Re: [asterisk-users] Is it possible to get full callin number from E1?

[MaxGao]

thanks all
 i found the telco only send me the normal number 87654321 
 i just want to start a fax service and people can direct dial some extend 
num like 87654321...but it never send to me ... so the only thing i can do 
is to provide a ivr and let the people enter the extend num then jump to fax.
 thanks alot ~



在2009-03-13 03:42:54，"D Tucny"  写道：

2009/3/12 ssmax 
Hi all

   i have just set up a asterisk in china, using DE410P and one E1 line
   and get a phone number like: +86 020 87654321 from my sp
   when somebody dial  +86 020 87654321 , the asterisk will get the call in 
number by ${EXTEN} variable, but it can only get 87654321, no area code .

As others have mentioned in this thread, this is pretty normal... Telco's all 
have different policies on how many digits they will pass for the dialled 
number, some may let you specify, but often each telco has an internal 
standard... You don't need the rest of the number, but if you're doing 
something where it would be useful to have it, you can easily add the prefix 
yourself... It's also worth pointing out at this point, the prefix is either 
020 or +86-20 for Guangzhou depending on whether it's written for national or 
international use, the 0 is not dialled internationally...
 

   when someone dial  +86 020 87654321 ,  means 4 digits,  the phone 
can call in, and the  ${EXTEN}  is only  87654321 too ,
   is it possible to get  full call in number 87654321  in asterisk ?  
thanks

 
This is confusing... Guangzhou numbers are only 8 digits long + prefixes, so, 
+86 20 87654321 would be the number... Where do you get '' from? What would 
be more normal would be that you'd by DDIs with your ISDN30 service, such that 
you'd have for example, a range of 100 numbers say from 87654000 to 87654099, 
though you can buy more... With this, the number dialled (or the last 8 digits 
in your case) would be accessible through the ${EXTEN} variable... I guess it 
could be possible that your telco is attempting to offer some other type of 
service that allows them to issue you a single number, but, that they will 
accept as many as four additional digits that they will pass to you in some 
way, but, it's impossible to say what they are doing here and how it's been 
implemented... I would further suspect that if this is the case that there 
would be a very distinct chance that this would only work with inbound calls 
from other users of the same telco, and in China, in the same city...

Within China, telco interoperability can be flaky at best and, from experience, 
the fact the telco engineers don't entirely understand their huawei switches 
for local problems and for problems over a wider area, the fact that the 
national telcos are split into provincial and city organisations that don't 
appear to communicate without going through some circuitous route via the 
telco's head office both lead to some very long fix times if there ever is a 
fix... That said, as long as you have cash, you can get them to provide almost 
any service you want, though they won't often offer anything but the most basic 
service, you have to ask for it...


d
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] an easy way to deal with/without leading "1" ?

[Cary Fitch]

In my previous reply, I may be wrong, "877" is probably a valid toll free
NPA, add it in the mix.

Cary Fitch

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Thursday, March 12, 2009 7:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] an easy way to deal with/without leading "1" ?

I posted this before, but it didn't show up. So if it's a dup...

I'm setting up dialplans to deal with 800 dialing through a different
channel than regular long distance in the US.

The regular long distance is set up so users can but don't have to
dial one. That's pretty easy, just one more exten statement. But it's
a pain dealing with all the 8xx area codes that are toll free.

I tried

exten=>exten => _!877NXX,2,Dial(${pstnline}/ww1${EXTEN:-10})

but that matches everything. I'd hoped it would only match strings
that had zero or more characters, followed by the 877 pattern.

sean

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] an easy way to deal with/without leading "1" ?

[Cary Fitch]

I am not sure of the complexity of your issue.

There are only 4 toll free NPAs.  800, 888, 877, 866.
Any thing starting with them is toll free. So, filtering them is not a major
task, it would take only 4 statements at most and could be done with 2.

I don't program on a daily basis... so this is a pseudo programming example.

Set a variable: NPA = CalledNum[1-3]
Then parse it with 
If (NPA = "800" | NPA = "866" | NPA = "877" | NPA = "888"), then
dowhatyouwannado

Where the "|" is an "or".

Now make that real asterisk code and you can filter the toll free calls.

We use similar code to allow local calls to Overlaid NPAs like 713, 282 and
832.

Cary Fitch 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Thursday, March 12, 2009 7:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] an easy way to deal with/without leading "1" ?

I posted this before, but it didn't show up. So if it's a dup...

I'm setting up dialplans to deal with 800 dialing through a different
channel than regular long distance in the US.

The regular long distance is set up so users can but don't have to
dial one. That's pretty easy, just one more exten statement. But it's
a pain dealing with all the 8xx area codes that are toll free.

I tried

exten=>exten => _!877NXX,2,Dial(${pstnline}/ww1${EXTEN:-10})

but that matches everything. I'd hoped it would only match strings
that had zero or more characters, followed by the 877 pattern.

sean

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] MOH Realtime

[Sebastian]

I've finally got it working on 1.6.0.6, but it seems to be a problem:

 

 

Situation:

 

Queue realtime musiconhold class = prueba

I have default class on musiconhold.conf

When a call is made to the Queue checks that is not on memory so goes to the
db, I had 2 situations:

 

1)  If digit was #. Asterisk crash

2)  With no digit, when I check the classes

 

 

localhost*CLI> moh show classes

Class: default 

Mode: files

Directory: /var/lib/asterisk/moh

Class: 

Mode:
e,clid,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposi
tion,amaflags,accountcode,uniqueid,userfield,test) VALUES ('2009-03-12
20:48:11','"1001"
<1001>','outgoing','SIP/1001-08a490a8','','Queue','Pedidos,tTc,,,120,,CTI,,'
,'2','2','ANSWERED','DOCUMENTATION','','1236901691.0','','')

Directory: 

Application: 

Format: unknown

[Mar 12 20:48:15] ERROR[3007]: astobj2.c:115 INTERNAL_OBJ: bad magic number
0x0 for 0x8a50640

[Mar 12 20:48:15] ERROR[3007]: astobj2.c:115 INTERNAL_OBJ: bad magic number
0x0 for 0x8a50640

 

Seems bad memory management, any idea??

 

Asterisk version 1.6.0.6

 

Regards,

 

Sebasitan

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4)

[David Ruggles]

The patch doesn't work for me. Here's what I did:

Changed to my asterisk-1.4.23.1 directory
Executed the wget / patch command from the link you provided
> make
>> saw that res_features.so was recompiled
Moved /usr/lib/asterisk/modules/res_features.so to res_features.so.old
> make install
Confirmed that res_features.so was recreated in /usr/lib/asterisk/modules
> asterisk -r   --  I never shut asterisk down
> module unload res_features.so
> module load res_features.so

After this there was no change, it worked using the macro but using the
Set(DYN... on the caller only.

Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200  da...@safedatausa.com



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson
Sent: Thursday, March 12, 2009 8:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trying to get sample applicationmap to work
(*1.4)


David Ruggles wrote:
> Wow!
> 
> Thanks! That's a very clear answer and completely understandable. Is this
> something I should open a bug report on?
> 
> Thanks,

Nope, I've already got that taken care of.

http://bugs.digium.com/view.php?id=14657

There's a patch there that I have tested and "it works for me (TM)." See if
it 
works out for you, too.

Mark Michelson

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
No virus found in this incoming message.
Checked by AVG - www.avg.com 
Version: 8.0.237 / Virus Database: 270.11.10/1996 - Release Date: 03/12/09
10:38:00


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] an easy way to deal with/without leading "1" ?

[sean darcy]

I posted this before, but it didn't show up. So if it's a dup...

I'm setting up dialplans to deal with 800 dialing through a different
channel than regular long distance in the US.

The regular long distance is set up so users can but don't have to
dial one. That's pretty easy, just one more exten statement. But it's
a pain dealing with all the 8xx area codes that are toll free.

I tried

exten=>exten => _!877NXX,2,Dial(${pstnline}/ww1${EXTEN:-10})

but that matches everything. I'd hoped it would only match strings
that had zero or more characters, followed by the 877 pattern.

sean

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4)

[Mark Michelson]

David Ruggles wrote:
> Wow!
> 
> Thanks! That's a very clear answer and completely understandable. Is this
> something I should open a bug report on?
> 
> Thanks,

Nope, I've already got that taken care of.

http://bugs.digium.com/view.php?id=14657

There's a patch there that I have tested and "it works for me (TM)." See if it 
works out for you, too.

Mark Michelson

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4)

[David Ruggles]

Wow!

Thanks! That's a very clear answer and completely understandable. Is this
something I should open a bug report on?

Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200  da...@safedatausa.com



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson
Sent: Thursday, March 12, 2009 7:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trying to get sample applicationmap to work
(*1.4)


David Ruggles wrote:
> I don't really think that's a problem, because I'm able to use the other
> built in options: *1 to record; ## transfer (I changed this from a single
> pound) and there have been a couple times that I wouldn't hit them quickly
> enough.
> 
> Thanks,

Ah, sorry about that. The featuredigittimeout burns so many people that it's

pretty much a knee-jerk reaction on my part now to suggest that as a
potential fix.

To test out, I set up the same feature and gave it a try with a current 
subversion checkout of Asterisk 1.4. I placed a call from SIP/2001 to
SIP/2000 
and here's what I found.

When SIP/2001 pressed #9, tt-monkeys played on SIP/2000's channel
When SIP/2000 pressed #9, nothing happened.

I tried modifying the features.conf line to have "peer/callee" instead of
just 
"peer" and that caused neither side to successfully use the dynamic feature.

There appears to be a bug which does not allow for the callee to use dynamic

features. The problem appears to be that when DTMF is pressed, we try to 
interpret the presses to determine if there is a corresponding feature. The 
DYNAMIC_FEATURES variable has been set on the caller's channel, but has not
been 
set on the callee's channel. As a result, we don't properly read the value
of 
the DYNAMIC_FEATURES variable if the callee is the one to press DTMF.

You can work around the bug, although it's not exactly optimal. What you can
do 
is to modify your dialplan as follows:

exten => 301,n,Set(DYNAMIC_FEATURES=monkey)
exten => 301,n,Dial(SIP/DavidR1,,M(dynamic_features))

[macro-dynamic_features]
exten => s,1,Set(DYNAMIC_FEATURES=monkey)

By doing this, the dynamic_features macro will be called on SIP/DavidR1 when
he 
answers. This will allow for the DYNAMIC_FEATURES variable to be set on both

channels so both sides can use the feature you have set.

This is a bug, and so there needs to be action to fix it correctly. What
I've 
suggested is just a workaround, but it should get you through your problem
for now.

Mark Michelson

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
No virus found in this incoming message.
Checked by AVG - www.avg.com 
Version: 8.0.237 / Virus Database: 270.11.10/1996 - Release Date: 03/12/09
10:38:00


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Queue Realtime agents LOGIN for ami

[Jim Dickenson]

What version of * are you running?
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/




From: Sebastian 
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion

Date: Thu, 12 Mar 2009 18:41:33 -0300
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Subject: [asterisk-users] Queue Realtime agents LOGIN for ami

Is there any AMI action that logs a realtime agent?
I mean, if you send it, queue_log and queue_member get the corresponding
inserts.
 
 
Regards
 
Sebastian


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4)

[Mark Michelson]

David Ruggles wrote:
> I don't really think that's a problem, because I'm able to use the other
> built in options: *1 to record; ## transfer (I changed this from a single
> pound) and there have been a couple times that I wouldn't hit them quickly
> enough.
> 
> Thanks,

Ah, sorry about that. The featuredigittimeout burns so many people that it's 
pretty much a knee-jerk reaction on my part now to suggest that as a potential 
fix.

To test out, I set up the same feature and gave it a try with a current 
subversion checkout of Asterisk 1.4. I placed a call from SIP/2001 to SIP/2000 
and here's what I found.

When SIP/2001 pressed #9, tt-monkeys played on SIP/2000's channel
When SIP/2000 pressed #9, nothing happened.

I tried modifying the features.conf line to have "peer/callee" instead of just 
"peer" and that caused neither side to successfully use the dynamic feature. 
There appears to be a bug which does not allow for the callee to use dynamic 
features. The problem appears to be that when DTMF is pressed, we try to 
interpret the presses to determine if there is a corresponding feature. The 
DYNAMIC_FEATURES variable has been set on the caller's channel, but has not 
been 
set on the callee's channel. As a result, we don't properly read the value of 
the DYNAMIC_FEATURES variable if the callee is the one to press DTMF.

You can work around the bug, although it's not exactly optimal. What you can do 
is to modify your dialplan as follows:

exten => 301,n,Set(DYNAMIC_FEATURES=monkey)
exten => 301,n,Dial(SIP/DavidR1,,M(dynamic_features))

[macro-dynamic_features]
exten => s,1,Set(DYNAMIC_FEATURES=monkey)

By doing this, the dynamic_features macro will be called on SIP/DavidR1 when he 
answers. This will allow for the DYNAMIC_FEATURES variable to be set on both 
channels so both sides can use the feature you have set.

This is a bug, and so there needs to be action to fix it correctly. What I've 
suggested is just a workaround, but it should get you through your problem for 
now.

Mark Michelson

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Serving 120 concurrent calls

[Klaus Darilion]

There was already lots of discussion, e.g. google for
   asterisk monitor nfs
or
   asterisk monitor ramdisk

regards
klaus

Tarek Sawah wrote:
> Hello,
> a local prison contacted us regarding some calling card solution.
> they need 4 E1s to serve 120 rooms in that prison.
> we are planning on using 4 servers to serve the calls and one for the 
> database
> servers' specifications are:
> 2.8 Dual Core Proccessors
> 2 GB Ram
> 160 Sata Drive
> each server will be provided with 1 E1 card
> Questions are:
> 1- will those servers be able to handle that ammount of calls?'
> 2- the important issue is that they require call recording on all 
> calls.. which means we will have to record ALL calls going out of the 
> system .. which means we will need a call recroding.. will the four 
> Asterisk servers handle the recording process or we will need external 
> assistant? and if it was the second choice what is the best suggestion? 
> is there a way to force an Asterisk server to record remote channels?
>  
>  
> -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP 
> Syria: +963 944 618286 USA: +1 347 562 2308
> 
> 
> Express your personality in color! Preview and select themes for 
> Hotmail®. See how. 
> 
> 
> 
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Timeout for Queue

[Mark Michelson]

Darrin Henshaw wrote:
> Hello,
> 
>  
> 
> We had an incident recently where a call was in queue for an extended 
> period of time. We use queuemetrics for reporting, and it reports that 
> the call was waiting for 20 minutes. The different thing about it is 
> that the disconnect reason is stated as Timeout. Is there a set maximum 
> time a call will wait in the queue before being automatically 
> disconnected? I tried looking through the code directly, but I humbly 
> admit my programming skills are lax.
> 
>  
> 
> I’m running Asterisk 1.2.31 on CentOS 4.7. Thanks.
> 

A brief disclaimer: I admittedly don't have much experience with Asterisk 1.2, 
but I don't *think* this behavior is different between 1.2 and any later 
versions. What I'm stating below holds true for Asterisk 1.4.X and Asterisk 
1.6.X

The Queue application has no default timeout. A caller can stay in the queue 
forever if no timeout was given when calling the Queue application.

Mark Michelson

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Serving 120 concurrent calls

[Tarek Sawah]

Hello, 
a local prison contacted us regarding some calling card solution. 
they need 4 E1s to serve 120 rooms in that prison.
we are planning on using 4 servers to serve the calls and one for the database
servers' specifications are:
2.8 Dual Core Proccessors
2 GB Ram
160 Sata Drive
each server will be provided with 1 E1 card
Questions are:
1- will those servers be able to handle that ammount of calls?'
2- the important issue is that they require call recording on all calls.. which 
means we will have to record ALL calls going out of the system .. which means 
we will need a call recroding.. will the four Asterisk servers handle the 
recording process or we will need external assistant? and if it was the second 
choice what is the best suggestion? is there a way to force an Asterisk server 
to record remote channels?
 
 
-- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: 
+963 944 618286 USA: +1 347 562 2308 

_
Express your personality in color! Preview and select themes for Hotmail®. 
http://www.windowslive-hotmail.com/LearnMore/personalize.aspx?ocid=TXT_MSGTX_WL_HM_express_032009#colortheme___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Queue Realtime agents LOGIN for ami

[Sebastian]

Is there any AMI action that logs a realtime agent?

I mean, if you send it, queue_log and queue_member get the corresponding
inserts.

 

 

Regards

 

Sebastian

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4)

[David Ruggles]

I don't really think that's a problem, because I'm able to use the other
built in options: *1 to record; ## transfer (I changed this from a single
pound) and there have been a couple times that I wouldn't hit them quickly
enough.

Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200  da...@safedatausa.com



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Michelson
Sent: Thursday, March 12, 2009 2:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Trying to get sample applicationmap to work
(*1.4)


David Ruggles wrote:
> I'm trying to actually use the example application map in features.conf:
> 
> testfeature => #9,peer,Playback,tt-monkeys  ;Allow both the caller and
> callee to play
> ;tt-monkeys to the opposite
> channel
> 
> I see the feature get registered at the CLI:
> 
>   == Registered Feature 'monkey'
>   == Mapping Feature 'monkey' to app 'Playback(tt-monkeys)' with code '#9'
> 
> But I'm unable to actually use it. This *doesn't* work:
> exten => 301,n,Set(DYNAMIC_FEATURES=monkey)
> exten => 301,n,Dial(SIP/DavidR1)
> 
> 
> Anyone done this before and/or able to give me any suggestions?
> Thanks,
> 

I strongly suspect that you have fallen prey to the featuredigittimeout.
Check 
your features.conf file for the featuredigittimeout option. By default, this
is 
set to 500 ms. You probably want to increase this to something like 2000 ms.

This option specifies the amount of time Asterisk should wait between DTMF 
presses when you are dialing a feature code. So in your case, I'm guessing
that 
you pressed # but could not press 9 in time for Asterisk to recognize this
input 
as part of the same feature.

Mark Michelson

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
No virus found in this incoming message.
Checked by AVG - www.avg.com 
Version: 8.0.237 / Virus Database: 270.11.10/1996 - Release Date: 03/11/09
20:42:00


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SetVar (CDR var) from cli

[Sebastian]

No, I'm not using ForokCDR, I just have a bridged call and try to set the
CDR(userfield) and others on the channels, I look at the channes and I can
see the new value. But when I hangup the cdr record does not contain these
value, if I set it from dialplan works ok, but from cli or from ami is not
working.

Any idea??

Regards

Sebastian


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: jueves, 12 de marzo de 2009 05:57 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SetVar (CDR var) from cli

On Thursday 12 March 2009 15:34:20 Sebastian wrote:
> I'm using 1.6.0.5 I'm trying to set CDR(userfield) for example from the
> cli, I look at the channel variables and I can see the new status, but que
> it hang-ups the CDR doesn't have this value.

Are you using ForkCDR()?  Only a single record would have the changed
userfield unless you specified the r (recursive) option.

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
Se certificó que el correo entrante no contiene virus.
Comprobada por AVG - www.avg.es 
Versión: 8.5.278 / Base de datos de virus: 270.11.10/1996 - Fecha de la
versión: 03/11/09 20:42:00


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Timeout for Queue

[Darrin Henshaw]

Hello,

We had an incident recently where a call was in queue for an extended period of 
time. We use queuemetrics for reporting, and it reports that the call was 
waiting for 20 minutes. The different thing about it is that the disconnect 
reason is stated as Timeout. Is there a set maximum time a call will wait in 
the queue before being automatically disconnected? I tried looking through the 
code directly, but I humbly admit my programming skills are lax.

I'm running Asterisk 1.2.31 on CentOS 4.7. Thanks.


Cheers,

[cid:image001.jpg@01C9A33E.72349BC0]
Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC
Ignition Support Center | www.ignition.bm
Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288
Atlanta | Bermuda | Cayman | Halifax



This email and its attachments may be confidential and are intended solely for 
the use of the individual or parties' to whom it is addressed. All comments are 
solely those of the author and do not necessarily represent those of Ignition. 
If you are not the intended recipient of this email and its attachments, you 
must take no action based upon them, nor must you copy or show them to anyone. 
Please contact the sender if you believe you have received this email in error. 
Thanks for considering the environmental impact before printing this email.
<>___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SetVar (CDR var) from cli

[Tilghman Lesher]

On Thursday 12 March 2009 15:34:20 Sebastian wrote:
> I'm using 1.6.0.5 I'm trying to set CDR(userfield) for example from the
> cli, I look at the channel variables and I can see the new status, but que
> it hang-ups the CDR doesn't have this value.

Are you using ForkCDR()?  Only a single record would have the changed
userfield unless you specified the r (recursive) option.

-- 
Tilghman

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call

[Jim Dickenson]

In version 1.6.0.x if you enable dialplan events in AMI you then get a
packet for each dialplan step executed. You should be able to capture that
data and generate records in a database.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



> From: nik600 
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Thu, 12 Mar 2009 20:21:40 +0100
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] log to cdr each dialpan action, not only one
> record for each call
> 
> On Thu, Mar 12, 2009 at 8:13 PM, Matt Riddell  wrote:
>> On 13/03/2009 8:02 a.m., nik600 wrote:
>>> Hi to all.
>>> 
>>> What can i do if a customer needs to log in the CDR all the dialpan
>>> actions related to a call?
>>> I mean, not only the lastapp e the lastdata but all the dialpan actions!
>>> 
>>> I know that the actual CDR system store one record for each call (and
>>> for billing purposes this can be correct) but in some cases the
>>> approach needed is something similar to the queue_log.
>>> 
>>> I know that exists ResetCDR and ForkCDR but they don't do what i need,
>>> expecially because they fill-in lastdata and lastapp with "ResetCDR"
>>> 
>>> So, what can i do?
>> 
>> Use the Asterisk Manager with UserEvent?
>> 
>> --
>> Kind Regards,
>> 
> UserEvent can be useful, but i have to place it into the dialplan in
> many points.
> 
> With a large dialplan it's a problem.
> 
> -- 
> /*/
> nik600
> http://www.kumbe.it
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call

[nik600]

On Thu, Mar 12, 2009 at 9:22 PM, BJ Weschke  wrote:
>
>  We generated a patch for a client probably about a year ago against the
> 1.4 branch that logged apps for each call, params, and exit status codes
> into a separate file. Like others have said, it generates a tremendous
> amount of data and probably does impact performance on very high load
> servers, but it was very useful to determine EXACTLY what happened with
> a given call.
>
>

You know, sometimes the information is more important that the space
required to store it.

It depends due to the client needs.

If it's possible, can you tell me where you have to place the code to
log when an app is called?

Thanks

-- 
/*/
nik600
http://www.kumbe.it

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SetVar (CDR var) from cli

[Sebastian]

 

Hi,

 

I'm using 1.6.0.5 I'm trying to set CDR(userfield) for example from the cli,
I look at the channel variables and I can see the new status, but que it
hang-ups the CDR doesn't have this value.

I'm using mysql backend for cdr

 

Any idea?

 

Thnks

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] configuring channels for dahdi

[Aqua Man]

Okay, then why do I still have error "unable to create channel of type "DAHDI" 
(cause 66 - Channel not implemented)  after typing dahdi restart;  after typing 
dahdi show staus and I see the card?



> Date: Thu, 12 Mar 2009 19:59:47 +
> From: j...@jeff.net
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] configuring channels for dahdi
> 
> 
> On Thu, 12 Mar 2009, Aqua Man wrote:
> 
> >
> > After typing ps aux | grep asterisk I noticed there are five instances of 
> > asterisk running
> >
> > asterisk   4493  0.0  0.6  23720  3540  ?  S15:12  0:00 
> > /usr/sbin/apache2 -k start
> > asterisk   4494  0.0  0.6  23720  3540  ?  S15:12  0:00 
> > /usr/sbin/apache2 -k start
> > asterisk   4495  0.0  0.6  23720  3540  ?  S15:12  0:00 
> > /usr/sbin/apache2 -k start
> > asterisk   4496  0.0  0.6  23720  3540  ?  S15:12  0:00 
> > /usr/sbin/apache2 -k start
> > asterisk   4497  0.0  0.6  23720  3540  ?  S15:12  0:00 
> > /usr/sbin/apache2 -k start
> > root 4617  0.0  0.1   3004764  pts/0 R+  15:33  0.00 
> > grep asterisk
> 
> Actually there are none.  You do have five apache's, though, which is 
> normal.
> 
> j
> 
> >
> > After running cat/proc/dahdi/* :  It is all in caps
> >
> >
> >
> >> Date: Tue, 10 Mar 2009 23:25:18 +0200
> >> From: tzafrir.co...@xorcom.com
> >> To: asterisk-users@lists.digium.com
> >> Subject: Re: [asterisk-users] configuring channels for dahdi
> >>
> >> Useful VIM command:
> >>
> >> :%s/#\[.;3.;40m//g
> >>
> >> On Tue, Mar 10, 2009 at 10:03:03AM -1000, Aqua Man wrote:
> >>>
> >>> Typed the dahdi restart command and the output is listed in the
> >>> email as per your request.
> >>
> >>>
> >>> *CLI> dahdi restart
> >>> Destroying channels and reloading DAHDI configuration.
> >>> Initial softhangup of all DAHDI channels complete.
> >>> Final softhangup of all DAHDI channels complete.
> >>>   == Unregistered channel -2
> >>>   == Unregistered channel 1
> >>>   == Unregistered channel 4
> >>>   == Parsing '/etc/asterisk/chan_dahdi.conf': Found
> >>> [Mar 10 15:39:56] WARNING[4633]: chan_dahdi.c:930 dahdi_open: Unable to 
> >>> specify channel 1: Device or resource busy
> >>
> >> Huh?
> >>
> >> Any chance there's a different instance of Asterisk running?
> >>
> >> ps aux | grep asterisk
> >>
> >> But see also further below,
> >>
> >>> chan_dahdi.c:930 dahdi_open: Unable to specify channel 1: Device or 
> >>> resource busy
> >>> -- Reconfigured channel 1, FXO Kewlstart signalling
> >>> -- Reconfigured channel 4, FXS Kewlstart signalling
> >>>   == Parsing '/etc/asterisk/users.conf': Found
> >>> *CLI> -- Starting simple switch on 'Zap/1-1'
> >>> -- Executing [18005551...@default:1] Dial("Zap/1-1", 
> >>> "DAHDI/4/w13475823549") in new stack
> >>> [Mar 10 15:40:15] WARNING[4679]: channel.c:3168 ast_request: No channel 
> >>> type registered for 'DAHDI'
> >>> [Mar 10 15:40:15] WARNING[4679]: app_dial.c:1286 dial_exec_full: Unable 
> >>> to create channel of type 'DAHDI' (cause 66 - Channel not implemented)
> >>>   == Everyone is busy/congested at this time (1:0/0/1)
> >>> -- Executing [18005551...@default:2] Hangup("Zap/1-1", "") in new 
> >>> stack
> >>>   == Spawn extension (default, 13475823549, 2) exited non-zero on 
> >>> 'Zap/1-1'
> >>> -- Hungup 'Zap/1-1'
> >>> -- Starting simple switch on 'Zap/1-1'
> >>> -- Hungup 'Zap/1-1'
> >>>
> >>>
> >>> sudo cat /proc/dahdi/*
> >>> Span 1: wctdm/4 'wildcard tdm400p rev I board 5' (master)
> >>>
> >>> 1 wctdm/4/0 fxoks (ec: mg2)
> >>> 2 wctdm/4/1
> >>> 3 wctdm/4/2
> >>> 4 wctdm/4/3/ fxsks (ec:mg2)
> >>
> >> Are all of those lower case?
> >>
> >> No '(In Use)'
> >>
> >>>
> >>>
> >>> [channels]
> >>> usecallerid=yes
> >>> callerid=asreceived
> >>> cidsignalling=bell
> >>> hidecallerid=no
> >>> callwaiting=yes
> >>> musiconhold=default
> >>> usecallingpres=yes
> >>> callwaitingcallerid=yes
> >>> threewaycalling=yes
> >>> transfer=yes
> >>> canpark=yes
> >>> cancallforward=yes
> >>> callreturn=yes
> >>> echocancel=yes
> >>> echocancelwhenbridged=no
> >>> echotraining=400
> >>> rxgain=0.0
> >>> txgain=0.0
> >>> signalling=fxo_ks
> >>> language=en
> >>> context=default
> >>> channel => 1
> >>
> >> An FXO channel should have FXS signalling ...
> >>
> >>> signalling=fxs_ks
> >>> channel => 4
> >>
> >> ... and vice versa
> >>
> >>>
> >>> I used same settings from zapata.conf.  This worked fine before the 
> >>> dahdi.  Any help would much appreciated
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
> >>>
>  Date: Tue, 10 Mar 2009 18:37:35 +0200
>  From: tzafrir.co...@xorcom.com
>  To: asterisk-users@lists.digium.com
>  Subject: Re: [asterisk-users] configuring channels for dahdi
> 
>  On Tue, Mar 10, 2009 at 04:20:47AM -1000, Aqua Man wrote:
> >
> > after installing asterisk 1.4.23.1 and dahdi-linux-2.1.0.4 and at CLI>  
> > module load chan_dahdi.so receive the following:

Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call

[BJ Weschke]

nik600 wrote:
> Hi to all.
>
> What can i do if a customer needs to log in the CDR all the dialpan
> actions related to a call?
> I mean, not only the lastapp e the lastdata but all the dialpan actions!
>
> I know that the actual CDR system store one record for each call (and
> for billing purposes this can be correct) but in some cases the
> approach needed is something similar to the queue_log.
>
> I know that exists ResetCDR and ForkCDR but they don't do what i need,
> expecially because they fill-in lastdata and lastapp with "ResetCDR"
>
> So, what can i do?
>
> Is it better to do some customization to generate a CDR event on each
> dialplan step or is better to parse the logfile and extract the
> information needed?
>
> I'm using Asterisk 1.4.23.1
>   

 We generated a patch for a client probably about a year ago against the 
1.4 branch that logged apps for each call, params, and exit status codes 
into a separate file. Like others have said, it generates a tremendous 
amount of data and probably does impact performance on very high load 
servers, but it was very useful to determine EXACTLY what happened with 
a given call.


> --
> Bird's The Word Technologies, Inc.
> http://www.btwtech.com/
>
>
>   

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] configuring channels for dahdi

[Jeff LaCoursiere]

On Thu, 12 Mar 2009, Aqua Man wrote:

>
> After typing ps aux | grep asterisk I noticed there are five instances of 
> asterisk running
>
> asterisk   4493  0.0  0.6  23720  3540  ?  S15:12  0:00 
> /usr/sbin/apache2 -k start
> asterisk   4494  0.0  0.6  23720  3540  ?  S15:12  0:00 
> /usr/sbin/apache2 -k start
> asterisk   4495  0.0  0.6  23720  3540  ?  S15:12  0:00 
> /usr/sbin/apache2 -k start
> asterisk   4496  0.0  0.6  23720  3540  ?  S15:12  0:00 
> /usr/sbin/apache2 -k start
> asterisk   4497  0.0  0.6  23720  3540  ?  S15:12  0:00 
> /usr/sbin/apache2 -k start
> root 4617  0.0  0.1   3004764  pts/0 R+  15:33  0.00 grep 
> asterisk

Actually there are none.  You do have five apache's, though, which is 
normal.

j

>
> After running cat/proc/dahdi/* :  It is all in caps
>
>
>
>> Date: Tue, 10 Mar 2009 23:25:18 +0200
>> From: tzafrir.co...@xorcom.com
>> To: asterisk-users@lists.digium.com
>> Subject: Re: [asterisk-users] configuring channels for dahdi
>>
>> Useful VIM command:
>>
>> :%s/#\[.;3.;40m//g
>>
>> On Tue, Mar 10, 2009 at 10:03:03AM -1000, Aqua Man wrote:
>>>
>>> Typed the dahdi restart command and the output is listed in the
>>> email as per your request.
>>
>>>
>>> *CLI> dahdi restart
>>> Destroying channels and reloading DAHDI configuration.
>>> Initial softhangup of all DAHDI channels complete.
>>> Final softhangup of all DAHDI channels complete.
>>>   == Unregistered channel -2
>>>   == Unregistered channel 1
>>>   == Unregistered channel 4
>>>   == Parsing '/etc/asterisk/chan_dahdi.conf': Found
>>> [Mar 10 15:39:56] WARNING[4633]: chan_dahdi.c:930 dahdi_open: Unable to 
>>> specify channel 1: Device or resource busy
>>
>> Huh?
>>
>> Any chance there's a different instance of Asterisk running?
>>
>> ps aux | grep asterisk
>>
>> But see also further below,
>>
>>> chan_dahdi.c:930 dahdi_open: Unable to specify channel 1: Device or 
>>> resource busy
>>> -- Reconfigured channel 1, FXO Kewlstart signalling
>>> -- Reconfigured channel 4, FXS Kewlstart signalling
>>>   == Parsing '/etc/asterisk/users.conf': Found
>>> *CLI> -- Starting simple switch on 'Zap/1-1'
>>> -- Executing [18005551...@default:1] Dial("Zap/1-1", 
>>> "DAHDI/4/w13475823549") in new stack
>>> [Mar 10 15:40:15] WARNING[4679]: channel.c:3168 ast_request: No channel 
>>> type registered for 'DAHDI'
>>> [Mar 10 15:40:15] WARNING[4679]: app_dial.c:1286 dial_exec_full: Unable to 
>>> create channel of type 'DAHDI' (cause 66 - Channel not implemented)
>>>   == Everyone is busy/congested at this time (1:0/0/1)
>>> -- Executing [18005551...@default:2] Hangup("Zap/1-1", "") in new stack
>>>   == Spawn extension (default, 13475823549, 2) exited non-zero on 'Zap/1-1'
>>> -- Hungup 'Zap/1-1'
>>> -- Starting simple switch on 'Zap/1-1'
>>> -- Hungup 'Zap/1-1'
>>>
>>>
>>> sudo cat /proc/dahdi/*
>>> Span 1: wctdm/4 'wildcard tdm400p rev I board 5' (master)
>>>
>>> 1 wctdm/4/0 fxoks (ec: mg2)
>>> 2 wctdm/4/1
>>> 3 wctdm/4/2
>>> 4 wctdm/4/3/ fxsks (ec:mg2)
>>
>> Are all of those lower case?
>>
>> No '(In Use)'
>>
>>>
>>>
>>> [channels]
>>> usecallerid=yes
>>> callerid=asreceived
>>> cidsignalling=bell
>>> hidecallerid=no
>>> callwaiting=yes
>>> musiconhold=default
>>> usecallingpres=yes
>>> callwaitingcallerid=yes
>>> threewaycalling=yes
>>> transfer=yes
>>> canpark=yes
>>> cancallforward=yes
>>> callreturn=yes
>>> echocancel=yes
>>> echocancelwhenbridged=no
>>> echotraining=400
>>> rxgain=0.0
>>> txgain=0.0
>>> signalling=fxo_ks
>>> language=en
>>> context=default
>>> channel => 1
>>
>> An FXO channel should have FXS signalling ...
>>
>>> signalling=fxs_ks
>>> channel => 4
>>
>> ... and vice versa
>>
>>>
>>> I used same settings from zapata.conf.  This worked fine before the dahdi.  
>>> Any help would much appreciated
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
 Date: Tue, 10 Mar 2009 18:37:35 +0200
 From: tzafrir.co...@xorcom.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] configuring channels for dahdi

 On Tue, Mar 10, 2009 at 04:20:47AM -1000, Aqua Man wrote:
>
> after installing asterisk 1.4.23.1 and dahdi-linux-2.1.0.4 and at CLI>  
> module load chan_dahdi.so receive the following:
>
>
>
>
>
> signalling must be specified before any channels are.
>
>
>
>
>
> CLI> Warning [4663]: chan_dahdi.c:11627 process_dahdi: Ignoring signalling
>
>
>  Error[4663]: chan_dahdi.c:10946 build_channels: Unable to 
> reconfigure channel '1'

 You got the real error emssage on module load. The error message you get
 now is:

 1. chan_dahdi tells you that it does not change major configuration on
 reload

 2. And that its configuration is still broken.

 Try e.g. 'dahdi restart' from the Asterisk CLI to get the real error.

 For more help in troubleshootin

Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call

[nik600]

On Thu, Mar 12, 2009 at 8:44 PM, Steve Murphy  wrote:


> My current thinking
> is to specify exactly which app invocations you want to track; those
> involved
> with dialing would be automatically tracked. Or time groups of invocations
> via
> forcing a leg-split via a simple dialplan application call...

well, this will be surely the best.

I'll read the documentation and let you know, thanks.

-- 
/*/
nik600
http://www.kumbe.it

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call

[Steve Edwards]

On Thu, 12 Mar 2009, nik600 wrote:

> On Thu, Mar 12, 2009 at 8:13 PM, Matt Riddell  wrote:
>> On 13/03/2009 8:02 a.m., nik600 wrote:
>>> Hi to all.
>>>
>>> What can i do if a customer needs to log in the CDR all the dialpan 
>>> actions related to a call? I mean, not only the lastapp e the lastdata 
>>> but all the dialpan actions!
>>>
>>> I know that the actual CDR system store one record for each call (and 
>>> for billing purposes this can be correct) but in some cases the 
>>> approach needed is something similar to the queue_log.
>>>
>>> I know that exists ResetCDR and ForkCDR but they don't do what i need, 
>>> expecially because they fill-in lastdata and lastapp with "ResetCDR"
>>>
>>> So, what can i do?
>>
>> Use the Asterisk Manager with UserEvent?
>>
> UserEvent can be useful, but i have to place it into the dialplan in
> many points.
>
> With a large dialplan it's a problem.

I do this with "resetcdr(w)" at the top of each context and then log each 
of the CDRs to a modified copy of cdr_addon_mysql.c.

It was a bit of a hassle setting up, but it works great.

For this client, each context represents different products which are 
billed at different rates. At the end of the call, an AGI sums the 
duration for each "product," calculates the total and bills their credit 
card.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Outgoing call drops

[D Tucny]

2009/3/12 Danny Nicholas 

>  Greetings Listers,
>
>   I’m running 1.4.21.2 on SUSE 11.0 with and zaptel
> 1.4.12.1 on a TDM400P.  Most of my calls work great, but occasionally we try
> to connect to a customer or vendor external conference call and the call
> will drop after 60-65 seconds unless I have an Answer before the Dial in the
> dialplan.  Isn’t this solution a hack and what would be a better one?
> Thanks in Advance.
>

Unfortunately, when dealing with analogue lines, hacks are quite often
needed... I'd suggest that in this case something is causing asterisk (or
perhaps the card if polarity reversal is used) to fail to detect the fact
that ringing has ended or an answer signal has been passed when calling
these numbers... As such a timeout occurs...

You'd need to provide more information to pinpoint the exact issue... but...
the only real solution to getting reliable in call event signalling is to
use digital lines (or IP services that use digital lines or all IP) though
some analogue services do a better job using polarity reversals which makes
it much more reliable, the fact that these services are few and far between
makes that not an option for most...

d
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] configuring channels for dahdi

[Aqua Man]

After typing ps aux | grep asterisk I noticed there are five instances of 
asterisk running

asterisk   4493  0.0  0.6  23720  3540  ?  S15:12  0:00 
/usr/sbin/apache2 -k start
asterisk   4494  0.0  0.6  23720  3540  ?  S15:12  0:00 
/usr/sbin/apache2 -k start
asterisk   4495  0.0  0.6  23720  3540  ?  S15:12  0:00 
/usr/sbin/apache2 -k start
asterisk   4496  0.0  0.6  23720  3540  ?  S15:12  0:00 
/usr/sbin/apache2 -k start
asterisk   4497  0.0  0.6  23720  3540  ?  S15:12  0:00 
/usr/sbin/apache2 -k start
root 4617  0.0  0.1   3004764  pts/0 R+  15:33  0.00 grep 
asterisk

After running cat/proc/dahdi/* :  It is all in caps



> Date: Tue, 10 Mar 2009 23:25:18 +0200
> From: tzafrir.co...@xorcom.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] configuring channels for dahdi
> 
> Useful VIM command:
> 
> :%s/#\[.;3.;40m//g
> 
> On Tue, Mar 10, 2009 at 10:03:03AM -1000, Aqua Man wrote:
> > 
> > Typed the dahdi restart command and the output is listed in the 
> > email as per your request.
> 
> > 
> > *CLI> dahdi restart
> > Destroying channels and reloading DAHDI configuration.
> > Initial softhangup of all DAHDI channels complete.
> > Final softhangup of all DAHDI channels complete.
> >   == Unregistered channel -2
> >   == Unregistered channel 1
> >   == Unregistered channel 4
> >   == Parsing '/etc/asterisk/chan_dahdi.conf': Found
> > [Mar 10 15:39:56] WARNING[4633]: chan_dahdi.c:930 dahdi_open: Unable to 
> > specify channel 1: Device or resource busy
> 
> Huh?
> 
> Any chance there's a different instance of Asterisk running?
> 
> ps aux | grep asterisk
> 
> But see also further below,
> 
> > chan_dahdi.c:930 dahdi_open: Unable to specify channel 1: Device or 
> > resource busy
> > -- Reconfigured channel 1, FXO Kewlstart signalling
> > -- Reconfigured channel 4, FXS Kewlstart signalling
> >   == Parsing '/etc/asterisk/users.conf': Found
> > *CLI> -- Starting simple switch on 'Zap/1-1'
> > -- Executing [18005551...@default:1] Dial("Zap/1-1", 
> > "DAHDI/4/w13475823549") in new stack
> > [Mar 10 15:40:15] WARNING[4679]: channel.c:3168 ast_request: No channel 
> > type registered for 'DAHDI'
> > [Mar 10 15:40:15] WARNING[4679]: app_dial.c:1286 dial_exec_full: Unable to 
> > create channel of type 'DAHDI' (cause 66 - Channel not implemented)
> >   == Everyone is busy/congested at this time (1:0/0/1)
> > -- Executing [18005551...@default:2] Hangup("Zap/1-1", "") in new stack
> >   == Spawn extension (default, 13475823549, 2) exited non-zero on 'Zap/1-1'
> > -- Hungup 'Zap/1-1'
> > -- Starting simple switch on 'Zap/1-1'
> > -- Hungup 'Zap/1-1'
> > 
> > 
> > sudo cat /proc/dahdi/* 
> > Span 1: wctdm/4 'wildcard tdm400p rev I board 5' (master) 
> > 
> > 1 wctdm/4/0 fxoks (ec: mg2) 
> > 2 wctdm/4/1 
> > 3 wctdm/4/2 
> > 4 wctdm/4/3/ fxsks (ec:mg2) 
> 
> Are all of those lower case?
> 
> No '(In Use)'
> 
> > 
> > 
> > [channels]
> > usecallerid=yes
> > callerid=asreceived
> > cidsignalling=bell
> > hidecallerid=no
> > callwaiting=yes
> > musiconhold=default
> > usecallingpres=yes
> > callwaitingcallerid=yes
> > threewaycalling=yes
> > transfer=yes
> > canpark=yes
> > cancallforward=yes
> > callreturn=yes
> > echocancel=yes
> > echocancelwhenbridged=no
> > echotraining=400
> > rxgain=0.0
> > txgain=0.0
> > signalling=fxo_ks
> > language=en
> > context=default
> > channel => 1
> 
> An FXO channel should have FXS signalling ...
> 
> > signalling=fxs_ks
> > channel => 4
> 
> ... and vice versa
> 
> > 
> > I used same settings from zapata.conf.  This worked fine before the dahdi.  
> > Any help would much appreciated
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > > Date: Tue, 10 Mar 2009 18:37:35 +0200
> > > From: tzafrir.co...@xorcom.com
> > > To: asterisk-users@lists.digium.com
> > > Subject: Re: [asterisk-users] configuring channels for dahdi
> > > 
> > > On Tue, Mar 10, 2009 at 04:20:47AM -1000, Aqua Man wrote:
> > > > 
> > > > after installing asterisk 1.4.23.1 and dahdi-linux-2.1.0.4 and at CLI>  
> > > > module load chan_dahdi.so receive the following:
> > > > 
> > > > 
> > > > 
> > > > 
> > > > 
> > > > signalling must be specified before any channels are.
> > > > 
> > > > 
> > > > 
> > > > 
> > > > 
> > > > CLI> Warning [4663]: chan_dahdi.c:11627 process_dahdi: Ignoring 
> > > > signalling
> > > > 
> > > > 
> > > >  Error[4663]: chan_dahdi.c:10946 build_channels: Unable to 
> > > > reconfigure channel '1'
> > > 
> > > You got the real error emssage on module load. The error message you get
> > > now is:
> > > 
> > > 1. chan_dahdi tells you that it does not change major configuration on
> > > reload
> > > 
> > > 2. And that its configuration is still broken.
> > > 
> > > Try e.g. 'dahdi restart' from the Asterisk CLI to get the real error.
> > > 
> > > For more help in troubleshooting, also provide your zapata.conf /
> > > chan_dahdi.conf as well as

Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call

[Steve Murphy]

On Thu, Mar 12, 2009 at 1:13 PM, Matt Riddell  wrote:

> On 13/03/2009 8:02 a.m., nik600 wrote:
> > Hi to all.
> >
> > What can i do if a customer needs to log in the CDR all the dialpan
> > actions related to a call?
> > I mean, not only the lastapp e the lastdata but all the dialpan actions!
> >
> > I know that the actual CDR system store one record for each call (and
> > for billing purposes this can be correct) but in some cases the
> > approach needed is something similar to the queue_log.
> >
> > I know that exists ResetCDR and ForkCDR but they don't do what i need,
> > expecially because they fill-in lastdata and lastapp with "ResetCDR"
> >
> > So, what can i do?
>
> Use the Asterisk Manager with UserEvent?


I think Matt's approach is more practical; Really, you can't just use the
CDR logs, because that's what isn't enough for you. Some folks have reported
getting by with a mix of CDR's and manager events, but ooo, it's ugly!
It can be done, tho.

But  you could read my proposed CDR doc by checking it out of SVN:

svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs

and reading the pdf in the dir that will be created, and letting me know
what you think of it. The Leg-based approach with your own splits might
be the ticket you are interested in.

Now, I might add this: surely you are jesting about *every* dialplan
command!
Most dialplan apps that get used are things like "gotoIf" and such, that
take
some number of microseconds to execute. It'd be silly timing such, as the
CDR records are currently rounded off to whole seconds. My current thinking
is to specify exactly which app invocations you want to track; those
involved
with dialing would be automatically tracked. Or time groups of invocations
via
forcing a leg-split via a simple dialplan application call...

again, read the doc, and let me know what you think.

murf



-- 
Steve Murphy
ParseTree Corp
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Is it possible to get full callin number from E1?

[D Tucny]

2009/3/12 ssmax 

> Hi all
>
>i have just set up a asterisk in china, using DE410P and one E1 line
>and get a phone number like: +86 020 87654321 from my sp
>when somebody dial  +86 020 87654321 , the asterisk will get the call in
> number by ${EXTEN} variable, but it can only get 87654321, no area code .


As others have mentioned in this thread, this is pretty normal... Telco's
all have different policies on how many digits they will pass for the
dialled number, some may let you specify, but often each telco has an
internal standard... You don't need the rest of the number, but if you're
doing something where it would be useful to have it, you can easily add the
prefix yourself... It's also worth pointing out at this point, the prefix is
either 020 or +86-20 for Guangzhou depending on whether it's written for
national or international use, the 0 is not dialled internationally...


>
>when someone dial  +86 020 87654321 ,  means 4 digits,  the
> phone can call in, and the  ${EXTEN}  is only  87654321 too ,
>is it possible to get  full call in number 87654321  in asterisk ?
>  thanks
>

This is confusing... Guangzhou numbers are only 8 digits long + prefixes,
so, +86 20 87654321 would be the number... Where do you get '' from?
What would be more normal would be that you'd by DDIs with your ISDN30
service, such that you'd have for example, a range of 100 numbers say from
87654000 to 87654099, though you can buy more... With this, the number
dialled (or the last 8 digits in your case) would be accessible through the
${EXTEN} variable... I guess it could be possible that your telco is
attempting to offer some other type of service that allows them to issue you
a single number, but, that they will accept as many as four additional
digits that they will pass to you in some way, but, it's impossible to say
what they are doing here and how it's been implemented... I would further
suspect that if this is the case that there would be a very distinct chance
that this would only work with inbound calls from other users of the same
telco, and in China, in the same city...

Within China, telco interoperability can be flaky at best and, from
experience, the fact the telco engineers don't entirely understand their
huawei switches for local problems and for problems over a wider area, the
fact that the national telcos are split into provincial and city
organisations that don't appear to communicate without going through some
circuitous route via the telco's head office both lead to some very long fix
times if there ever is a fix... That said, as long as you have cash, you can
get them to provide almost any service you want, though they won't often
offer anything but the most basic service, you have to ask for it...

d
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call

[nik600]

On Thu, Mar 12, 2009 at 8:13 PM, Matt Riddell  wrote:
> On 13/03/2009 8:02 a.m., nik600 wrote:
>> Hi to all.
>>
>> What can i do if a customer needs to log in the CDR all the dialpan
>> actions related to a call?
>> I mean, not only the lastapp e the lastdata but all the dialpan actions!
>>
>> I know that the actual CDR system store one record for each call (and
>> for billing purposes this can be correct) but in some cases the
>> approach needed is something similar to the queue_log.
>>
>> I know that exists ResetCDR and ForkCDR but they don't do what i need,
>> expecially because they fill-in lastdata and lastapp with "ResetCDR"
>>
>> So, what can i do?
>
> Use the Asterisk Manager with UserEvent?
>
> --
> Kind Regards,
>
UserEvent can be useful, but i have to place it into the dialplan in
many points.

With a large dialplan it's a problem.

-- 
/*/
nik600
http://www.kumbe.it

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call

[Matt Riddell]

On 13/03/2009 8:02 a.m., nik600 wrote:
> Hi to all.
>
> What can i do if a customer needs to log in the CDR all the dialpan
> actions related to a call?
> I mean, not only the lastapp e the lastdata but all the dialpan actions!
>
> I know that the actual CDR system store one record for each call (and
> for billing purposes this can be correct) but in some cases the
> approach needed is something similar to the queue_log.
>
> I know that exists ResetCDR and ForkCDR but they don't do what i need,
> expecially because they fill-in lastdata and lastapp with "ResetCDR"
>
> So, what can i do?

Use the Asterisk Manager with UserEvent?

-- 
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of nik600
Sent: Thursday, March 12, 2009 2:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] log to cdr each dialpan action,not only one record
for each call

Hi to all.

What can i do if a customer needs to log in the CDR all the dialpan
actions related to a call?
I mean, not only the lastapp e the lastdata but all the dialpan actions!

I know that the actual CDR system store one record for each call (and
for billing purposes this can be correct) but in some cases the
approach needed is something similar to the queue_log.

I know that exists ResetCDR and ForkCDR but they don't do what i need,
expecially because they fill-in lastdata and lastapp with "ResetCDR"

So, what can i do?

Is it better to do some customization to generate a CDR event on each
dialplan step or is better to parse the logfile and extract the
information needed?

I'm using Asterisk 1.4.23.1

TIA

-- 
/*/
nik600
http://www.kumbe.it

I'd probably go for parsing the logfile and extracting what you want.  Are
you using a SQL log or the flat Master.csv as your logfile?

Regards,
Danny Nicholas

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk and WebIntegration

[Kurian Thayil]

Hi Geriant,

My apologies for the delay in reply. We won't be using php but Perl and
there is an AGI module for perl Asterisk::AGI. I may be using Manager API
for sending Hangup signal. Im planning to write a bash script which perl
invokes when hangup button is pressed in the web interface. Bash script
telnets and sends Hangup signal to the manager API. I am not yet able to
acheive sending commands via bash script using telnet. But I am trying.

One thing that's confusing me is if in future, incoming facility needs to be
activated and since Auto Answer feature in EyeBeam/Twinkle is ON, don't you
think that would be a problem? I think for that, the possible work around
will be using 2 softphones, say EyeBeam and Xlite together in the same PC.
Configuring one extension in EyeBeam to make outbound calls (with Auto
Answer enabled) and configuring Xlite with an extension which receives
inbound calls. Do you have any suggestion on that?

Regards,

Kurian Mathew Thayil.


On Tue, Mar 10, 2009 at 7:32 PM, Geraint Lee  wrote:

> If you're using a php i'd take a look at phpagi - there are others around
> for various different languages too. our agents use twinkle with
> auto-answer, the only reason they need to look at twinkle is if they need to
> perform a transfer (that too will soon be done from the web browser), you
> can do pretty much anything with the asterisk manager (originate the call
> and hangup the call and a load of other useful stuff)
>
> Cheers
>
> 2009/3/10 Kurian Thayil 
>
> Hi Steve,
>>
>> That worked beautifully. Thank you so much. But one question though.
>> Imagine if I keep a Hangup Button in the interface and it should terminate
>> the call. Will that be possible? This scenario happens when the user gets
>> connected to an invalid phone number where the user have to manually
>> disconnect. I don't plan to confuse the user by asking them to use eyebeam
>> to disconnect the call. If it could be integrated to the web interface they
>> just have to stick on to that alone. Is there any way?
>>
>> Regards,
>>
>> Kurian Mathew Thayil.
>>
>> On Tue, Mar 10, 2009 at 4:51 PM, Steve Totaro <
>> stot...@first-notification.com> wrote:
>>
>>>
>>>
>>>   On Tue, Mar 10, 2009 at 6:40 AM, Kurian Thayil <
>>> kurianmtha...@gmail.com> wrote:
>>>
 Hi All,

 Is there a way that I can include call dialing functionality in a
 webinterface. I have EyeBeam configured with a SIP user say
 8440. Will I be able to design an inteface which agent can choose a
 number and the Dial without punching in the number in
 Eyebeam.
 I tried using the .call file. ie The agent can choose which number to
 dial from a web interface. Then, a .call file is
 created with the following informations.

 Channel: Zap/g2/9444204943
 Context: inbound_support
 Extension: 8440
 Priority: 0

 Now, in the extensions.conf file, I mentioned the following under
 inbound_support context.

 [inbound_support]
 exten =>8440,1,Dial(SIP/8440,55,tTo)
 exten =>8440,2,Answer
 exten =>8440,3,Hangup

 But, here the call gets connected only when the receiver end receives
 the call. When the receiver end picks up the phone,
 SIP/8440 rings.

 Is there any other way to implement this. I am not ready to use Vicidial
 (AstGUIClient) because the interface to be designed
 is too custom and the agent should have the list of numbers in front of
 them while they dial which cannot be done using
 Vicicial.

 Regards,

 Kurian Mathew Thayil.

>>>
>>> The following will ring the internal support personnel (8440) first,
>>> after answered, it will then dial the customer (14109850123) (Are you in
>>> Maryland?)
>>>
>>> Turn on auto-answer and it should be seamless.
>>>
>>>
>>> Stolen from Wiki:
>>>
>>> To create a call to 14109850123 on a SIP phones called bt101, here's the
>>> file you'd create in /var/spool/asterisk/outgoing (whatever name is good, of
>>> course must be accessible and deletable by asterisk GNU/Linux user):
>>>
>>> Channel: SIP/8440
>>> MaxRetries: 1
>>> RetryTime: 60
>>> WaitTime: 30
>>>
>>> #
>>> # Assuming that your outgoing call logic is kept in the #  context called 
>>> [outgoing]
>>>
>>> # Context: outgoing
>>> # Extension: 14109850123
>>> # Priority: 1
>>>
>>>
>>> --
>>> Thanks,
>>> Steve Totaro
>>> +18887771888 (Toll Free)
>>> +12409381212 (Cell)
>>>
>>>
>>> ___
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> _

[asterisk-users] log to cdr each dialpan action, not only one record for each call

[nik600]

Hi to all.

What can i do if a customer needs to log in the CDR all the dialpan
actions related to a call?
I mean, not only the lastapp e the lastdata but all the dialpan actions!

I know that the actual CDR system store one record for each call (and
for billing purposes this can be correct) but in some cases the
approach needed is something similar to the queue_log.

I know that exists ResetCDR and ForkCDR but they don't do what i need,
expecially because they fill-in lastdata and lastapp with "ResetCDR"

So, what can i do?

Is it better to do some customization to generate a CDR event on each
dialplan step or is better to parse the logfile and extract the
information needed?

I'm using Asterisk 1.4.23.1

TIA

-- 
/*/
nik600
http://www.kumbe.it

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] phone emulator for doing interop testing

[Matt Riddell]

On 13/03/2009 7:01 a.m., Simon P. Ditner wrote:
> I'm looking for some software to do emulation of phones so that I can
> test out a whole range of phones and their firmware revisions against
> asterisk. Anyone know of something like that?
>
> I'm hoping that the hardware vendors have something like that, and
> _maybe_ I'm really lucky and they are using something somewhat standard
> like qemu.

Nope.  Doesn't exist.  Won't ever exist.

What you really want is to go to an Interop meeting at Sipit.

The hardware vendors will never speak with each other.

-- 
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Is it possible to get full callin number from E1?

[Klaus Darilion]

Jimmy Godbout wrote:
> ssmax,
> 
> Use CALLERID(num) to get the number that was dialed.

CALLERID(num) is the calling number, not the called number

klaus



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Is it possible to get full callin number from E1?

[Klaus Darilion]

ssmax wrote:
> Hi all
> 
> i have just set up a asterisk in china, using DE410P and one E1 line 
> and get a phone number like: +86 020 87654321 from my sp when
> somebody dial  +86 020 87654321 , the asterisk will get the call in
> number by ${EXTEN} variable, but it can only get 87654321, no area
> code . when someone dial  +86 020 87654321 ,  means 4 digits,
> the phone can call in, and the  ${EXTEN}  is only  87654321 too , is
> it possible to get  full call in number 87654321  in asterisk ?
> thanks  -- ssmax 2009-03-12

On Asterisk side you can not configure what your telco will send you.

Enable pri debugging ("pri debug span 1" or "pri set debug on span 1" 
debpending on your Asterisk version).

Then call the number and watch the logging on the Asterisk console. you 
will see an incoming SETUP message. Take a look at the "Called Numer" 
Information Element. The called number is signaled as a number within a 
certain context (ToN=type of number). Probably the provider is sending 
you the called number with ToN=local or unknown, as you do not see the 
area code. Ask the provider to send you the called number with different 
ToN e.g. national and you should see the whole number.

regards
klaus


> 
> 
> 
> ___ -- Bandwidth and
> Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list To UNSUBSCRIBE or update options visit: 
> http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Are .call files working with extensions.ael ? bristuff problem

[Olivier]

2009/3/12 Tzafrir Cohen 

> On Thu, Mar 12, 2009 at 10:18:22AM -0500, Mark Michelson wrote:
>
> >
> > Apparently bristuff has added new required parameters to call files.
>
> Rather: a new optional parameter. When you initiate a call from a
> channel it can also have either a 'message' or a 'pdo' message.
>

Maybe this parameter is optional but here, the fact is I can't find a way to
write any single call file accepted by Bristuff.

If someone have a working Bristuff-compliant example, I would be happy to
test it in my Bristuff setup and report to the list if the troubles I met
came from by ignorance (likely) or a bug (unlikely).
I'll also test early next week the example given in the original thread.

At this moment, we know there's Bristuff patch enhancing .call files but
we're still wondering if there's a bug in it.

Regards


>
> Those are then passed along (eventually) to the new channel method
> send_message. Which is a bit like send_text:
>
>  res = chan->tech->send_message(chan, dest, text, ispdu);
>
> This is part of the code to support the Junghanns GSM cards.
>
> See
>
> http://repo.or.cz/w/asterisk-bristuff.git?a=shortlog;h=refs/heads/bristuff-part-gsm
>
> And specifically:
>
>
> http://repo.or.cz/w/asterisk-bristuff.git?a=shortlog;h=refs/heads/bristuff-part-gsm
>
> http://repo.or.cz/w/asterisk-bristuff.git?a=shortlog;h=refs/heads/bristuff-part-gsm
>
> http://repo.or.cz/w/asterisk-bristuff.git?a=commit;h=6fb93cd5fc4506dda3eb0cfc3a5d5156b3ee7583
>
> http://repo.or.cz/w/asterisk-bristuff.git?a=commit;h=fffd308c5c1fdce3fe43b2d1420cb7cfa3de9fde
>
> /me needs to get that tree in shape.
>
> --
>   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
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] UK ISDN-30 and ANI

[Steve Totaro]

On Thu, Mar 12, 2009 at 2:07 PM, Steve Davies  wrote:

> 2009/3/12 Julian Lyndon-Smith :
> > Has anyone in the UK got ANI to work on an inbound call ?
> >
> > Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30
> >
> > Julian
> >
>
> Have you asked the Telco to send the ANI data? AFAIK, this is disabled
> by default on all BT lines. I assume they are able to send it, but
> being BT they will probably want to charge you a monthly fee for it.
>
> It may be that if it is being sent, it is not being picked up by
> Asterisk - A "pri debug" dump might help you see what is being
> received.
>
> Regards,
> Steve
>
>
ANI is generally sent on toll free circuits by default since you are paying
the bill for inbound calls.

BT may do it, I am in the States and never used BT, but just bear in mind
that ANI can be spoofed similar to CallerID.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] UK ISDN-30 and ANI

[Steve Davies]

2009/3/12 Julian Lyndon-Smith :
> Has anyone in the UK got ANI to work on an inbound call ?
>
> Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30
>
> Julian
>

Have you asked the Telco to send the ANI data? AFAIK, this is disabled
by default on all BT lines. I assume they are able to send it, but
being BT they will probably want to charge you a monthly fee for it.

It may be that if it is being sent, it is not being picked up by
Asterisk - A "pri debug" dump might help you see what is being
received.

Regards,
Steve

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4)

[Mark Michelson]

David Ruggles wrote:
> I'm trying to actually use the example application map in features.conf:
> 
> testfeature => #9,peer,Playback,tt-monkeys  ;Allow both the caller and
> callee to play
> ;tt-monkeys to the opposite
> channel
> 
> I see the feature get registered at the CLI:
> 
>   == Registered Feature 'monkey'
>   == Mapping Feature 'monkey' to app 'Playback(tt-monkeys)' with code '#9'
> 
> But I'm unable to actually use it. This *doesn't* work:
> exten => 301,n,Set(DYNAMIC_FEATURES=monkey)
> exten => 301,n,Dial(SIP/DavidR1)
> 
> 
> Anyone done this before and/or able to give me any suggestions?
> Thanks,
> 

I strongly suspect that you have fallen prey to the featuredigittimeout. Check 
your features.conf file for the featuredigittimeout option. By default, this is 
set to 500 ms. You probably want to increase this to something like 2000 ms. 
This option specifies the amount of time Asterisk should wait between DTMF 
presses when you are dialing a feature code. So in your case, I'm guessing that 
you pressed # but could not press 9 in time for Asterisk to recognize this 
input 
as part of the same feature.

Mark Michelson

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] phone emulator for doing interop testing

[Simon P. Ditner]

I'm looking for some software to do emulation of phones so that I can 
test out a whole range of phones and their firmware revisions against 
asterisk. Anyone know of something like that?

I'm hoping that the hardware vendors have something like that, and 
_maybe_ I'm really lucky and they are using something somewhat standard 
like qemu.

Cheers,
spd

--
| It ain't what you don't know that gets you into trouble. It's what
| you know for sure that just ain't so.   -- Mark Twain
|
| Network: http://www.linkedin.com/in/spditner
|  http://facebook.com/people/Simon-P-Ditner/776370031
|  http://twitter.com/spditner

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Are .call files working with extensions.ael ? bristuff problem

[Tzafrir Cohen]

On Thu, Mar 12, 2009 at 10:18:22AM -0500, Mark Michelson wrote:

> 
> Apparently bristuff has added new required parameters to call files. 

Rather: a new optional parameter. When you initiate a call from a
channel it can also have either a 'message' or a 'pdo' message. 

Those are then passed along (eventually) to the new channel method
send_message. Which is a bit like send_text:

  res = chan->tech->send_message(chan, dest, text, ispdu);

This is part of the code to support the Junghanns GSM cards.

See
http://repo.or.cz/w/asterisk-bristuff.git?a=shortlog;h=refs/heads/bristuff-part-gsm

And specifically:

  
http://repo.or.cz/w/asterisk-bristuff.git?a=shortlog;h=refs/heads/bristuff-part-gsm
  
http://repo.or.cz/w/asterisk-bristuff.git?a=shortlog;h=refs/heads/bristuff-part-gsm
  
http://repo.or.cz/w/asterisk-bristuff.git?a=commit;h=6fb93cd5fc4506dda3eb0cfc3a5d5156b3ee7583
  
http://repo.or.cz/w/asterisk-bristuff.git?a=commit;h=fffd308c5c1fdce3fe43b2d1420cb7cfa3de9fde

/me needs to get that tree in shape.

-- 
   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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Sending faxes with T.38 problem. Asterisk - 1.6.0.6

[Santiago Gimeno]

Thanks for the responses.

I have solved the problem by using a different tiff generator. I used the gs
command:

# gs -q -sDEVICE=tiffg3 -dSAFER -dNOPAUSE -sOutputFile=test.tif test.pdf

Best regards,

Santi



On Thu, Mar 12, 2009 at 3:30 PM, David Backeberg wrote:

> On Wed, Mar 11, 2009 at 7:32 AM, Santiago Gimeno
>  wrote:
> > I finally solved the issue by changing the resolution and the width of
> the
> > TIFF file to one that is accepted by the fax standard. In my case I
> changed
> > to a resolution of 96x96 and a width of 1728.
> >
> > Now I am able to send faxes, but something weird is happening, the fax
> > received in the fax-machine has the black and white colours inverted. Any
> > ideas why this could be happening?
>
> The way I got my tiff file for testing was to use ReceiveFax to make a
> tiff from an inbound fax.
>
> I then used that tiff outbound for testing outbound faxing.
>
> Something you might want to consider doing?
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Trying to get sample applicationmap to work (*1.4)

[David Ruggles]

I'm trying to actually use the example application map in features.conf:

testfeature => #9,peer,Playback,tt-monkeys  ;Allow both the caller and
callee to play
;tt-monkeys to the opposite
channel

I see the feature get registered at the CLI:

  == Registered Feature 'monkey'
  == Mapping Feature 'monkey' to app 'Playback(tt-monkeys)' with code '#9'

But I'm unable to actually use it. This *doesn't* work:
exten => 301,n,Set(DYNAMIC_FEATURES=monkey)
exten => 301,n,Dial(SIP/DavidR1)


Anyone done this before and/or able to give me any suggestions?
Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200  da...@safedatausa.com



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] chanspy problems (asterisk 1.6.0.6) - When spying starts, the spied parties can't hear each other

[Deepak]

Hi, I am in a predicament and any help/pointers would be appreciated.

We are using chanspy to listen in on conversations. We are doing this via a
web interface. The web interface lists all the ongoing calls. We click on a
call and then my local phone rings allowing me to spy on the session I
clicked on.

 But "most" of the time, when I start listening in, the two parties that are
in conversation stop hearing each other. I can hear that they cant hear each
other.

We are using chanspy and passing the exact session-name to it as an
argument.

Asterisk 1.6.0.6 is being used a B2BUA. It receives calls from users
connected via analog phones to a Quintum and forwards them to a PSTNGW.

Analogphone--Quintim<-- SIP--->Asterisk<-SIP-->PSTNGW(T1s)

All media is passing thru asterisk (canreinvite = no).


Thanks in advance for any help
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Is it possible to get full callin number fromE1?

[Dan Austin]

Steve wrote:
Speaking from T1 PRI and E1 PRI in West Africa, you tell the telco how many 
digits to send.  Often times, at least in my experience, if not specified, they 
will only send the last four providing there are no conflicts.

They should be able to send however many digits you require, but maybe they 
wont.
I have found that each telco, and in fact each CO, may have a different 
practice for how many
digits they send.  If you have a good sales person when you place the order you 
can specify, or
at least they will provide that information.  If you have an uninformed sales 
person, the engineer
who sets up you circuit can usually provide that.  If you end up with 
uninformed sales and engineering
personnel, then it is time to test and debug.
After dealing with telcos in China, Korea, Taiwan, Japan, India, Israel, 
Germany, The Netherlands and
the USA, I have found *1* telco that provided all of the circuit details in 
advance without asking.  Most
will tell you they are using the standard values, but have no idea what those 
values are.
Here is my check-list I ask for on new orders-
Linecoding  (If E1 I also ask if G.703 
or not)
Framing
Switchtype
Digits sent/outpulsed
Digits expected (City or local code required?)

Some telcos have been willing to change the number of digits sent, but I 
usually just
rewrite the received number to prepend the missing digits.  It is a little more 
work for
me, but it reduces the chance that future maintenance by the telco will revert 
the
circuit back to their normal values and bust inbound calls.
Dan
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] an easy way to deal with/without leading "1" ?

[sean darcy]

I'm setting up dialplans to deal with 800 dialing through a different
channel than regular long distance in the US.

The regular long distance is set up so users can but don't have to
dial one. That's pretty easy, just one more exten statement. But it's
a pain dealing with all the 8xx area codes that are toll free.

I tried

exten=>exten => _!877NXX,2,Dial(${pstnline}/ww1${EXTEN:-10})

but that matches everything. I'd hoped it would only match strings
that had zero or more characters, followed by the 877 pattern.

sean

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Is it possible to get full callin number fromE1?

[Alex Balashov]

All of this really depends on the signaling.

You did not specify whether E1 was ISDN, CAS, etc.

--
Sent from mobile device

On Mar 12, 2009, at 11:18 AM, Steve Totaro  
 wrote:


Speaking from T1 PRI and E1 PRI in West Africa, you tell the telco  
how many digits to send.  Often times, at least in my experience, if  
not specified, they will only send the last four providing there are  
no conflicts.


They should be able to send however many digits you require, but  
maybe they wont.


--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)


On Thu, Mar 12, 2009 at 10:38 AM, Jimmy Godbout  wrote:
Sorry, I indicated the wrong variable.
You can always ask your provider what is sent.

> -Original Message-
> From: ss...@126.com
> Sent: Thu, 12 Mar 2009 22:11:32 +0800
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Is it possible to get full callin  
number

> fromE1?
>
> hi,Jimmy Godbout
>
> when  +86 136  make a call to ->
> +86 020 87654321  -> asterisk
>
> the CALLERID(num)  will show the caller number +86136
> the ${EXTEN} is the dialed number  87654321
> i will try the CALLERID(dnid) tomorrow, will this get the whole  
dialed

> number 87654321 ?
> i don't know whether the sp will send me the whole number
>
> --
> ssmax
> 2009-03-12
>
> -
> 发件人：Jimmy Godbout
> 发送日期：2009-03-12 21:35:06
> 收件人：Asterisk Users Mailing List - Non-Commercial Discussion
> 抄送：
> 主题：Re: [asterisk-users] Is it possible to get full callin  
number fromE1?

>
> ssmax,
>
> Use CALLERID(num) to get the number that was dialed.
>
> Jimmy
>
>> -Original Message-
>> From: ss...@126.com
>> Sent: Thu, 12 Mar 2009 20:40:59 +0800
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] Is it possible to get full callin  
number from

>> E1?
>>
>> Hi all
>>
>> i have just set up a asterisk in china, using DE410P and one  
E1 line

>> and get a phone number like: +86 020 87654321 from my sp
>> when somebody dial  +86 020 87654321 , the asterisk will get  
the

>> call
>> in number by ${EXTEN} variable, but it can only get 87654321, no  
area

>> code .
>> when someone dial  +86 020 87654321 ,  means 4  
digits,  the

>> phone can call in, and the  ${EXTEN}  is only  87654321 too ,
>> is it possible to get  full call in number 87654321  in  
asterisk

>> ?  thanks
>>
>> --
>> ssmax
>> 2009-03-12
>>




___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Is it possible to get full callin number fromE1?

[Steve Totaro]

Speaking from T1 PRI and E1 PRI in West Africa, you tell the telco how many
digits to send.  Often times, at least in my experience, if not specified,
they will only send the last four providing there are no conflicts.

They should be able to send however many digits you require, but maybe they
wont.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)


On Thu, Mar 12, 2009 at 10:38 AM, Jimmy Godbout  wrote:

> Sorry, I indicated the wrong variable.
> You can always ask your provider what is sent.
>
> > -Original Message-
> > From: ss...@126.com
> > Sent: Thu, 12 Mar 2009 22:11:32 +0800
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [asterisk-users] Is it possible to get full callin number
> > fromE1?
> >
> > hi,Jimmy Godbout
> >
> > when  +86 136  make a call to ->
> > +86 020 87654321  -> asterisk
> >
> > the CALLERID(num)  will show the caller number +86136
> > the ${EXTEN} is the dialed number  87654321
> > i will try the CALLERID(dnid) tomorrow, will this get the whole dialed
> > number 87654321 ?
> > i don't know whether the sp will send me the whole number
> >
> > --
> > ssmax
> > 2009-03-12
> >
> > -
> > 发件人：Jimmy Godbout
> > 发送日期：2009-03-12 21:35:06
> > 收件人：Asterisk Users Mailing List - Non-Commercial Discussion
> > 抄送：
> > 主题：Re: [asterisk-users] Is it possible to get full callin number fromE1?
> >
> > ssmax,
> >
> > Use CALLERID(num) to get the number that was dialed.
> >
> > Jimmy
> >
> >> -Original Message-
> >> From: ss...@126.com
> >> Sent: Thu, 12 Mar 2009 20:40:59 +0800
> >> To: asterisk-users@lists.digium.com
> >> Subject: [asterisk-users] Is it possible to get full callin number from
> >> E1?
> >>
> >> Hi all
> >>
> >> i have just set up a asterisk in china, using DE410P and one E1 line
> >> and get a phone number like: +86 020 87654321 from my sp
> >> when somebody dial  +86 020 87654321 , the asterisk will get the
> >> call
> >> in number by ${EXTEN} variable, but it can only get 87654321, no area
> >> code .
> >> when someone dial  +86 020 87654321 ,  means 4 digits,  the
> >> phone can call in, and the  ${EXTEN}  is only  87654321 too ,
> >> is it possible to get  full call in number 87654321  in asterisk
> >> ?  thanks
> >>
> >> --
> >> ssmax
> >> 2009-03-12
> >>
>
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Outgoing call drops

[Danny Nicholas]

Greetings Listers,

  I'm running 1.4.21.2 on SUSE 11.0 with and zaptel
1.4.12.1 on a TDM400P.  Most of my calls work great, but occasionally we try
to connect to a customer or vendor external conference call and the call
will drop after 60-65 seconds unless I have an Answer before the Dial in the
dialplan.  Isn't this solution a hack and what would be a better one?
Thanks in Advance.

 

Danny Nicholas

 

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Are .call files working with extensions.ael ? bristuff problem

[Mark Michelson]

Peer Oliver Schmidt wrote:
> Olivier wrote:
>>> Do you by chance use bristuff?
>> Yes, I do.
> 
> bristuff patches pbx/pbx_spool.c
> 
> I have no knowledge of C, but there seems to be a problem around line 
> 266.
> 
> The original line (pre-bristuff) looks like this:
> 
> if (ast_strlen_zero(o->tech) || ast_strlen_zero(o->dest) || 
> (ast_strlen_zero(o->app) && ast_strlen_zero(o->exten))) {
> 
> 
> The patched line looks like this:
> 
> if (ast_strlen_zero(o->tech) || ast_strlen_zero(o->dest) || 
> (ast_strlen_zero(o->app) && ast_strlen_zero(o->exten)) || 
> (ast_strlen_zero(o->message) && ast_strlen_zero(o->pdu))) {
> 
> Try reverting that line, and see if that helps with your problem. And 
> maybe someone with a better understanding of C can take a look at the 
> above problem.

Apparently bristuff has added new required parameters to call files. Basically, 
it has the same requirements as vanilla Asterisk (you must specify a full 
channel name and either an app or an extension) and it also requires that one 
of 
the "message" or "pdu" fields of the outgoing call are filled in. You may want 
to check bristuff documentation to figure out what these mean since they are 
not 
part of a regular Asterisk installation.

Mark Michelson

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] BRI/ISDN, misdn.conf/misdn-init.conf, OpenVOX B100P and Etisalat in Dubai

[Paulo Santos]

Gavin Henry wrote:
> Hi All,
> 
> We've got msidn configured:
> 
> Port  1: TE-mode BRI S/T interface line (for phone lines)
>  -> Protocol: DSS1 (Euro ISDN)
>  -> childcnt: 2
> 

I don't know if it depends on the card, but in my case I need to set the 
termination jumper on TE mode for lines from PSTN. Mind to check the 
TE/NT jumper as well.


> te_ptmp=1
> 
> (...)
> 
> [isdn]
> ports=1
> context=from-pstn
> msns=*

Here you have set PTMP, usually used with DID, and then you have MSN, 
usually used with PTP (at least here is how my telephone company use it).

Find out if you use PTP or PTMP.


Regards,
Paulo Santos
-- 
HTML e-mail is evil. Go plain text.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Is it possible to get full callin number fromE1?

[Jimmy Godbout]

Sorry, I indicated the wrong variable.
You can always ask your provider what is sent.

> -Original Message-
> From: ss...@126.com
> Sent: Thu, 12 Mar 2009 22:11:32 +0800
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Is it possible to get full callin number
> fromE1?
> 
> hi,Jimmy Godbout
> 
> when  +86 136  make a call to ->
> +86 020 87654321  -> asterisk
> 
> the CALLERID(num)  will show the caller number +86136
> the ${EXTEN} is the dialed number  87654321
> i will try the CALLERID(dnid) tomorrow, will this get the whole dialed
> number 87654321 ?
> i don't know whether the sp will send me the whole number
> 
> --
> ssmax
> 2009-03-12
> 
> -
> 发件人：Jimmy Godbout
> 发送日期：2009-03-12 21:35:06
> 收件人：Asterisk Users Mailing List - Non-Commercial Discussion
> 抄送：
> 主题：Re: [asterisk-users] Is it possible to get full callin number fromE1?
> 
> ssmax,
> 
> Use CALLERID(num) to get the number that was dialed.
> 
> Jimmy
> 
>> -Original Message-
>> From: ss...@126.com
>> Sent: Thu, 12 Mar 2009 20:40:59 +0800
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] Is it possible to get full callin number from
>> E1?
>> 
>> Hi all
>> 
>> i have just set up a asterisk in china, using DE410P and one E1 line
>> and get a phone number like: +86 020 87654321 from my sp
>> when somebody dial  +86 020 87654321 , the asterisk will get the
>> call
>> in number by ${EXTEN} variable, but it can only get 87654321, no area
>> code .
>> when someone dial  +86 020 87654321 ,  means 4 digits,  the
>> phone can call in, and the  ${EXTEN}  is only  87654321 too ,
>> is it possible to get  full call in number 87654321  in asterisk
>> ?  thanks
>> 
>> --
>> ssmax
>> 2009-03-12
>> 
>> 
>> 
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> 
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> GET FREE 5GB EMAIL - Check out spam free email with many cool features!
> Visit http://www.inbox.com/email to find out more!
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


GET FREE 5GB EMAIL - Check out spam free email with many cool features!
Visit http://www.inbox.com/email to find out more!

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Sending faxes with T.38 problem. Asterisk - 1.6.0.6

[David Backeberg]

On Wed, Mar 11, 2009 at 7:32 AM, Santiago Gimeno
 wrote:
> I finally solved the issue by changing the resolution and the width of the
> TIFF file to one that is accepted by the fax standard. In my case I changed
> to a resolution of 96x96 and a width of 1728.
>
> Now I am able to send faxes, but something weird is happening, the fax
> received in the fax-machine has the black and white colours inverted. Any
> ideas why this could be happening?

The way I got my tiff file for testing was to use ReceiveFax to make a
tiff from an inbound fax.

I then used that tiff outbound for testing outbound faxing.

Something you might want to consider doing?

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Is it possible to get full callin number fromE1?

[ssmax]

hi,Jimmy Godbout

when  +86 136  make a call to ->
+86 020 87654321  -> asterisk

the CALLERID(num)  will show the caller number +86136
the ${EXTEN} is the dialed number  87654321
i will try the CALLERID(dnid) tomorrow, will this get the whole dialed number 
87654321 ?
i don't know whether the sp will send me the whole number

--   
ssmax
2009-03-12

-
发件人：Jimmy Godbout
发送日期：2009-03-12 21:35:06
收件人：Asterisk Users Mailing List - Non-Commercial Discussion
抄送：
主题：Re: [asterisk-users] Is it possible to get full callin number fromE1?

ssmax,

Use CALLERID(num) to get the number that was dialed.

Jimmy

> -Original Message-
> From: ss...@126.com
> Sent: Thu, 12 Mar 2009 20:40:59 +0800
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Is it possible to get full callin number from
> E1?
> 
> Hi all
> 
> i have just set up a asterisk in china, using DE410P and one E1 line
> and get a phone number like: +86 020 87654321 from my sp
> when somebody dial  +86 020 87654321 , the asterisk will get the call
> in number by ${EXTEN} variable, but it can only get 87654321, no area
> code .
> when someone dial  +86 020 87654321 ,  means 4 digits,  the
> phone can call in, and the  ${EXTEN}  is only  87654321 too ,
> is it possible to get  full call in number 87654321  in asterisk
> ?  thanks
> 
> --
> ssmax
> 2009-03-12
> 
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


GET FREE 5GB EMAIL - Check out spam free email with many cool features!
Visit http://www.inbox.com/email to find out more!

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] BRI/ISDN, misdn.conf/misdn-init.conf, OpenVOX B100P and Etisalat in Dubai

[Giorgio Incantalupo]

Hi Gavin,

if you can make and receive calls it works...do not worry if your line 
is shown as DOWN, some telco turns it off but it works without problem.
Remember to ask your telco for the right signalling and set it the right 
way (PTP or PMP).

Giorgio Incantalupo

Gavin Henry wrote:
> Hi All,
>
> We've got msidn configured:
>
> Port  1: TE-mode BRI S/T interface line (for phone lines)
>  -> Protocol: DSS1 (Euro ISDN)
>  -> childcnt: 2
> 
>
> mISDN_close: fid(3) isize(131072) inbuf(0x8fd5060) irp(0x8fd5060)
> iend(0x8fd5060)
>
>
>  and running on Asterisk 1.4.21.2:
>
> pbx*CLI> misdn show stacks
> BEGIN STACK_LIST:
>   * Port 1 Type TE Prot. PMP L2Link UP L1Link:UP Blocked:0  Debug:0
>
> but I'm not sure how to check our settings are right with Etisalat, as
> above then goes to:
>
> pbx*CLI> misdn show stacks
> BEGIN STACK_LIST:
>   * Port 1 Type TE Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0  Debug:0
>
> /etc/misdn-init.conf:
>
> card=1,hfcpci
> te_ptmp=1
> poll=128
> dsp_poll=128
> dsp_options=0
> dtmfthreshold=100
> debug=0
>
> /etc/asterisk/misdn.conf:
>
> [general]
> debug = 0
> method=standard
> bridging=no
> stop_tone_after_first_digit=yes
> append_digits2exten=yes
> dynamic_crypt=no
> crypt_prefix=**
> crypt_keys=test,muh
>
> [default]
> context=misdn
> language=en
> musicclass=default
> senddtmf=yes
> far_alerting=no
> allowed_bearers=all
> nationalprefix=0
> internationalprefix=00
> rxgain=0
> txgain=0
> te_choose_channel=no
> pmp_l1_check=no
> reject_cause=16
> need_more_infos=no
> nttimeout=no
> method=standard
> dialplan=0
> localdialplan=0
> cpndialplan=0
> early_bconnect=yes
> incoming_early_audio=no
> nodialtone=no
> callgroup=1
> pickupgroup=1
> presentation=-1
> screen=-1
> echocancel=yes
> jitterbuffer=4000
> jitterbuffer_upper_threshold=0
> hdlc=no
>
> [isdn]
> ports=1
> context=from-pstn
> msns=*
>
>
> Now we've setup PRi before, but not BRI and not in Dubai.
>
> If anyone has an idea or config that would be great and I'd make sure
> it goes on voip-info.org for others too.
>
> Thanks,
> Gavin.
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>   



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] how to configure for incoming message-summary SUBSCRIBE

[Klaus Darilion]

answering myself ...

Klaus Darilion schrieb:
> Hi!
> 
> AFAIS the incoming SUBSCRIBE is handled in the same context as INVITE - 

This is a bug which is fixed in Asterisk 1.4 branch, thus probably will 
be fixed in 1.4.24

regards
klaus


> but how should I handle the SUBSCRIBE in the context?
> 
> thanks
> klaus
> 
> SUBSCRIBE sip:u+431234...@foobar.at:5160 SIP/2.0
> Via: SIP/2.0/UDP 
> 192.168.2.82:39982;branch=z9hG4bK-d8754z-3116e1207913aa4e-1---d8754z-;rport
> Max-Forwards: 70
> Contact: 
> To: "schlopy"
> From: "schlopy";tag=376b6b2e
> Call-ID: ZDY1MmExZDdlNGE0MGI0NzgxZGQxMjA5YWNmMTZiYzA.
> CSeq: 2 SUBSCRIBE
> Expires: 300
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
> SUBSCRIBE, INFO
> User-Agent: X-Lite release 1100l stamp 47546
> Authorization: Digest 
> username="u+431234567",realm="foobar.at",nonce="fdsfa709",uri="sip:u+431234...@foobar.at:5160",response="foobar",algorithm=MD5
> Event: message-summary
> Content-Length: 0
> 
> <->
> --- (14 headers 0 lines) ---
> Creating new subscription
> Sending to 81.189.99.50 : 39982 (NAT)
> Found peer 'u+431234567'
> Looking for u+431234567 in fromSipPots (domain foobar.at)
> 
> <--- Transmitting (NAT) to 11.111.11.11:39982 --->
> SIP/2.0 404 Not Found
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] BRI/ISDN, misdn.conf/misdn-init.conf, OpenVOX B100P and Etisalat in Dubai

[Gavin Henry]

Hi All,

We've got msidn configured:

Port  1: TE-mode BRI S/T interface line (for phone lines)
 -> Protocol: DSS1 (Euro ISDN)
 -> childcnt: 2


mISDN_close: fid(3) isize(131072) inbuf(0x8fd5060) irp(0x8fd5060)
iend(0x8fd5060)


 and running on Asterisk 1.4.21.2:

pbx*CLI> misdn show stacks
BEGIN STACK_LIST:
  * Port 1 Type TE Prot. PMP L2Link UP L1Link:UP Blocked:0  Debug:0

but I'm not sure how to check our settings are right with Etisalat, as
above then goes to:

pbx*CLI> misdn show stacks
BEGIN STACK_LIST:
  * Port 1 Type TE Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0  Debug:0

/etc/misdn-init.conf:

card=1,hfcpci
te_ptmp=1
poll=128
dsp_poll=128
dsp_options=0
dtmfthreshold=100
debug=0

/etc/asterisk/misdn.conf:

[general]
debug = 0
method=standard
bridging=no
stop_tone_after_first_digit=yes
append_digits2exten=yes
dynamic_crypt=no
crypt_prefix=**
crypt_keys=test,muh

[default]
context=misdn
language=en
musicclass=default
senddtmf=yes
far_alerting=no
allowed_bearers=all
nationalprefix=0
internationalprefix=00
rxgain=0
txgain=0
te_choose_channel=no
pmp_l1_check=no
reject_cause=16
need_more_infos=no
nttimeout=no
method=standard
dialplan=0
localdialplan=0
cpndialplan=0
early_bconnect=yes
incoming_early_audio=no
nodialtone=no
callgroup=1
pickupgroup=1
presentation=-1
screen=-1
echocancel=yes
jitterbuffer=4000
jitterbuffer_upper_threshold=0
hdlc=no

[isdn]
ports=1
context=from-pstn
msns=*


Now we've setup PRi before, but not BRI and not in Dubai.

If anyone has an idea or config that would be great and I'd make sure
it goes on voip-info.org for others too.

Thanks,
Gavin.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Is it possible to get full callin number from E1?

[Jimmy Godbout]

ssmax,

Use CALLERID(num) to get the number that was dialed.

Jimmy

> -Original Message-
> From: ss...@126.com
> Sent: Thu, 12 Mar 2009 20:40:59 +0800
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Is it possible to get full callin number from
> E1?
> 
> Hi all
> 
> i have just set up a asterisk in china, using DE410P and one E1 line
> and get a phone number like: +86 020 87654321 from my sp
> when somebody dial  +86 020 87654321 , the asterisk will get the call
> in number by ${EXTEN} variable, but it can only get 87654321, no area
> code .
> when someone dial  +86 020 87654321 ,  means 4 digits,  the
> phone can call in, and the  ${EXTEN}  is only  87654321 too ,
> is it possible to get  full call in number 87654321  in asterisk
> ?  thanks
> 
> --
> ssmax
> 2009-03-12
> 
> 
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


GET FREE 5GB EMAIL - Check out spam free email with many cool features!
Visit http://www.inbox.com/email to find out more!

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Is it possible to get full callin number from E1?

[ssmax]

Hi all

i have just set up a asterisk in china, using DE410P and one E1 line
and get a phone number like: +86 020 87654321 from my sp
when somebody dial  +86 020 87654321 , the asterisk will get the call in 
number by ${EXTEN} variable, but it can only get 87654321, no area code .
when someone dial  +86 020 87654321 ,  means 4 digits,  the phone 
can call in, and the  ${EXTEN}  is only  87654321 too , 
is it possible to get  full call in number 87654321  in asterisk ?  
thanks

--
ssmax
2009-03-12



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Are .call files working with extensions.ael ? bristuff problem

[Peer Oliver Schmidt]

Olivier wrote:
>> Do you by chance use bristuff?
> 
> Yes, I do.

bristuff patches pbx/pbx_spool.c

I have no knowledge of C, but there seems to be a problem around line 
266.

The original line (pre-bristuff) looks like this:

if (ast_strlen_zero(o->tech) || ast_strlen_zero(o->dest) || 
(ast_strlen_zero(o->app) && ast_strlen_zero(o->exten))) {


The patched line looks like this:

if (ast_strlen_zero(o->tech) || ast_strlen_zero(o->dest) || 
(ast_strlen_zero(o->app) && ast_strlen_zero(o->exten)) || 
(ast_strlen_zero(o->message) && ast_strlen_zero(o->pdu))) {

Try reverting that line, and see if that helps with your problem. And 
maybe someone with a better understanding of C can take a look at the 
above problem.
-- 
Best regards

Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] VLC

[Bex Vincent]

Hi Steve,

I am not too much worried about the quality, but soon we will have 7000 users 
with voicemail. We don't have accurate statistics on the voicemail usage but it 
might well be that it will take a lot of space on the servers.

Ogg has been implemented in asterisk but it seems that nobody uses it.

Regards

vincent

-Message d'origine-
De : asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Steve Edwards
Envoyé : mercredi, 11. mars 2009 21:19
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] VLC

On Wed, 11 Mar 2009, Bex Vincent wrote:

> When our users receive a voicemail we send it attached to an email. It 
> used to work fine, encoded in wav49 and read by Windows media player. 
> Recently the default player in the company has become VLC which is 
> unable to read wav49. I am trying to use OGG/VORBIS instead of wav49. I 
> can't get it working:

Would "format = wav" do?

While the files are 10x bigger, it's still only 1mb per minute.

Unless you have to retain the files for a very long time and/or have a 
huge number of users, I can't see spending the CPU time to compress and 
decompress something that will probably only be listened to once and 
discarded.

Personally, the increase in fidelity is good enough reason to me.

Try "format = wav|wav49" and listen to the files with a decent set of 
speakers. I know the typical handset approaches two cans and a piece of 
wet string fidelity wise, but, since you say "attached to an email," your 
users will hear the difference.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Printing faxes

[Doug Lytle]

voip crazy wrote:
> Hello list,
>
> is working nicely. Now I need the fax to be print when arriving.
>
> ¿Anybody have this feature implementing in their systems?
>   

We have several network printers mapped on the HylaFAX+ server and do 
the following from the FaxDispatch script:


FILETYPE=tif;
tiff2pdf $FULLPATH -p letter -o faxdata/$CALLID4/pdf/$FILENAME.pdf
lpr -P HPLasJet1320 faxdata/$CALLID4/pdf/$FILENAME.pdf
cp $FULLPATH /var/spool/hylafax/faxdata/$CALLID4/tif/
;;

Doug


-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Are .call files working with extensions.ael ?

[Olivier]

2009/3/12 Peer Oliver Schmidt 

> Hello Olivier,
>
> > With an extensions.ael enabled system, I keep getting whatever I change
> > into my "astup.call" file :
> >
> > [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At
> > least one of app or extension (or keyword message/pdu) must be
> > specified, along with tech and dest in file
> > /var/spool/asterisk/outgoing/astup.call
> > [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service: Invalid
> > file contents in /var/spool/asterisk/outgoing/astup.call, deleting
> > [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:505 scan_thread: Failed to
> > scan service '/var/spool/asterisk/outgoing/astup.call'
>
> Do you by chance use bristuff?

Yes, I do.

>
>
> --
> Best regards
>
> Peer Oliver Schmidt
> PGP Key ID: 0x83E1C2EA
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] VLC

[Bex Vincent]

Hi,

I thought of this but I don't know where to intercept the file to convert it. 
The email is automatically sent, this is configured in voicemail.conf. I tried 
to change the mailcmd with a custom script thinking I would get the parameters 
passed to sendmail but it doesn't work. Has anybody an idea what the mail 
sending mechanism is?

Regards
vincent

De : asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Danny Nicholas
Envoyé : mercredi, 11. mars 2009 14:35
À : 'Asterisk Users Mailing List - Non-Commercial Discussion'
Objet : Re: [asterisk-users] VLC




From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bex Vincent
Sent: Wednesday, March 11, 2009 3:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] VLC

Hi All,

When our users receive a voicemail we send it attached to an email. It used to 
work fine, encoded in wav49 and read by Windows media player. Recently the 
default player in the company has become VLC which is unable to read wav49. I 
am trying to use OGG/VORBIS instead of wav49. I can't get it working:

In voicemail.conf:

format = ogg

The result is as follow:

[Mar 11 09:42:17] WARNING[24867]: format_ogg_vorbis.c:527 ogg_vorbis_seek: 
Seeking is not supported on OGG/Vorbis streams!
-- x=0, open writing:  
/var/spool/asterisk/voicemail/default/0213149689/tmp/V2adNF format: ogg, 
0x81ec648
-- User hung up
[Mar 11 09:42:25] WARNING[24867]: format_ogg_vorbis.c:533 ogg_vorbis_tell: 
Telling is not supported on OGG/Vorbis streams!
[Mar 11 09:42:25] WARNING[24867]: format_ogg_vorbis.c:514 ogg_vorbis_trunc: 
Truncation is not supported on OGG/Vorbis streams!
-- Recording was 0 seconds long but needs to be at least 2 - abandoning

Nothing gets recorded in the file.

Has anybody done this before? Either get ogg work with voicemail or get VLC to 
read wav49.

Cheers
Vincent

Why don't you just install SOX and convert the file that way?  You could just 
do a command like this
System(/usr/bin/sox in.ogg out.mp3)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Are .call files working with extensions.ael ?

[Peer Oliver Schmidt]

Hello Olivier,

> With an extensions.ael enabled system, I keep getting whatever I change 
> into my "astup.call" file :
> 
> [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At 
> least one of app or extension (or keyword message/pdu) must be 
> specified, along with tech and dest in file 
> /var/spool/asterisk/outgoing/astup.call
> [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service: Invalid 
> file contents in /var/spool/asterisk/outgoing/astup.call, deleting
> [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:505 scan_thread: Failed to 
> scan service '/var/spool/asterisk/outgoing/astup.call'

Do you by chance use bristuff?

-- 
Best regards

Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Printing faxes

[Grygoriy Dobrovolskyy]

2009/3/12 voip crazy 

> Hello list,
>
> I have an asterisk / hylafax / iaxmodem configured in one machine. All
> is working nicely. Now I need the fax to be print when arriving.
>
> ¿Anybody have this feature implementing in their systems?
>
> ¿How is the best way to get that?
>
> Any clue will be welcomed.
>
> Thanks.
>
> VoipCrazy
>
> ___
>

I am printing at the end of faxrcvd script, just configure printer you like
read the script, it is pretty simple, and add some command for lpd for that
$File.
Hope i helped.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Printing faxes

[Grygoriy Dobrovolskyy]

2009/3/12 Tristan 

> Hi,
>
> Send it to cups via the FaxDispatch script ;)
>
> Regards,
>
> Tristan
>
> voip crazy a écrit :
> > Hello list,
> >
> > I have an asterisk / hylafax / iaxmodem configured in one machine. All
> > is working nicely. Now I need the fax to be print when arriving.
> >
> > ¿Anybody have this feature implementing in their systems?
> >
> > ¿How is the best way to get that?
> >
> > Any clue will be welcomed.
> >
> > Thanks.
> >
> > VoipCrazy
> >
> > ___
>

Oh sorry i meant Faxdispatch not faxrcvd ;)
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATCom Phones - AT 510/AT530

[Administrator TOOTAI]

Gordon Henderson a écrit :
> Anyone here used these phones?
>   
We stopped to use them, not stable IAX or SIP. A customer ask us to 
exchange 26 pieces after 6 month battling to make the phones working. We 
did it.

[...]

-- 
Daniel

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Printing faxes

[Tristan]

Hi,

Send it to cups via the FaxDispatch script ;)

Regards,

Tristan

voip crazy a écrit :
> Hello list,
>
> I have an asterisk / hylafax / iaxmodem configured in one machine. All
> is working nicely. Now I need the fax to be print when arriving.
>
> ¿Anybody have this feature implementing in their systems?
>
> ¿How is the best way to get that?
>
> Any clue will be welcomed.
>
> Thanks.
>
> VoipCrazy
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATCom Phones - AT 510/AT530

[bails]

Gordon Henderson wrote:
> Anyone here used these phones?
> 
> I'm getting more and more frustrated by todays modern crop of routers with 
> their so-called SIP ALGs which are invariably broken, or routers with 
> built-in ATAs which block internal SIP phones from working, so looking to 
> use IAX for some end-users.
> 
> I already support it for people who want to use (eg) Zoiper and use IAX a 
> lot to plumb boxes together, but never used IAX for many end-uers.
> 
> Atcom seem to make a lot of Asterisk compatible kit, so I'd hope the 
> phones were OK, but anyone here actually used them - functionality, voice 
> quality, general "usability" and so on?
> 
> I'm not looking for a high-end phone here - the sort of people who're 
> going to a "hosted" solution aren't those people (at least in my market), 
> so what I'm after is a "fire and forget" solution - as by the time you've 
> contacted the customer, persuaded them to buy a a new router, gone 
> on-site, etc. you've lost any margin possible...
> 
> So any comments on them welcome!

I had 9 of these set up as remote IAX phones for a customer.  After a 
few days of testing and many complaints about register timeouts we 
configured them to be SIP phones and I never had a complaint again.

They are IMHO physically too light, nothing a chunk of lead wont fix though!

Thanks

Bails

> 
> Cheers,
> 
> Gordon
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 


-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Printing faxes

[voip crazy]

Hello list,

I have an asterisk / hylafax / iaxmodem configured in one machine. All
is working nicely. Now I need the fax to be print when arriving.

¿Anybody have this feature implementing in their systems?

¿How is the best way to get that?

Any clue will be welcomed.

Thanks.

VoipCrazy

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] UK ISDN-30 and ANI

[Julian Lyndon-Smith]

Has anyone in the UK got ANI to work on an inbound call ?

Using asterisk 1.4 trunk and zaptel 1.4 trunk, with a Euro-ISDN 30

Julian

__
This email has been scanned by the MessageLabs Email Security System.
For more information please visit http://www.messagelabs.com/email 
__

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Portech MV3770 & Caller-ID

[Szabó András]

Hi
It's the same, *31# for Hungary. It may be common then.

Cheers
András


2009/3/12 

>
> On 3/11/2009, "Hĺkan Källberg"  wrote:
>
> >On Wed, Mar 11, 2009 at 04:16:43PM +0100, Christian Victor wrote:
> >> 2009/3/11 Hĺkan Källberg 
> >> > Does anyone of you have Caller Presentation working in the other
> >> > direction?? My mv370 is working well, execpt the Caller ID on outgoing
> >> > GSM calls. This works with the SIM card/Provider I am using if I put
>
> >> All I had to do is to enable the Caller ID ind the Mobile->Settings
> dialog
> >> for each SIM (something like presentation/revocation afair). I did NOT
> set
> >> the GSM number anywhere nor do I send it from Asterisk.
>
> >That is what I'd expect too, but, no...
> >
> >Mobile->Settings->CLID Presentation-> Supression or Invocation
> >
> >it makes no difference. (and yes - I do reboot:-) When I move the SIM
> >to a phone, it works well...
> >
>
>
> I'm not sure wether it's an operator-specific setting or not.  But i
> don't handle callerid supression/invocation with the MV370, i rather do
> it with asterisk.
>
> Try simply prefix the number with *31# for invocation and #31# for
> supression.  Example:
>
> [...]
> exten => _06[237]0NXX!,n,Dial(SIP/*31#${ext...@gsmgw)
> [...]
>
> Again: this method may be country or even operator specific, check with
> the provider if the above works.  If it does, simply use the prefixes
> above and forget about the MV370's settings.
>
>
>
>
> regards
> adam
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Manager API and astmanproxy

[Julian Lyndon-Smith]

For a long time now we've used the astmanproxy process to handle manager 
connections (75+ clients) so that these clients can tap into the 
dialplan / send commands etc.

We use astmanproxy because at that time the manager connection routines 
of asterisk did not cope well with numerous connections, either 
permanent or as-needed.

Does anyone have any experience of later versions of asterisk (we 
currently use 1.4 trunk, although we started to use astmanproxy with 
1.2)  and how the manager API works under a large load ?

I want to remove the need to have astmanproxy (just another process that 
may go wrong).

Julian


__
This email has been scanned by the MessageLabs Email Security System.
For more information please visit http://www.messagelabs.com/email 
__

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ATCom Phones - AT 510/AT530

[Bruno Castelo Branco]

hi

I had use more than 200 pcs of atcom model at530 phones with IAX and SIP 
for more than 3 years whit no problems until now.


Gordon Henderson wrote:
> Anyone here used these phones?
>
> I'm getting more and more frustrated by todays modern crop of routers with 
> their so-called SIP ALGs which are invariably broken, or routers with 
> built-in ATAs which block internal SIP phones from working, so looking to 
> use IAX for some end-users.
>
> I already support it for people who want to use (eg) Zoiper and use IAX a 
> lot to plumb boxes together, but never used IAX for many end-uers.
>
> Atcom seem to make a lot of Asterisk compatible kit, so I'd hope the 
> phones were OK, but anyone here actually used them - functionality, voice 
> quality, general "usability" and so on?
>
> I'm not looking for a high-end phone here - the sort of people who're 
> going to a "hosted" solution aren't those people (at least in my market), 
> so what I'm after is a "fire and forget" solution - as by the time you've 
> contacted the customer, persuaded them to buy a a new router, gone 
> on-site, etc. you've lost any margin possible...
>
> So any comments on them welcome!
>
> Cheers,
>
> Gordon
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] ATCom Phones - AT 510/AT530

[Gordon Henderson]

Anyone here used these phones?

I'm getting more and more frustrated by todays modern crop of routers with 
their so-called SIP ALGs which are invariably broken, or routers with 
built-in ATAs which block internal SIP phones from working, so looking to 
use IAX for some end-users.

I already support it for people who want to use (eg) Zoiper and use IAX a 
lot to plumb boxes together, but never used IAX for many end-uers.

Atcom seem to make a lot of Asterisk compatible kit, so I'd hope the 
phones were OK, but anyone here actually used them - functionality, voice 
quality, general "usability" and so on?

I'm not looking for a high-end phone here - the sort of people who're 
going to a "hosted" solution aren't those people (at least in my market), 
so what I'm after is a "fire and forget" solution - as by the time you've 
contacted the customer, persuaded them to buy a a new router, gone 
on-site, etc. you've lost any margin possible...

So any comments on them welcome!

Cheers,

Gordon

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Grandstream speakerphone?

[Gordon Henderson]

On Wed, 11 Mar 2009, Lutgring, Sam wrote:

> I have been using a number of the Grandstream GXP-2000 (74 in 
> production), GXP-2010 (1 in production), and BT-200 (15 in production) 
> with great success.  The only issue that we have had is killing power 
> supplies, not sure if this is related to our power source or product. 
> So far they have replaced all of them for us.  I prefer the speaker 
> phone on the GXP-2010 and the BT-200 phones.

I've used Grandtreams for a long time and while they have their faults, 
the speaker phone is actually very good. So much so that callers have not 
known I'm on speakerphone when I've used it. They are a little bit 
plasticy, a bit light and the audio quality (when not on speakerphone) 
isn't quite as good as Snoms, but I've not found a phone to beat it at 
that price...

(Saying that, see next post!)

Gordon


>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
> Sent: Wednesday, March 11, 2009 5:06 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Grandstream speakerphone?
>
> Idle curiosity: I like the look and feel of the Grandstreams, but it's
> been my experience that the speakerphones suck (esp. when compared to the
> pretty damn flawless Polycoms).  I've used the BT-100/101 and GS-2000;
> have any of their newer models changed that?
>
> Thanks!
>
> -Ken
>
>
> --
> This message has been scanned for viruses and
> dangerous content by MailScanner, and is
> believed to be clean.
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Portech MV3770 & Caller-ID

On 3/11/2009, "Hĺkan Källberg"  wrote:

>On Wed, Mar 11, 2009 at 04:16:43PM +0100, Christian Victor wrote:
>> 2009/3/11 Hĺkan Källberg 
>> > Does anyone of you have Caller Presentation working in the other
>> > direction?? My mv370 is working well, execpt the Caller ID on outgoing
>> > GSM calls. This works with the SIM card/Provider I am using if I put

>> All I had to do is to enable the Caller ID ind the Mobile->Settings dialog
>> for each SIM (something like presentation/revocation afair). I did NOT set
>> the GSM number anywhere nor do I send it from Asterisk.

>That is what I'd expect too, but, no...
>
>Mobile->Settings->CLID Presentation-> Supression or Invocation
>
>it makes no difference. (and yes - I do reboot:-) When I move the SIM
>to a phone, it works well...
>


I'm not sure wether it's an operator-specific setting or not.  But i
don't handle callerid supression/invocation with the MV370, i rather do
it with asterisk.

Try simply prefix the number with *31# for invocation and #31# for
supression.  Example:

[...]
exten => _06[237]0NXX!,n,Dial(SIP/*31#${ext...@gsmgw)
[...]

Again: this method may be country or even operator specific, check with
the provider if the above works.  If it does, simply use the prefixes
above and forget about the MV370's settings.




regards
adam

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] (no subject)

[Umar Lais]

  ___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users