cause real headache!) Check it first. Then you may want  
> to
> continue what your former IT admin tried to start :-)

Are D Channels mostly working just intermittant? I assume you have  
Ditech set for passthrough on these channels? ie no echo cancel?
>
>
> Martin
>
>
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[asterisk-users] second trunk in extensions.conf

[Nick Wolf]

I have a TE210P digium card that has 2 E1/T1 ports.

the code in my extensions.conf file for span 1 is  :

[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=Zap/g1; Trunk interface
TRUNKX=Zap/g2   ; 2nd trunk interface
...
...
; dial a long distance outbound number to SPAIN
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls,
exten => _00034X,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _00034X,2,Dial(${TRUNK}/${EXTEN:1},55,To)
exten => _00034X,3,Hangup
...
...
and it works fine, but I need to start working with my second span & I don't
know how to add it in extensions.conf file.
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Re: [asterisk-users] Manager API - standardization?

[Wesley Haut]

> Isn't there one already?

Yeah, but none of them have worked for me...maybe their way of doing things
is just different from my approach but I wasn't happy with any of the
existing classes.  I wasn't planning on releasing my code to the wild (I'm
not a programmer by trade I just play one on TV).

> A *Complete even is sent as and end of a series of events. e.g. SIPPeers
> retuns a separate event for each peer and hence you need an event to
> tell you that the series is done. For Login, Ping and whatever you know
> you should not expect further events to follow after the initial reply.

Um, not exactly.  I just ran a SIPPeers on the CLI via telnet and each of
the PeerEntry events does not have a Complete flag, just the double return.
I guess what I was after was a standard way to finalize the larger event as
a whole, and keep the existing double line return for separating the
"individual" events.  Again, maybe I'm going about this the wrong way as
well.

> For a larger audince of such proposals, try the -dev list .
> Be sure to refer to the trunk version. Note, however, that any change to
> the manager interface may break existing programs.

Thanks, I'll wait until I've gotten further along on my project before
bothering the devs

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Re: [asterisk-users] Outbound fax issues

[Mikel Lindsaar]

On Wed, Dec 24, 2008 at 12:02 AM, Barry L. Kline wrote:

> > What does putting ww at the front do?
> Each w makes Asterisk wait a 1/2 second before sending the DTMF to dial.
> (It may be a 1/4 second each 'w')


I thought so, in that case, it is not the problem here.

My problem is that the fax dials out, connects, but can't handle some
perceived "line noise"

Need to work out or find out what the best settings are for a fax machine
connected to a VOIP device per my diagram previously...

Mikel
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Re: [asterisk-users] Good comparisons on cheaper VOIP phones

[Mikel Lindsaar]

On Tue, Dec 23, 2008 at 11:01 PM, Yehavi Bourvine  wrote:

> We have one 7912 which we bought for evaluation. The main drawback is that
> it has hands free speaker but no microphone.
>

That's true. But we will be getting higher models for the speaker function.

Did you find or know of a way to do paging with the Cisco 7912G ?

Looking around on Google didn't come up with much.

Mikel
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Re: [asterisk-users] Outbound fax issues

[Barry L. Kline]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Mikel Lindsaar wrote:

> 
> What does putting ww at the front do?

Each w makes Asterisk wait a 1/2 second before sending the DTMF to dial.
(It may be a 1/4 second each 'w')

Barry
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Version: GnuPG v1.4.5 (GNU/Linux)

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0kJs1pvYx9rYIoLOys4eQTs=
=cTza
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Re: [asterisk-users] regarding query registered or online users fro out of asterisk

[Godson Gera]

On Tue, Dec 23, 2008 at 5:23 PM, yavuzhan canli wrote:

> Hi all,
>
> anyone have any experience regarding query whether our sip accounts
> registered (online) or not registered (offline) from out of asterisk with
> mysql or another tool. My goal is taking this information with query and
> put it to my intranet to check my users.
>
> any help would be appreciated


you can use sip show peers console command, or use SIPPees manager Action,
in peerentry if you find IPaddress as none that account is offline other
wise online.

http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+SIPpeers

-- 

Godson Gera
http://godson.in/voip-asterisk-consultant-hyderabad-india
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Re: [asterisk-users] Outbound fax issues

[Mikel Lindsaar]

On Tue, Dec 23, 2008 at 1:28 AM, Danny Nicholas  wrote:

>  What does your extensions.conf look like for this call?  If you can
> insert a ww into your Dial command (ie, change 18005551212 to ww18005551212)
> this may improve your dialing behavior.
>
> In an attempt to isolate the problem, I reduced the extensions.conf to:

; Fax Lines
exten => _.,1,Dial(${AAPT}${EXTEN},,R)


What does putting ww at the front do?

Regards

Mikel
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Re: [asterisk-users] Good comparisons on cheaper VOIP phones

[Yehavi Bourvine]

 We have one 7912 which we bought for evaluation. The main drawback is that
it has hands free speaker but no microphone.

__Yehavi:

2008/12/23 Mikel Lindsaar 

> Thanks all for your replies.
> I have an aastra 9133i here for testing and am getting a polycom 320 to try
> out.
>
> But today, I got my hands on an older Cisco 7912G with SIP software
> installed.  It connected fine to the Asterisk box, works with the PoE stuff
> I have, sounds good and doesn't seem to have any problems.  Best all, I can
> buy "near new" for about $60 each in Australian dollars (thats about 45USD
> with the Aussie dollar being what it is :)
>
> The handsets look OK, they are nice and solid feeling and very easy to use
> / not complex.
>
> Any reason not to use the 7912G ?  Seems with the SIP image they work just
> dandy...
>
>
> Mikel
>
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Re: [asterisk-users] outging ---asterisk -bug

[Tzafrir Cohen]

On Tue, Dec 23, 2008 at 09:06:26AM +0100, Stefan Schmidt wrote:
> 
> 
> jordan pan schrieb:
> >
> > Hi everyone,
> >
> > when i use the automated dial out,I found that once the zap
> > answerd,the contex will be exectued, but i don't hope do it ,i hope
> > when extern phone answered ,then ,the context will be exectued.
> > Anyone can help me solve the problem!
> > the call file is:
> > Channel: Zap/g0/15015895665
> > Context: myivr
> > RetryTime: 60
> > MaxRetries: 2
> > Waittime: 60
> > Extension: 808
> > Priority: 1
> > Callerid: 15015895665
> >
> > [myivr]
> > exten => s,1,Background(test)
> > exten => s,n,WaitExten

Where is extension 808 there?

What do you see in the CLI trace there? Try: core set verbose 3'
beforehand.

> you should have a look at originate CLI command. you can call using
> asterisk -rx "originate Zap/g0/1501589... Background(test).

You meant:

  asterisk -rx "originate Zap/g0/15015895665 application Background test"

-- 
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[asterisk-users] regarding query registered or online users fro out of asterisk

[yavuzhan canli]

Hi all,

anyone have any experience regarding query whether our sip accounts
registered (online) or not registered (offline) from out of asterisk with
mysql or another tool. My goal is taking this information with query and
put it to my intranet to check my users.

any help would be appreciated

Yavuzhan




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Re: [asterisk-users] Good comparisons on cheaper VOIP phones

[Mikel Lindsaar]

Thanks all for your replies.
I have an aastra 9133i here for testing and am getting a polycom 320 to try
out.

But today, I got my hands on an older Cisco 7912G with SIP software
installed.  It connected fine to the Asterisk box, works with the PoE stuff
I have, sounds good and doesn't seem to have any problems.  Best all, I can
buy "near new" for about $60 each in Australian dollars (thats about 45USD
with the Aussie dollar being what it is :)

The handsets look OK, they are nice and solid feeling and very easy to use /
not complex.

Any reason not to use the 7912G ?  Seems with the SIP image they work just
dandy...


Mikel
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Re: [asterisk-users] IMAP Voicemail and Directory not working?

[Tzafrir Cohen]

On Mon, Dec 22, 2008 at 03:30:02PM -0500, Noah Miller wrote:
> Hi All -
> 
> I'm wondering if anybody has IMAP Voicemail AND the directory working
> together.  I haven't had any success.  IMAP voicemail works fine, but
> when it's active, the Directory does not work.  The problem seems to
> be with libc-client.  Specifically, asterisk is not able to access the
> mm_dlog function.
> 
> I've tried with Asterisk 1.4.22+ and 1.6.0+ using CentOS 5.2, Ubuntu
> 8.10 and Fedora 9.  In each case, I used the native package manager to
> install libc-client, and in each case, after asterisk is compiled and
> voicemail users are configured, I get an error in the log that says
> this:

On Ubuntu and Debian (Lenny/Sid) - 

  apt-get source asterisk
  # as root / using sudo:
  apt-get build-dep asterisk
  cd asterisk-1
  ASTERISK_NO_DOCS=yes fakeroot debian/rules build

Does it build? If so, you have a similar version of Asterisk that builds
with IMAP support.

(ASTERISK_NO_DOCS=yes is merely intended to save you some unnecessary 5
minutes and ~80MB of generating the API docs)

-- 
   Tzafrir Cohen
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+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] why does users.conf generate SIP peer and SIP user?

[Tzafrir Cohen]

On Tue, Dec 23, 2008 at 10:35:19AM +0100, Klaus Darilion wrote:
> Hi!
> 
> I wonder why users.conf generates a SIP user and a SIP peer? Why is it 
> not possible to set type=... in users.conf? (Asterisk 1.4.22)

users.conf is a hack to generate a "typical" Asterisk configuration
easily.

So I figure that patches are welcomed. Assuming that the "type" won't
have any strange ill-effects on any other module that parses that file:

A quick grep in trunk gives:

  apps/app_directory.c
  apps/app_voicemail.c
  channels/chan_agent.c
  channels/chan_dahdi.c
  channels/chan_h323.c
  channels/chan_iax2.c
  channels/chan_sip.c
  main/manager.c
  pbx/pbx_config.c
  res/res_phoneprov.c

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Re: [asterisk-users] Using Asterisk to measure call quality: Introducing Recqual

[Tzafrir Cohen]

On Mon, Dec 22, 2008 at 10:37:01AM -0500, Kristian Kielhofner wrote:
> Hey everyone,
> 
>A while back I worked on a project to measure call quality.  I've
> finally gotten around to releasing it and I'm calling it recqual (Real
> Call Quality).  There isn't much to it and it should be considered
> alpha quality.  I'm hoping some of the bright minds on the list can
> help me out with it.  I'll include the intro text from the README in
> the tarball:

You seem to have ommited the relevant links:

http://blog.krisk.org/2008/12/introducing-recqual.html
http://admin.star2star.com/recqual/

Looks interesting...

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Re: [asterisk-users] Manager API - standardization?

[Tzafrir Cohen]

On Mon, Dec 22, 2008 at 09:04:13AM -0600, Wesley Haut wrote:
> Hi all,
> 
> I know I'm probably stirring up a hornet's nest with this question/comment
> but I've spent the last few days working on a PHP-based class for the
> manager interface 

Isn't there one already?

> as we're preparing for a pretty big upgrade at our call
> center and I'm revamping all of the management apps I've written.  I  can
> connect to the manager interface and send query strings back and forth all
> day long, but I'm having issues when it comes to parsing the return data.
> My issue isn't a PHP one, but rather issues with the consistency of the
> Manager interface return values.
> 
> For instance, for the built in actions such as Login, SIPPeers, QueueStatus
> (the three that I will be calling the most) NONE of those three has a
> standard return value.  Login doesn't return an Event: LoginComplete flag,
> so the login function can't use the generic Action function I've written.

A *Complete even is sent as and end of a series of events. e.g. SIPPeers
retuns a separate event for each peer and hence you need an event to
tell you that the series is done. For Login, Ping and whatever you know
you should not expect further events to follow after the initial reply.

> SIPPeers returns an additional line after Event: PeerlistComplete, and
> Peerlist != SIPPeers (meaning the action and eventcomplete flags should
> match, IMHO).
> 
> The way QueueStatus returns data is the ideal, the action flag (QueueStatus)
> has a corresponding complete tag (QueueStatusComplete) and I can (fairly)
> easily parse the return data by tacking 'Complete' onto the action flag.  I
> just wanted to see what everyone else though of coming up with a standard
> for ALL manager commands so building applications to hook into Asterisk
> isn't a crap shoot like it is now.
> 
> My initial proposal:
> 
> Any $action has a corresponding $actionComplete event (ie SIPPeers would be
> SIPPeersComplete instead of PeerlistComplete)
> The ActionComplete event is the LAST line for any return
> 
> Like I said, I'm sure this will ruffle some feathers and that is not my
> intent, maybe I'm missing something here so please (kindlly) inform me if I
> am.

For a larger audince of such proposals, try the -dev list .
Be sure to refer to the trunk version. Note, however, that any change to
the manager interface may break existing programs.

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Re: [asterisk-users] Install app_rxfax and app_txfax in 1.4with Lenny

[Tzafrir Cohen]

On Mon, Dec 22, 2008 at 10:46:46AM +0100, Olivier wrote:
> Hi Andrew,
> 
> 2008/12/22 Andrew Thomas 
> 
> > JFYI - I run (successfully) agx-addons with 1.4.22 and Etch.
> >
> > Make sure you have the right version of SpanDSP installed (as well as the
> > tiff libraries).
> 
> 
> which are (thinking of both SpanDSP and libiff) ?

libtiff, I guess.

$ rmadison libspandsp-dev
libspandsp-dev | 0.0.2pre26-1 | etch-m68k | m68k
libspandsp-dev | 0.0.2pre26-1 |stable | alpha, amd64, arm, hppa, i386, 
ia64, mips, mipsel, powerpc, s390, sparc
libspandsp-dev | 0.0.5~pre4-1 |   testing | alpha, amd64, arm, armel, hppa, 
i386, ia64, mips, mipsel, powerpc, s390, sparc
libspandsp-dev | 0.0.5~pre4-1 |  unstable | alpha, amd64, arm, armel, hppa, 
hurd-i386, i386, ia64, mips, mipsel, powerpc, s390, sparc

The SpanDSP one in Etch is too old and the one in Lenny is too new.
That said, I really don't see why people stick with that old spandsp
0.0.4pre16 . There have been bug fixes since. Not to mention some
crashes. This is why I bothered backporting spandsp from Asterisk 1.6.0
(app_fax.c is also a better behaving Asterisk module. e.g. does not keep
its own personal log files).

-- 
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Re: [asterisk-users] Setup ReceiveFax(), fax2mail, mime-construct - but now Sendmail :(

[Tzafrir Cohen]

On Mon, Dec 22, 2008 at 09:33:18AM -, Andrew Thomas wrote:
> You don't really need to use any local MTA if you use the sendEmail
> script.
> 
> I got it from - http://www.caspian.dotconf.net/menu/Software/SendEmail/

Which is essentially the same as using ssmtp  / esmtp / nullmailer 
(non-queuing sendmail replacements). 

The mentioned prorams actually give you the same command-line interface
as sendmail, and hence require less (if at all) change  of
voicemail.conf.

> 
> This actually works by 'talking' directly to any SMTP server - even
> remote ones (I use our Exchange server for our e-mails).

And this is a reminder: they don't queue mail. Hence if they fail to
deliver once, the mail is lost. May not be the best idea for sending
mail over the internet.

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Re: [asterisk-users] why does users.conf generate SIP peer and SIP user?

[Steve Totaro]

"It's all ball bearings these days"

On Tue, Dec 23, 2008 at 4:35 AM, Klaus Darilion
 wrote:
> Hi!
>
> I wonder why users.conf generates a SIP user and a SIP peer? Why is it
> not possible to set type=... in users.conf? (Asterisk 1.4.22)
>
> thanks
> klaus
>
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[asterisk-users] why does users.conf generate SIP peer and SIP user?

[Klaus Darilion]

Hi!

I wonder why users.conf generates a SIP user and a SIP peer? Why is it 
not possible to set type=... in users.conf? (Asterisk 1.4.22)

thanks
klaus

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Re: [asterisk-users] Good comparisons on cheaper VOIP phones

[Steve Totaro]

On Tue, Dec 23, 2008 at 1:48 AM, Yehavi Bourvine
 wrote:
> I have one ST2030 bought for testing. Indeed it has a very intuitive user's
> interface, bue I've found two drawbacks:
>
> Its sound quality has some place to be improved...
> It has no RPID support (displaying the name of the called party).
>
> If these two issues are fixed, then it might be the better choice
> for cheaper price.
>
>   __Yehavi:
>
> 2008/12/21 Olivier 
>>
>> I don't know if Thomson ST2030 SIP phones are distributed where you live
>> but those have the best feature set-price ratio.
>> They integrate smoothly with Asterisk (one touch pickup, BLF, MWI, ...)
>> with up to 5 simultaneous calls.
>>
>> Here in France, those are selected everywhere ...
>>
>> I would recommend them without any hesitation.
>>
>> ___


While feeling cheap, the BT101 seemed reasonable during testinng.  It
was the day to day customer issues that made me go back and replace
with Polycom for hardware cost only, no labor.

I guess you have to define "testing".

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Re: [asterisk-users] Good comparisons on cheaper VOIP phones

[Yehavi Bourvine]

I'll have to run some TCPDUMP to see what happens. I'll also try this with
OpenSIPS where there is more flexbility with the header fields.

   Thanks, __Yehavi:

 2008/12/23 Olivier 

>
>
> 2008/12/23 Yehavi Bourvine 
>
>>  I have one ST2030 bought for testing. Indeed it has a very intuitive
>> user's interface, bue I've found two drawbacks:
>>
>>- Its sound quality has some place to be improved...
>>- It has no RPID support (displaying the name of the called party).
>>
>> Hi,
>
> Have you tried P-Asserted-Identity features (from latest Asterisk and
> ST2030) ?
> I
> 'm hoping that using this feature, you can have your ST2030 displaying the
> name of the person you're calling : whenever, Asterisk is acking the INVITE
> message it received from the ST2030, it should include the name of called
> party.
>
> To be honest, I really don't know how you can teach Asterisk to behave this
> way (ie to include called party name in ACK replies) but I'm sure the phone
> support displaying names from trusted parties.
>
> It's on my TODO list, anyway, as I would also like to let this phone
> display caller's name when picking up a call (instead of the dialed string).
>
> Regards
>
>
>>-
>>
>> If these two issues are fixed, then it might be the better choice
>> for cheaper price.
>>
>>   __Yehavi:
>>
>> 2008/12/21 Olivier 
>>
>>> I don't know if Thomson ST2030 SIP phones are distributed where you live
>>> but those have the best feature set-price ratio.
>>> They integrate smoothly with Asterisk (one touch pickup, BLF, MWI, ...)
>>> with up to 5 simultaneous calls.
>>>
>>> Here in France, those are selected everywhere ...
>>>
>>> I would recommend them without any hesitation.
>>>
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>>>
>>
>>
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>
>
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Re: [asterisk-users] Good comparisons on cheaper VOIP phones

[Olivier]

2008/12/23 Yehavi Bourvine 

> I have one ST2030 bought for testing. Indeed it has a very intuitive user's
> interface, bue I've found two drawbacks:
>
>- Its sound quality has some place to be improved...
>- It has no RPID support (displaying the name of the called party).
>
> Hi,

Have you tried P-Asserted-Identity features (from latest Asterisk and
ST2030) ?
I
'm hoping that using this feature, you can have your ST2030 displaying the
name of the person you're calling : whenever, Asterisk is acking the INVITE
message it received from the ST2030, it should include the name of called
party.

To be honest, I really don't know how you can teach Asterisk to behave this
way (ie to include called party name in ACK replies) but I'm sure the phone
support displaying names from trusted parties.

It's on my TODO list, anyway, as I would also like to let this phone display
caller's name when picking up a call (instead of the dialed string).

Regards


>-
>
> If these two issues are fixed, then it might be the better choice
> for cheaper price.
>
>   __Yehavi:
>
> 2008/12/21 Olivier 
>
>> I don't know if Thomson ST2030 SIP phones are distributed where you live
>> but those have the best feature set-price ratio.
>> They integrate smoothly with Asterisk (one touch pickup, BLF, MWI, ...)
>> with up to 5 simultaneous calls.
>>
>> Here in France, those are selected everywhere ...
>>
>> I would recommend them without any hesitation.
>>
>> ___
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>>
>> 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] outging ---asterisk -bug

[Stefan Schmidt]

jordan pan schrieb:
>
> Hi everyone,
>
> when i use the automated dial out,I found that once the zap
> answerd,the contex will be exectued, but i don't hope do it ,i hope
> when extern phone answered ,then ,the context will be exectued.
> Anyone can help me solve the problem!
> the call file is:
> Channel: Zap/g0/15015895665
> Context: myivr
> RetryTime: 60
> MaxRetries: 2
> Waittime: 60
> Extension: 808
> Priority: 1
> Callerid: 15015895665
>
> [myivr]
> exten => s,1,Background(test)
> exten => s,n,WaitExten
>
> Thanks in advance!
> -- 
> Best regards!
> jordan pan
> Location:Shenzhen China
> Company:www.justcall.cn 
> 
>
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Hello,

you should have a look at originate CLI command. you can call using
asterisk -rx "originate Zap/g0/1501589... Background(test).

and maybe you should try an answer before starting the background in myivr.

best regards.

steve smith

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-
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