Re: [asterisk-users] Asterisk to PBX

[Trevor Hammonds]

On Fri, Jul 17, 2009 logan wrote:
> Hi,
>
> I'm an absolute newbie and wanted to know the following.
>
> I want to have a setup where I have a PSTN line connected to my
> Asterisk box and want to know if it is possible to make more than one
> simultaneous outbound call through that VoIP gateway? Can Asterisk do
> this magic of concurrent calls on one PSTN line?? If I put it in other
> words then can I receive more than one simultaneous call on a PSTN
> number through Asterisk (the dialplan would forward those calls to
> different extensions) and the phone line still be able to receive more
> calls?
>
> Do I need some special hardware for the above or a simple SIPURA3000
> would be good enough?
>
> Please pardon me if this is not the correct list for this question.
>
> Thanks.
>
> Best Regards,
> Hitesh

Hitesh,

The short answer is no.  The long answer is that Asterisk (and,
indeed, any other PBX) is not be able to make or receive any more
analogue calls than the number of available analogue trunks (or lines)
to which it has access.  You may, however, use an ITSP to send
outbound calls over the Internet, leaving your analogue line free for
an incoming call.  You may also be able to subscribe to Busy Call
Forwarding on your POTS line.  This way, when your analogue line is in
use, additional inbound calls may be forwarded to a DID (telephone
number) provided by an ITSP -- essentially increasing the number of
"lines" available to Asterisk.

Sincerely,
Trevor Hammonds

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[asterisk-users] Asterisk to PBX

[logan]

Hi,

I'm an absolute newbie and wanted to know the following.

I want to have a setup where I have a PSTN line connected to my
Asterisk box and want to know if it is possible to make more than one
simultaneous outbound call through that VoIP gateway? Can Asterisk do
this magic of concurrent calls on one PSTN line?? If I put it in other
words then can I receive more than one simultaneous call on a PSTN
number through Asterisk (the dialplan would forward those calls to
different extensions) and the phone line still be able to receive more
calls?

Do I need some special hardware for the above or a simple SIPURA3000
would be good enough?

Please pardon me if this is not the correct list for this question.

Thanks.

Best Regards,
Hitesh

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[asterisk-users] Possible WaitUntil Bug

[Trevor Hammonds]

I am having trouble with the WaitUntil application in Asterisk
SVN-branch-1.6.1-r206879.  I believe this is a bug, but would like
confirmation.

The relevant dialplan entry is:

exten => 8765,n,WaitUntil(${FutureTime})

The console indicates the following notice:

[Jul 17 17:13:57] NOTICE[4609]: app_waituntil.c:72 waituntil_exec:
WaitUntil called in the past (now 1247876037, arg 1247876040)

After this, the dialplan continues to the next entry without having waited.

My C programming skills are very basic, but looking at
app_waituntil.c, I do not see anything that actually waits for the
appropriate Unix epoch as in the original patch.  However, the
function that evaluates if the argument is called in the past is in
error as 1247876037 < 1247876040.

Original patchl:
https://issues.asterisk.org/view.php?id=11487

I would appreciate if anyone else is able to confirm this error.

Sincerely,
Trevor Hammonds

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[asterisk-users] Truecall

[Gavin Henry]

This has to be an Asterisk based appliance no?

http://www.truecall.co.uk/acatalog/trueCall_Features.html

Looks pretty easy to setup using AstLinux or similar.

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Re: [asterisk-users] 2 Problems with 1.6.2

[Ira]

At 01:09 PM 7/17/2009, you wrote:
Sorry, that's the most frequent
problem that people have with MWI in 1.6, so
it was worth mentioning.  I would suggest that you file a bug report
on

https://issues.asterisk.org.  It would be helpful if you would
include SIP
debug output for both a machine that is working, as well as a machine
that is
Not so embarrassing as I thought, I kept my notes and so here is the SIP
output. I attached it in case the inserted section gets all messed
up.
Reliably Transmitting (no NAT) to
192.168.233.237:5060: 
NOTIFY sip:10277x...@192.168.233.237:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK3b6f21b4
Max-Forwards: 70
From: Robyns ;tag=as405c743f
To: Ira
;tag=90f0be2565982a7
Contact: 
Call-ID: 5381b179eb2eec40ffa7dc855e671...@192.168.233.237
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX 1.6.2.0-beta3
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 210



confirmed


---
<--- SIP read from UDP:192.168.233.237:5060 --->
SIP/2.0 200 OK
Call-ID: 5381b179eb2eec40ffa7dc855e671...@192.168.233.237
CSeq: 105 NOTIFY
From: Robyns ;tag=as405c743f
To: Ira
;tag=90f0be2565982a7
Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK3b6f21b4
Content-Length: 0
Contact: Ira

Supported: replaces
User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0
MxSF/v3.2.8.45

<->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
Reliably Transmitting (no NAT) to 192.168.233.237:5060:
NOTIFY sip:10277x...@192.168.233.237:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK5cb290bd
Max-Forwards: 70
From: Robyns ;tag=as405c743f
To: Ira
;tag=90f0be2565982a7
Contact: 
Call-ID: 5381b179eb2eec40ffa7dc855e671...@192.168.233.237
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX 1.6.2.0-beta3
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 211



terminated


---
<--- SIP read from UDP:192.168.233.237:5060 --->
SIP/2.0 200 OK
Call-ID: 5381b179eb2eec40ffa7dc855e671...@192.168.233.237
CSeq: 106 NOTIFY
From: Robyns ;tag=as405c743f
To: Ira
;tag=90f0be2565982a7
Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK5cb290bd
Content-Length: 0
Contact: Ira

Supported: replaces
User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0
MxSF/v3.2.8.45

<->
--- (10 headers 0 lines) ---



Reliably Transmitting (no NAT) to 192.168.233.237:5060: 
NOTIFY sip:10277x...@192.168.233.237:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK3b6f21b4
Max-Forwards: 70
From: Robyns ;tag=as405c743f
To: Ira ;tag=90f0be2565982a7
Contact: 
Call-ID: 5381b179eb2eec40ffa7dc855e671...@192.168.233.237
CSeq: 105 NOTIFY
User-Agent: Asterisk PBX 1.6.2.0-beta3
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 210




confirmed


---
<--- SIP read from UDP:192.168.233.237:5060 --->
SIP/2.0 200 OK
Call-ID: 5381b179eb2eec40ffa7dc855e671...@192.168.233.237
CSeq: 105 NOTIFY
From: Robyns ;tag=as405c743f
To: Ira ;tag=90f0be2565982a7
Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK3b6f21b4
Content-Length: 0
Contact: Ira 
Supported: replaces
User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45


<->
--- (10 headers 0 lines) ---
SIP Response message for INCOMING dialog NOTIFY arrived
Reliably Transmitting (no NAT) to 192.168.233.237:5060:
NOTIFY sip:10277x...@192.168.233.237:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK5cb290bd
Max-Forwards: 70
From: Robyns ;tag=as405c743f
To: Ira ;tag=90f0be2565982a7
Contact: 
Call-ID: 5381b179eb2eec40ffa7dc855e671...@192.168.233.237
CSeq: 106 NOTIFY
User-Agent: Asterisk PBX 1.6.2.0-beta3
Event: dialog
Content-Type: application/dialog-info+xml
Subscription-State: active
Content-Length: 211




terminated


---
<--- SIP read from UDP:192.168.233.237:5060 --->
SIP/2.0 200 OK
Call-ID: 5381b179eb2eec40ffa7dc855e671...@192.168.233.237
CSeq: 106 NOTIFY
From: Robyns ;tag=as405c743f
To: Ira ;tag=90f0be2565982a7
Via: SIP/2.0/UDP 192.168.233.235:5060;branch=z9hG4bK5cb290bd
Content-Length: 0
Contact: Ira 
Supported: replaces
User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45


<->
--- (10 headers 0 lines) ---

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Re: [asterisk-users] Delete voicemail after couple of days

[Carlos Chavez]

I did not catch all the messages on this thread but why not use the
messages-expire.pl script included in Asterisk for this simple task?  It
will delete and renumber all messages and you can program how many days
before a message is deleted.

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] 2 Problems with 1.6.2

[Ira]

At 01:09 PM 7/17/2009, you wrote:
>Sorry, that's the most frequent problem that people have with MWI in 1.6, so
>it was worth mentioning.  I would suggest that you file a bug report on
>https://issues.asterisk.org.  It would be helpful if you would include SIP
>debug output for both a machine that is working, as well as a machine that is
>not working.

So I'd be more than happy to file a bug report and include all the 
SIP debug anyone might need but it's been so many years since I did 
it that I've no idea how anymore.

So I grabbed a cordless handset, sat down at the console, typed sip 
set debug ip 192.xxx.xxx.xx and called that voicemail box to leave a 
message.  The instant I hung up a notify message was sent to my 
phone, but the red light did not come on. If you remind me the how, 
I'll grab that message and post it here.

Thanks so much for the help.

Ira



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Re: [asterisk-users] Delete voicemail after couple of days

[Tim Nelson]

- "Steve Edwards"  wrote:
> On Fri, 17 Jul 2009, Miguel Molina wrote:
> 
> >> I think the OP caught the humor -- note the "smiley." I'm sorry it
> 
> >> didn't translate to your language.
> 
> > Oops, well I'm not a native english speaker so it's really hard to
> catch 
> > some humor of a word that I don't know or I get as misspelled.
> Thanks 
> > for the definition, now I can laugh with you guys.
> >
> > Sorry for all the fuzz around this.
> >
> > PD: Es como si yo te contara un chiste en español!
> 
> Si, pero el Ingles es mejor que mi espanol!
> 
> (Google translate is my friend.)
> -- 

All the politics, list etiquette, and general bitching aside, here is how I 
would do what the OP wants.

Write up a small shell script that uses 'find /var/spool/asterisk/voicemail/ 
-mtime +2' for a list of files older than two days assuming you want ALL files 
deleted older than two days. You could always grep that output if you only 
wanted to delete voicemail that is not still in the INBOX or elsewhere. 
Anyways, then use -exec to rm the files. If the goal was to remove all files, 
it might look something like this:

#!/bin/bash
find /var/spool/asterisk/voicemail/ -mtime +2 -exec rm  {}\;

Run that from cron once a day/hour/whatever and you're set.


It still amazes me how often posters are unable to get a simple answer to a 
question and instead are inundated with 'you top posted', 'you didn't ask the 
question right', 'your spelling was wrong', etc...  I mean, is this list just a 
really big bridge with a bunch of trolls(no pun intended) waiting to pounce on 
people just wanting to get to the other side where "Asterisk Enlightenment" 
awaits?

And of course because I've diverted from the norm and possibly hurt someone's 
ego, I expect a full backlash or smarmy remarks etc. Thank you in advance.


--Tim

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Re: [asterisk-users] Voicemail ODBC storage table schema

[Hoggins!]

Thanks, problem solved.

Hoggins!


Tilghman Lesher a écrit :
> On Friday 17 July 2009 16:25:13 Hoggins! wrote:
>   
>> Hello,
>>
>> Upgraded from 1.6.1.0 to 1.6.1.1 and my voicemail setup does not work
>> anymore. I use ODBC storage for voicemail. Comes out that the
>> "voicemessages" table schema should have changed, because the log says
>> Asterisk needed to store data to an additional field called "flag". Any
>> new message cannot be saved.
>> The thing is that I'd like to know where I can find an updated schema
>> for the generic voicemail storage table. Apparently, only the "flag"
>> field has appeared, but I can't find out what is the type of the field.
>>
>> Here are the fields it's trying to update :
>>
>> [INSERT INTO voicemessages
>> (dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailb
>> oxuser,mailboxcontext,flag) VALUES (?,?,?,?,?,?,?,?,?,?,?)]
>>
>> I had to roll back to 1.6.1.0 in the meantime.
>> 
>
> Oops.  It's now documented in UPGRADE.txt and the table schema is in
> doc/tex/odbcstorage.tex (which is rendered into the PDF at release).
>
>   
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Re: [asterisk-users] Delete voicemail after couple of days

[Steve Edwards]

On Fri, 17 Jul 2009, Miguel Molina wrote:

I think the OP caught the humor -- note the "smiley." I'm sorry it 
didn't translate to your language.


Oops, well I'm not a native english speaker so it's really hard to catch 
some humor of a word that I don't know or I get as misspelled. Thanks 
for the definition, now I can laugh with you guys.


Sorry for all the fuzz around this.

PD: Es como si yo te contara un chiste en espa??ol!


Si, pero el Ingles es mejor que mi espanol!

(Google translate is my friend.)
--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
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Re: [asterisk-users] Voicemail ODBC storage table schema

[Tilghman Lesher]

On Friday 17 July 2009 16:25:13 Hoggins! wrote:
> Hello,
>
> Upgraded from 1.6.1.0 to 1.6.1.1 and my voicemail setup does not work
> anymore. I use ODBC storage for voicemail. Comes out that the
> "voicemessages" table schema should have changed, because the log says
> Asterisk needed to store data to an additional field called "flag". Any
> new message cannot be saved.
> The thing is that I'd like to know where I can find an updated schema
> for the generic voicemail storage table. Apparently, only the "flag"
> field has appeared, but I can't find out what is the type of the field.
>
> Here are the fields it's trying to update :
>
> [INSERT INTO voicemessages
> (dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailb
>oxuser,mailboxcontext,flag) VALUES (?,?,?,?,?,?,?,?,?,?,?)]
>
> I had to roll back to 1.6.1.0 in the meantime.

Oops.  It's now documented in UPGRADE.txt and the table schema is in
doc/tex/odbcstorage.tex (which is rendered into the PDF at release).

-- 
Tilghman & Teryl
with Peter, Cottontail, Midnight, Thumper, & Johnny (bunnies)
and Harry, BB, & George (dogs)

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Re: [asterisk-users] Delete voicemail after couple of days

[Miguel Molina]

Steve Edwards escribió:
> Un-top-posting...
>
>>> On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote:
>>>
>>> > Is there any tested script available for this purpose.
>
>>> On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards >> @sedwards.com > wrote:
>>>
>>> Sure. Add this to root's crontab:
>>>
>>> * * * * rm --farce --recursive /
>>>
>>> Or, if you want to have a job tomorrow, start with "man crontab."
>
>> Aloysius Thevarajah Lloyd escribi�:
>
>>> you want me to delete all the sytem files:)
>
> On Fri, 17 Jul 2009, Miguel Molina wrote:
>
>> Yeah he wants to make yourself silently blow your own system off to 
>> make you start from a beautiful clean fresh install or lose your job 
>> instantaneously. Fortunately, he did misspell the crontab (--force, 
>> one * more). It's a dangerous, agressive and sarcastic way to tell 
>> you that RTFM. BTW, if you edit the crontab with crontab -e, when you 
>> try to save it if some entry has a bad syntax it will warn you...
>
> From dictionary.com:
>
> "farce - a light, humorous play in which the plot depends upon a 
> skillfully exploited situation rather than upon the development of 
> character."
>
> I think the OP caught the humor -- note the "smiley." I'm sorry it 
> didn't translate to your language.
Oops, well I'm not a native english speaker so it's really hard to catch 
some humor of a word that I don't know or I get as misspelled. Thanks 
for the definition, now I can laugh with you guys.

Sorry for all the fuzz around this.

PD: Es como si yo te contara un chiste en español!

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center


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Re: [asterisk-users] 2 Problems with 1.6.2

[Ira]

At 01:09 PM 7/17/2009, you wrote:
>Sorry, that's the most frequent problem that people have with MWI in 1.6, so
>it was worth mentioning.  I would suggest that you file a bug report on
>https://issues.asterisk.org.  It would be helpful if you would include SIP
>debug output for both a machine that is working, as well as a machine that is
>not working.

No problem, I thought I'd read it or at least skimmed it, it's a 
small system, 3 POTS lines and 2 SIP numbers coming in, 3 SIP 
providers for outgoing calls, 3 Aastra 480i-CT handsets. We probably 
get 20 calls on a busy day and don't make many going out. Other than 
the POTs lines I spend under $10/month on phone calls and all 
outgoing calls use SIP.  I'd compare with a working system, but being 
the brave foolish sort, once 1.6.2 seemed to be mostly working the 
machine with 1.2 minus it's memory and HD went off to be recycled.

Ira 


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Re: [asterisk-users] 2 Problems with 1.6.2

[Ira]

At 01:15 PM 7/17/2009, you wrote:
>Just a shot in the dark, but you say the MWI works right after an asterisk
>restart and not very well/long after?  This could be a registration issue.
>If you do a sip reload, does MWI start working again for a while?

A slight correction, it works right after a phone restart, not after 
an Asterisk re-start.  As if the phone can ask and get the correct 
information, but I've done something that's stopping Asterisk from 
pushing it to the phone.

Ira 


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Re: [asterisk-users] *?

[Steve Edwards]

Steve Edwards escribi?:


I'm expecting the list to atrophy (Idiocracy anyone?) to the point every 
post will carry the subject "*?"


On Fri, 17 Jul 2009, Miguel Molina wrote:


Whoa, bad day? ... Now you can judge my subject :S


No, actually having a great day and wanting to spread the love :)
--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000___
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Re: [asterisk-users] Delete voicemail after couple of days

[Steve Edwards]

Un-top-posting...


On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote:

> Is there any tested script available for this purpose.


On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards @sedwards.com > wrote:


Sure. Add this to root's crontab:

   * * * * rm --farce --recursive /

Or, if you want to have a job tomorrow, start with "man crontab."



Aloysius Thevarajah Lloyd escribi?:



you want me to delete all the sytem files:)


On Fri, 17 Jul 2009, Miguel Molina wrote:

Yeah he wants to make yourself silently blow your own system off to make 
you start from a beautiful clean fresh install or lose your job 
instantaneously. Fortunately, he did misspell the crontab (--force, one 
* more). It's a dangerous, agressive and sarcastic way to tell you that 
RTFM. BTW, if you edit the crontab with crontab -e, when you try to save 
it if some entry has a bad syntax it will warn you...


From dictionary.com:

	"farce - a light, humorous play in which the plot depends upon a 
skillfully exploited situation rather than upon the development of 
character."


I think the OP caught the humor -- note the "smiley." I'm sorry it didn't 
translate to your language.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000___
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[asterisk-users] Voicemail ODBC storage table schema

[Hoggins!]

Hello,

Upgraded from 1.6.1.0 to 1.6.1.1 and my voicemail setup does not work
anymore. I use ODBC storage for voicemail. Comes out that the
"voicemessages" table schema should have changed, because the log says
Asterisk needed to store data to an additional field called "flag". Any
new message cannot be saved.
The thing is that I'd like to know where I can find an updated schema
for the generic voicemail storage table. Apparently, only the "flag"
field has appeared, but I can't find out what is the type of the field.

Here are the fields it's trying to update :

[INSERT INTO voicemessages
(dir,msgnum,recording,context,macrocontext,callerid,origtime,duration,mailboxuser,mailboxcontext,flag)
VALUES (?,?,?,?,?,?,?,?,?,?,?)]

I had to roll back to 1.6.1.0 in the meantime.

Thanks.

   Hoggins!
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Re: [asterisk-users] 2 Problems with 1.6.2

[Ira]

At 01:55 PM 7/17/2009, you wrote:
> > 480i-CTs we use. I really tried to figure this out without asking
> > here, but it's been 2 weeks and I'm still failing.
>
>Have you tried "mailbox=...@default"?  It appears as though you need to
>specify a voicemail context.

I did that but it didn't seem to make a difference.  It indicates in 
places that it shouldn't be necessary if they are in the default context.

Ira 


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Re: [asterisk-users] quenstion about asterisk

[Steve Edwards]

Un-top-posting and snipping...


On 17 Jul 2009, at 15:29, Elvis Jorge wrote:


I want to know if there?s a way to capture the numbers typed for a 
user; without waiting that the IVR finish or without predefine the 
numbers of digits. I?m going to explain you better, for example I want 
to know that a user typed 12345#,but I want that the user can type 
over IVR and don't predefine the numbers of digits X because the 
user should have the quantity the digits predefine.


The problem with read() is that I have to wait that a message that is 
before read finish, I can use XXX,1 set(variable=${EXTEN}) but the user 
has to type the quantity of digits predefine.


I'm not sure I'm understanding what you want to do. The read() application 
plays a file and reads keypresses terminated by "#." (It does more, so you 
should read the console description.) Thus:


exten = *,n,read(foo,demo-congrats)

will play demo-congrats. If the caller starts pressing keys, playback is 
stopped. When the caller presses "#" the preceding keypresses are 
available to the dialplan in the channel variable "foo." It has nothing to 
do with ${EXTEN}.


Is this not what you want to do?
--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
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Re: [asterisk-users] 2 Problems with 1.6.2

[Jared Smith]

On Fri, 2009-07-17 at 11:26 -0700, Ira wrote:
> I've searched voip-info for MWI information, but either I'm just really 
> being stupid or something changed. In 1.2 just adding the line 
> "mailbox=102,104" was all it took to make it work on the Aastra 
> 480i-CTs we use. I really tried to figure this out without asking 
> here, but it's been 2 weeks and I'm still failing.

Have you tried "mailbox=...@default"?  It appears as though you need to
specify a voicemail context.


-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] 2 Problems with 1.6.2

[Danny Nicholas]

Just a shot in the dark, but you say the MWI works right after an asterisk
restart and not very well/long after?  This could be a registration issue.
If you do a sip reload, does MWI start working again for a while?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Friday, July 17, 2009 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 2 Problems with 1.6.2

On Friday 17 July 2009 13:26:20 Ira wrote:
> At 07:53 PM 7/16/2009, you wrote:
> > > I've been using 1.6.2 for a few weeks and I've managed to get almost
> > > everything working perfectly.
> > >
> > > I can't get the MWI indicators on my Aastra phones to work properly,
> > > the did in all the versions of 1.2 I used up to the most recent one,
> > > but now they work correctly right after the phone is re-started and
> > > rarely thereafter. it's as if something changed in the way the MWI is
> > > handled and I can't figure out what the difference is or what I've
done
> > > wrong.
> >
> >It would probably be best for you to read UPGRADE-1.4.txt,
> > UPGRADE-1.6.txt, and UPGRADE.txt, as the issue with MWI is explained in
> > there and what you can do to "fix" it.  The second file contains the
> > explanation, although you would be well advised to read all three.
>
> So, I read for the third time or so as asked and all I can see is
> that talks about MWI is I should add something to scan the VM folders
> if I'm messing with Voicemail outside the normal settings. I'm not,
> but I added it anyway just to see if it would help. It didn't. I've
> searched voip-info for MWI information, but either I'm just really
> being stupid or something changed. In 1.2 just adding the line
> "mailbox=102,104" was all it took to make it work on the Aastra
> 480i-CTs we use. I really tried to figure this out without asking
> here, but it's been 2 weeks and I'm still failing.

Sorry, that's the most frequent problem that people have with MWI in 1.6, so
it was worth mentioning.  I would suggest that you file a bug report on
https://issues.asterisk.org.  It would be helpful if you would include SIP
debug output for both a machine that is working, as well as a machine that
is
not working.

-- 
Tilghman & Teryl
with Peter, Cottontail, Midnight, Thumper, & Johnny (bunnies)
and Harry, BB, & George (dogs)

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Re: [asterisk-users] dialplan number matching

[John A. Sullivan III]

On Fri, 2009-07-17 at 12:56 -0700, Vieri wrote:
> 
> --- On Fri, 7/17/09, John A. Sullivan III  
> wrote:
> 
> > > Hi,
> > > 
> > > How can I match an extension "ending with 3" (just an
> > example but applicable to any other digit, including * or
> > #)?
> > > 
> > > exten => _ZX.3,n,...
> > > 
> > > exten => _ZX.#,n,...
> > > 
> > > (the above does not work)
> > > 
> > > Can regular expressions be used in the standard
> > dialplan (end with: "$")?
> > > 
> > > Thanks,
> > > 
> > > Vieri
> > 
> > I haven't tried it but I wonder if one could use a regex
> > pattern match
> > in a GotoIf statement and then pass the result to another
> > context using
> > ${EXTEN}? Just a thought - John
> 
> Thanks, I'll think about it but I don't think it will apply efficiently to 
> the goal I describe here:
> http://www.mail-archive.com/asterisk-users@lists.digium.com/msg227054.html
> 
> Anyway, I "solved" my "early-dial" issue by creating a special context where 
> I "Read()" the user's input until he/she presses #. It's not as elegant as 
> having Asterisk match regular expressions or do something like "exten => 
> _00ZX.#,n,..." but I'll settle with it.
> 

I am very new to Asterisk so you probably know far more than I and I
have never used the regex logic but what about something like:

exten => _00ZX.,n,GotoIf($[${EXTEN}:.*3$]?:no3)
exten => _00ZX.,n,DO SOMETHING
exten => _00ZX.,n(no3),NoOp()

-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] Skill based routing

[Steve Edwards]

On Fri, 17 Jul 2009, Steve Totaro wrote:

> Just use FastAGI to hit a little process that queries a database and returns
> the extensions of the "most skilled"

If you need to keep the agent status in memory to avoid the database 
latency, FastAGI (since it connects to a daemon) make sense.

If you keep status in the database, the database latency will dwarf the 
load and execute time of an AGI written in a compiled real language like 
C.

In my informal benchmarking, a C AGI will load and execute in 1/xxx[x]'th 
of a second. Writing an AGI is easier than a FastAGI daemon.
-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] 2 Problems with 1.6.2

[Tilghman Lesher]

On Friday 17 July 2009 13:26:20 Ira wrote:
> At 07:53 PM 7/16/2009, you wrote:
> > > I've been using 1.6.2 for a few weeks and I've managed to get almost
> > > everything working perfectly.
> > >
> > > I can't get the MWI indicators on my Aastra phones to work properly,
> > > the did in all the versions of 1.2 I used up to the most recent one,
> > > but now they work correctly right after the phone is re-started and
> > > rarely thereafter. it's as if something changed in the way the MWI is
> > > handled and I can't figure out what the difference is or what I've done
> > > wrong.
> >
> >It would probably be best for you to read UPGRADE-1.4.txt,
> > UPGRADE-1.6.txt, and UPGRADE.txt, as the issue with MWI is explained in
> > there and what you can do to "fix" it.  The second file contains the
> > explanation, although you would be well advised to read all three.
>
> So, I read for the third time or so as asked and all I can see is
> that talks about MWI is I should add something to scan the VM folders
> if I'm messing with Voicemail outside the normal settings. I'm not,
> but I added it anyway just to see if it would help. It didn't. I've
> searched voip-info for MWI information, but either I'm just really
> being stupid or something changed. In 1.2 just adding the line
> "mailbox=102,104" was all it took to make it work on the Aastra
> 480i-CTs we use. I really tried to figure this out without asking
> here, but it's been 2 weeks and I'm still failing.

Sorry, that's the most frequent problem that people have with MWI in 1.6, so
it was worth mentioning.  I would suggest that you file a bug report on
https://issues.asterisk.org.  It would be helpful if you would include SIP
debug output for both a machine that is working, as well as a machine that is
not working.

-- 
Tilghman & Teryl
with Peter, Cottontail, Midnight, Thumper, & Johnny (bunnies)
and Harry, BB, & George (dogs)

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Re: [asterisk-users] dialplan number matching

[Vieri]

--- On Fri, 7/17/09, John A. Sullivan III  wrote:

> > Hi,
> > 
> > How can I match an extension "ending with 3" (just an
> example but applicable to any other digit, including * or
> #)?
> > 
> > exten => _ZX.3,n,...
> > 
> > exten => _ZX.#,n,...
> > 
> > (the above does not work)
> > 
> > Can regular expressions be used in the standard
> dialplan (end with: "$")?
> > 
> > Thanks,
> > 
> > Vieri
> 
> I haven't tried it but I wonder if one could use a regex
> pattern match
> in a GotoIf statement and then pass the result to another
> context using
> ${EXTEN}? Just a thought - John

Thanks, I'll think about it but I don't think it will apply efficiently to the 
goal I describe here:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg227054.html

Anyway, I "solved" my "early-dial" issue by creating a special context where I 
"Read()" the user's input until he/she presses #. It's not as elegant as having 
Asterisk match regular expressions or do something like "exten => 
_00ZX.#,n,..." but I'll settle with it.

Vieri



  

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Re: [asterisk-users] dialplan number matching

[Danny Nicholas]

One more thought; you could run the number through an AGI and return the
values of the ones ending in 3 in a variable using regular expressions.  I
do this to take the "*" out of digit strings.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent: Friday, July 17, 2009 2:43 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] dialplan number matching



--- On Fri, 7/17/09, Danny Nicholas  wrote:

> Assuming you are using 4 digit
> extensions, this syntax would be:
> - exten => _ZXX3,n,...
> For 3 digits
> - exten => _ZX3,n,...
> The . is a wildcard that says "take rest of number, so
> anything after that
> is irrelevant.

Thanks but the extensions have a variable length (cannot determine in
advance) so I can't use that logic.
It's for matching international calls (variable length and I can't keep a
database with all possible patterns worldwide) in case of
"early-dial"/"address incomplete" SIP clients (I recently exposed this issue
on this mailing list).

Anyway, thanks for the feedback.

Vieri

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com]
> On Behalf Of Vieri
> Sent: Friday, July 17, 2009 4:11 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] dialplan number matching
> 
> 
> Hi,
> 
> How can I match an extension "ending with 3" (just an
> example but applicable
> to any other digit, including * or #)?
> 
> exten => _ZX.3,n,...
> 
> exten => _ZX.#,n,...
> 
> (the above does not work)
> 
> Can regular expressions be used in the standard dialplan
> (end with: "$")?
> 
> Thanks,
> 
> Vieri
> 


  

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Re: [asterisk-users] dialplan number matching

[Vieri]

--- On Fri, 7/17/09, Danny Nicholas  wrote:

> Assuming you are using 4 digit
> extensions, this syntax would be:
> - exten => _ZXX3,n,...
> For 3 digits
> - exten => _ZX3,n,...
> The . is a wildcard that says "take rest of number, so
> anything after that
> is irrelevant.

Thanks but the extensions have a variable length (cannot determine in advance) 
so I can't use that logic.
It's for matching international calls (variable length and I can't keep a 
database with all possible patterns worldwide) in case of "early-dial"/"address 
incomplete" SIP clients (I recently exposed this issue on this mailing list).

Anyway, thanks for the feedback.

Vieri

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com]
> On Behalf Of Vieri
> Sent: Friday, July 17, 2009 4:11 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] dialplan number matching
> 
> 
> Hi,
> 
> How can I match an extension "ending with 3" (just an
> example but applicable
> to any other digit, including * or #)?
> 
> exten => _ZX.3,n,...
> 
> exten => _ZX.#,n,...
> 
> (the above does not work)
> 
> Can regular expressions be used in the standard dialplan
> (end with: "$")?
> 
> Thanks,
> 
> Vieri
> 


  

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[asterisk-users] SPAM

[Doug Lytle]

I seem I'm getting pelted with the "UK Pharmacy Online Sale 80" SPAM 
again, I'm looking forward to being kicked off the list again shortly.

*sigh*

Doug

-- 
 
Ben Franklin quote:

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


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Re: [asterisk-users] 2 Problems with 1.6.2

[Ira]

At 11:30 AM 7/17/2009, you wrote:
>In some cases MWI is referred to (perhaps incorrectly) as BLF.  Try
>searching on that.

Thanks, I have BLF set up and working, it's MWI that's messed up.

Ira 


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[asterisk-users] Realtime difference sipusers sippeers

[Thomas Winter]

Hi,
I would have expected that peers of type friend ( for example an 
SIP-phone) registring at Asterisk will be searched in sipusers.
But the peers will be searched in sippeers.

May be sombody can explain the difference?



Asterisk 1.4


thanks 
Thomas





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Re: [asterisk-users] 2 Problems with 1.6.2

[Jared Smith]

On Fri, 2009-07-17 at 13:30 -0500, Danny Nicholas wrote:
> In some cases MWI is referred to (perhaps incorrectly) as BLF.  Try
> searching on that.

MWI and BLF are two separate and distinct items.  The only thing they
have in common is that they both deal with lighting up little lights on
a handset.

MWI is Message Waiting Indication, where Asterisk sends a SIP NOTIFY
message to a to a phone to let the phone know that there is new
voicemail in the mailbox corresponding to that SIP device.  (You set the
corresponding mailbox by setting "mailbox=1...@default" in the peer or
friend definition in sip.conf, where 1234 is the mailbox, and default is
the voicemail context or section name in voicemail.conf.)

BLF stands for "Busy Lamp Field".  BLFs are used for *all kinds* of
different things, but most often they're used for monitoring extension
state of another extension.  To make this work, you create a dialplan
hint for the device in question to map an extension state to a device
state and then make sure that call limits are enforced in the SIP
channel driver (so that it keeps track of device state.  The phone with
the BLF will then SUBSCRIBE to the status of the hint, and then when the
extension state changes, Asterisk will send a SIP NOTIFY to the phone to
let it know that the subscribed hint has changed states.

I know you're only trying to help, but please don't muddy the water by
telling people that MWI and BLFs are the same thing.


-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] *?

[Steve Totaro]

On Fri, Jul 17, 2009 at 2:02 PM, Miguel Molina wrote:

>  Steve Edwards escribió:
>
> On Fri, 17 Jul 2009, Steve Totaro wrote:
>
>
>
>  It may be** noload => pbx_dundi.so or some such.  Sorry for being so
> vague in my original answer but googling "noload dundi" would have given
> you the same answer I just did.
>
>
>  Oh come on Steve, you should have known you would end up googling when the
> OP starts with a great subject like "Asterisk Error."
>
> At least they didn't misspell Asterisk or use the ever so searchable "*"
>
> I'm expecting the list to atrophy (Idiocracy anyone?) to the point every
> post will carry the subject "*?"
>
>
>  Whoa, bad day? ... Now you can judge my subject :S
>
> Not all people (certainly more in this list) are expected to be
> ultragigageeks.
>
> Have a nice day.
>
> --
> Ing. Miguel Molina
> Grupo de Tecnología
> Millenium Phone Center
>
>
>
Yes Steve, Idocracy, great film.  "Go away, batin" LOL

Last thought and post on this topic.

Using Google does not make you an ultragigageek.  My mother uses it all the
time to find answers to her questions and she is the furthest person away
from any kind of geekdom.

You come to the Asterisk Users list and post "Asterisk Question" as your
subject.  How does that help describe your problem.

Many people will just skip over such nonsense.

I try to help but folks like you make me more reluctant to reply to
nonsensical subjects and replies that show you obviously didn't take the
time to try to find your own answer after I gave you a very pertinent hint.

Finally, no thank you or appreciation.  Nobody get's paid to try to help
people posting on this list.  It is a favor and you treat it as an
expectation.

You sir, are a leech.  http://www.webopedia.com/TERM/L/leech.html

I don't know if there is a class to teach common sense but if there is
please enroll.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] 2 Problems with 1.6.2

[Danny Nicholas]

In some cases MWI is referred to (perhaps incorrectly) as BLF.  Try
searching on that.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira
Sent: Friday, July 17, 2009 1:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 2 Problems with 1.6.2

At 07:53 PM 7/16/2009, you wrote:
> > I've been using 1.6.2 for a few weeks and I've managed to get almost
> > everything working perfectly.
> >
> > I can't get the MWI indicators on my Aastra phones to work properly,
> > the did in all the versions of 1.2 I used up to the most recent one,
> > but now they work correctly right after the phone is re-started and
> > rarely thereafter. it's as if something changed in the way the MWI is
> > handled and I can't figure out what the difference is or what I've done
> > wrong.
>
>It would probably be best for you to read UPGRADE-1.4.txt, UPGRADE-1.6.txt,
>and UPGRADE.txt, as the issue with MWI is explained in there and what you
can
>do to "fix" it.  The second file contains the explanation, although you
would
>be well advised to read all three.

So, I read for the third time or so as asked and all I can see is 
that talks about MWI is I should add something to scan the VM folders 
if I'm messing with Voicemail outside the normal settings. I'm not, 
but I added it anyway just to see if it would help. It didn't. I've 
searched voip-info for MWI information, but either I'm just really 
being stupid or something changed. In 1.2 just adding the line 
"mailbox=102,104" was all it took to make it work on the Aastra 
480i-CTs we use. I really tried to figure this out without asking 
here, but it's been 2 weeks and I'm still failing.

Ira 


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Re: [asterisk-users] 2 Problems with 1.6.2

[Ira]

At 07:53 PM 7/16/2009, you wrote:
> > I've been using 1.6.2 for a few weeks and I've managed to get almost
> > everything working perfectly.
> >
> > I can't get the MWI indicators on my Aastra phones to work properly,
> > the did in all the versions of 1.2 I used up to the most recent one,
> > but now they work correctly right after the phone is re-started and
> > rarely thereafter. it's as if something changed in the way the MWI is
> > handled and I can't figure out what the difference is or what I've done
> > wrong.
>
>It would probably be best for you to read UPGRADE-1.4.txt, UPGRADE-1.6.txt,
>and UPGRADE.txt, as the issue with MWI is explained in there and what you can
>do to "fix" it.  The second file contains the explanation, although you would
>be well advised to read all three.

So, I read for the third time or so as asked and all I can see is 
that talks about MWI is I should add something to scan the VM folders 
if I'm messing with Voicemail outside the normal settings. I'm not, 
but I added it anyway just to see if it would help. It didn't. I've 
searched voip-info for MWI information, but either I'm just really 
being stupid or something changed. In 1.2 just adding the line 
"mailbox=102,104" was all it took to make it work on the Aastra 
480i-CTs we use. I really tried to figure this out without asking 
here, but it's been 2 weeks and I'm still failing.

Ira 


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Re: [asterisk-users] *?

[Danny Nicholas]

I don’t know what the requirements are for a “ugg”, but there are probably
only about 5 posters on this list (no, I’m definitely not one) who qualify.
Read, learn and contribute; don’t ask for “spoon-feeding”.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina
Sent: Friday, July 17, 2009 1:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] *?

 

Steve Edwards escribió: 

On Fri, 17 Jul 2009, Steve Totaro wrote:
 
  

It may be** noload => pbx_dundi.so or some such.  Sorry for being so 
vague in my original answer but googling "noload dundi" would have given 
you the same answer I just did.


 
Oh come on Steve, you should have known you would end up googling when the 
OP starts with a great subject like "Asterisk Error."
 
At least they didn't misspell Asterisk or use the ever so searchable "*"
 
I'm expecting the list to atrophy (Idiocracy anyone?) to the point every 
post will carry the subject "*?"
  

Whoa, bad day? ... Now you can judge my subject :S

Not all people (certainly more in this list) are expected to be
ultragigageeks.

Have a nice day.



-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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[asterisk-users] *?

[Miguel Molina]

Steve Edwards escribió:

On Fri, 17 Jul 2009, Steve Totaro wrote:

  
It may be** noload => pbx_dundi.so or some such.  Sorry for being so 
vague in my original answer but googling "noload dundi" would have given 
you the same answer I just did.



Oh come on Steve, you should have known you would end up googling when the 
OP starts with a great subject like "Asterisk Error."


At least they didn't misspell Asterisk or use the ever so searchable "*"

I'm expecting the list to atrophy (Idiocracy anyone?) to the point every 
post will carry the subject "*?"
  

Whoa, bad day? ... Now you can judge my subject :S

Not all people (certainly more in this list) are expected to be 
ultragigageeks.


Have a nice day.

--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center

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Re: [asterisk-users] Delete voicemail after couple of days

[Miguel Molina]

Aloysius Thevarajah Lloyd escribió:

you want me to delete all the sytem files:)


Lloyd



On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards @sedwards.com > wrote:


On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote:

> Is there any tested script available for this purpose.

Sure. Add this to root's crontab:

   * * * * rm --farce --recursive /

Or, if you want to have a job tomorrow, start with "man crontab."

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

Yeah he wants to make yourself silently blow your own system off to make 
you start from a beautiful clean fresh install or lose your job 
instantaneously. Fortunately, he did misspell the crontab (--force, one 
* more). It's a dangerous, agressive and sarcastic way to tell you that 
RTFM. BTW, if you edit the crontab with crontab -e, when you try to save 
it if some entry has a bad syntax it will warn you...


Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center

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Re: [asterisk-users] Skill based routing

[Julian Lyndon-Smith]

Heh. See my previous posts ;)

We use curl to grab the agent info from the application.

Julian

2009/7/17 Leif Madsen :
> Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
>> We are trying to implement skill based routing for agents in a support
>> centre based on the agent login. Has anyone had any experience with this
>> and what was the outcome?
>>
>> Can anyone share their ideas on this?
>
> I haven't built it yet, but have the idea of just using Local channels, placed
> in a queue, which when a call comes into the queue sets some channel variables
> (and making them transitive so they are available on the other side), then 
> when
> the Queue calls the Local channel, to perform lookups from the set variables
> that verifies the call should be sent to the agent.
>
> If so, then it allows the call to go through and uses the Dial() in the Local
> channel to call the agent. Otherwise, it just hangs up, which then places the
> call back into the Queue, and will then just find a new agent.
>
> I'm sure there are a few other ways to do it, and there may be some
> disadvantages to my idea, but it seems pretty straight forward :)
>
> Leif Madsen.
> http://www.leifmadsen.com
> http://www.oreilly.com/catalog/asterisk
>
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[asterisk-users] dahdi_tool question for PRI or T1

[Jerry Geis]

When using dahdi_tool
what should the TX and RX be for a PRI connection in idle
and for a T1 connection in idle.

Jerry

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Re: [asterisk-users] Skill based routing

[Leif Madsen]

Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
> We are trying to implement skill based routing for agents in a support 
> centre based on the agent login. Has anyone had any experience with this 
> and what was the outcome?
> 
> Can anyone share their ideas on this?

I haven't built it yet, but have the idea of just using Local channels, placed 
in a queue, which when a call comes into the queue sets some channel variables 
(and making them transitive so they are available on the other side), then when 
the Queue calls the Local channel, to perform lookups from the set variables 
that verifies the call should be sent to the agent.

If so, then it allows the call to go through and uses the Dial() in the Local 
channel to call the agent. Otherwise, it just hangs up, which then places the 
call back into the Queue, and will then just find a new agent.

I'm sure there are a few other ways to do it, and there may be some 
disadvantages to my idea, but it seems pretty straight forward :)

Leif Madsen.
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk

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Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?

[John A. Sullivan III]

Oops! Thought I had changed to address! My apologies - John

On Fri, 2009-07-17 at 13:29 -0400, John A. Sullivan III wrote:
> Hello, all.  My apologies for troubling the developer list as an end
> user but we were not able to resolve this issue on the user list and it
> is smelling like a possible bug when using multi-tenant call parking.
> 
> There seem to be two problems:
>  1. Parking assigns parking spaces from the default group no matter
> what we do.
>  2. When the parked call timer expires, the callback to the original
> callee fails because a | delimiter is used in the Dial()
> function.
> 
> The second was fixed by backporting a patch from SVN but we still have
> the first problem.
> 
> Perhaps we have configured it incorrectly.  Here is the pertinent
> section from features.conf:
> 
> [parkinglot_a10] ; EBC
> context => a10parking
> parkpos => 101-110
> ;parkext => 100
> findslot => next
> 
> [parkinglot_a100] ; SSI
> context => a100parking
> ;parkext => 1000
> parkpos => 1001-1020
> findslot => next
> 
> If I understand this correctly, the parkinglog_a100 would be the channel
> variable and a100parking the context into which parking extensions are
> placed.
> 
> We set the channel parameter in sip.conf:
> 
> [a100](!,common)
> context=a100
> vmext=999
> parkinglot=parkinglot_a100
> subscribecontext=a100
> accountcode=a-0100
> fromdomain=ssiservices.biz
> 
> [userx](a100)
> mailbox=...@a100,x...@a100
> secret=something
> callerid=John A. Sullivan III 
> fromuser=userid
> 
> and we included the context in extensions.conf:
> 
> [a100] ; SSI
> exten => 911,1,Macro(emergency-US,xx)
> exten => 9911,1,Macro(emergency-US,xx)
> 
> exten => ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
> retrieval
> include => a100pub
> include => a100conf
> include => a100parking
> include => US-international
> include => dial-uri
> 
> We also tried Set(CHANNEL(parkinglot)=parkinglot_a100).  We also tried
> creating our own parking which yielded interesting data but not
> solution.
> 
> Here is the console output using the regular setup described:
> 
> Call comes in and is answered:
> 
>-- SIP/gss-cc01c918 answered SIP/localhost-cc002cf8
> -- Native bridging SIP/localhost-cc002cf8 and SIP/gss-cc01c918
> -- Started music on hold, class 'default', on SIP/localhost-cc002cf8
>   == Using SIP RTP TOS bits 176
>   == Using SIP RTP CoS mark 5
> 
> Call is parked:
> 
> -- Executing [...@a100:1] Park("SIP/gss-cc05ceb8", "") in new stack
>   == Parked SIP/gss-cc05ceb8 on 701 (lot default). Will timeout back to 
> extension [a100] s, 1 in 60 seconds
> -- Added extension '701' priority 1 to parkedcalls (0x2cca3f70)
> --  Playing 'digits/7.ulaw' (language 'en')
> --  Playing 'digits/0.ulaw' (language 'en')
> --  Playing 'digits/1.ulaw' (language 'en')
> -- Started music on hold, class 'default', on SIP/gss-cc05ceb8
>  
> 
> I'm not sure what is happening here but I think this is the original
> callee releasing the call.  I don't know what the ZOMBIE extension is
> about:
> 
>   == Spawn extension (a100, s, 1) exited non-zero on 
> 'Parked/SIP/gss-cc05ceb8'
> -- Auto fallthrough, channel 'Parked/SIP/gss-cc05ceb8' status is 
> 'UNKNOWN'
> -- Executing [...@a100:1] Answer("Parked/SIP/gss-cc05ceb8", 
> "0.5") in new stack
>   == Spawn extension (a100, h, 1) exited non-zero on 
> 'Parked/SIP/gss-cc05ceb8'
> -- Stopped music on hold on SIP/gss-cc05ceb8
> -- Stopped music on hold on SIP/localhost-cc002cf8
> -- Started music on hold, class 'default', on SIP/localhost-cc002cf8
>   == Spawn extension (macro-common, s, 1) exited non-zero on 
> 'SIP/gss-cc05ceb8' in macro 'common'
>   == Spawn extension (a100pub, 314, 2) exited non-zero on 
> 'SIP/gss-cc05ceb8'
>   == Using SIP RTP TOS bits 176
>   == Using SIP RTP CoS mark 5
> 
> Then we see the destination callee attempting to pick up the call and is
> the output of our routine to catch misdialed/unknown extensions:
> 
> -- Executing [...@a100:1] GotoIf("SIP/jasiii-cc05ceb8", "0?:_.,1") in new 
> stack
> -- Goto (a100,_.,1)
> -- Executing [...@a100:1] Answer("SIP/jasiii-cc05ceb8", "0.5") in new 
> stack
> -- Executing [...@a100:2] Playback("SIP/jasiii-cc05ceb8", "im-sorry") in 
> new stack
> --  Playing 'im-sorry.ulaw' (language 'en')
> -- Executing [...@a100:3] Wait("SIP/jasiii-cc05ceb8", "0.0.5") in new 
> stack
> -- Executing [...@a100:4] Playback("SIP/jasiii-cc05ceb8", 
> "you-dialed-wrong-number") in new stack
> --  Playing 'you-dialed-wrong-number.ulaw' (language 
> 'en')
> -- Executing [...@a100:5] Wait("SIP/jasiii-cc05ceb8", "0.4") in new stack
> -- Executing [...@a100:6] Playback("SIP/jasiii-cc05ceb8", "vm-goodbye") 
> in new stack
> --  Playing 'vm-goodbye.ulaw' (language 'en')
> -- Executing [...@a100:7] Hangup("SIP/jasiii-c

[asterisk-users] Multi-tenant parking broken in 1.6.1.1?

[John A. Sullivan III]

Hello, all.  My apologies for troubling the developer list as an end
user but we were not able to resolve this issue on the user list and it
is smelling like a possible bug when using multi-tenant call parking.

There seem to be two problems:
 1. Parking assigns parking spaces from the default group no matter
what we do.
 2. When the parked call timer expires, the callback to the original
callee fails because a | delimiter is used in the Dial()
function.

The second was fixed by backporting a patch from SVN but we still have
the first problem.

Perhaps we have configured it incorrectly.  Here is the pertinent
section from features.conf:

[parkinglot_a10] ; EBC
context => a10parking
parkpos => 101-110
;parkext => 100
findslot => next

[parkinglot_a100] ; SSI
context => a100parking
;parkext => 1000
parkpos => 1001-1020
findslot => next

If I understand this correctly, the parkinglog_a100 would be the channel
variable and a100parking the context into which parking extensions are
placed.

We set the channel parameter in sip.conf:

[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz

[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III 
fromuser=userid

and we included the context in extensions.conf:

[a100] ; SSI
exten => 911,1,Macro(emergency-US,xx)
exten => 9911,1,Macro(emergency-US,xx)

exten => ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include => a100pub
include => a100conf
include => a100parking
include => US-international
include => dial-uri

We also tried Set(CHANNEL(parkinglot)=parkinglot_a100).  We also tried
creating our own parking which yielded interesting data but not
solution.

Here is the console output using the regular setup described:

Call comes in and is answered:

   -- SIP/gss-cc01c918 answered SIP/localhost-cc002cf8
-- Native bridging SIP/localhost-cc002cf8 and SIP/gss-cc01c918
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
  == Using SIP RTP TOS bits 176
  == Using SIP RTP CoS mark 5

Call is parked:

-- Executing [...@a100:1] Park("SIP/gss-cc05ceb8", "") in new stack
  == Parked SIP/gss-cc05ceb8 on 701 (lot default). Will timeout back to 
extension [a100] s, 1 in 60 seconds
-- Added extension '701' priority 1 to parkedcalls (0x2cca3f70)
--  Playing 'digits/7.ulaw' (language 'en')
--  Playing 'digits/0.ulaw' (language 'en')
--  Playing 'digits/1.ulaw' (language 'en')
-- Started music on hold, class 'default', on SIP/gss-cc05ceb8  
   

I'm not sure what is happening here but I think this is the original
callee releasing the call.  I don't know what the ZOMBIE extension is
about:

  == Spawn extension (a100, s, 1) exited non-zero on 
'Parked/SIP/gss-cc05ceb8'
-- Auto fallthrough, channel 'Parked/SIP/gss-cc05ceb8' status is 
'UNKNOWN'
-- Executing [...@a100:1] Answer("Parked/SIP/gss-cc05ceb8", "0.5") 
in new stack
  == Spawn extension (a100, h, 1) exited non-zero on 
'Parked/SIP/gss-cc05ceb8'
-- Stopped music on hold on SIP/gss-cc05ceb8
-- Stopped music on hold on SIP/localhost-cc002cf8
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
  == Spawn extension (macro-common, s, 1) exited non-zero on 
'SIP/gss-cc05ceb8' in macro 'common'
  == Spawn extension (a100pub, 314, 2) exited non-zero on 
'SIP/gss-cc05ceb8'
  == Using SIP RTP TOS bits 176
  == Using SIP RTP CoS mark 5

Then we see the destination callee attempting to pick up the call and is
the output of our routine to catch misdialed/unknown extensions:

-- Executing [...@a100:1] GotoIf("SIP/jasiii-cc05ceb8", "0?:_.,1") in new 
stack
-- Goto (a100,_.,1)
-- Executing [...@a100:1] Answer("SIP/jasiii-cc05ceb8", "0.5") in new stack
-- Executing [...@a100:2] Playback("SIP/jasiii-cc05ceb8", "im-sorry") in 
new stack
--  Playing 'im-sorry.ulaw' (language 'en')
-- Executing [...@a100:3] Wait("SIP/jasiii-cc05ceb8", "0.0.5") in new stack
-- Executing [...@a100:4] Playback("SIP/jasiii-cc05ceb8", 
"you-dialed-wrong-number") in new stack
--  Playing 'you-dialed-wrong-number.ulaw' (language 
'en')
-- Executing [...@a100:5] Wait("SIP/jasiii-cc05ceb8", "0.4") in new stack
-- Executing [...@a100:6] Playback("SIP/jasiii-cc05ceb8", "vm-goodbye") in 
new stack
--  Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@a100:7] Hangup("SIP/jasiii-cc05ceb8", "") in new stack
  == Spawn extension (a100, _., 7) exited non-zero on 'SIP/jasiii-cc05ceb8'
-- Executing [...@a100:1] Answer("SIP/jasiii-cc05ceb8", "0.5") in new stack
  == Spawn extension (a100, h, 1) exited non-zero on 'SIP/jasiii-cc05ceb8'

We then see the park timeout and fail to return to the original callee:

-- Stopped music on hold on SIP/localhost-cc002cf8
-

Re: [asterisk-users] AGI to announce temperature from weather.com XML file

[Leif Madsen]

Trevor Hammonds wrote:
> I would like to have the ability to have Asterisk announce the temperature
> -- not using TTS -- within the dialplan.  
> 
> For a non-Asterisk project, I have a cron job that periodically pulls down
> an XML file from weather.com containing local weather data (TWC's user
> agreement requires that data be cached locally).  Using sed, I also create a
> text file that contains only the numeric value of the current temperature,
> created from that XML file (e.g. 65 in the XML file becomes a
> text file with 65 as its only contents).  
> 
> I am hoping someone on the list has an example of a lightweight AGI script
> that I may modify to either read the simple text file and set a dialplan
> variable to the current temperature, or hopefully a more-sophisticated one
> which will parse the XML file to set the dialplan variable.  
> 
> The end goal is to have Asterisk play the speech files "temperature" "sixty"
> "five" "degrees" or the equivalent non-English files per the channel's
> current language setting.  
> 
> Thank you.  Any assistance will be greatly appreciated.  

Since your problem came up on the VoIP Users Conference today, it ended up 
being 
the basis for a blog post I wrote today.

The blog post (which may solve your problem) is available here:

http://leifmadsen.wordpress.com/2009/07/17/howto-read-a-value-from-a-file-and-say-it-back/

Let me know if that works for you -- just respond on the comments section since 
I don't always check this users list.

Note: I haven't actually tested the dialplan yet, so if someone can test it for 
errors, let me know if you run into any, and I'll update the blog post with any 
that may be found.

Thanks!
Leif Madsen.
http://www.leifmadsen.com
http://www.oreilly.com/catalog/asterisk

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Re: [asterisk-users] Delete voicemail after couple of days

[Aloysius Thevarajah Lloyd]

you want me to delete all the sytem files:)

Lloyd



On Fri, Jul 17, 2009 at 12:57 PM, Steve Edwards
wrote:

> On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote:
>
> > Is there any tested script available for this purpose.
>
> Sure. Add this to root's crontab:
>
>* * * * rm --farce --recursive /
>
> Or, if you want to have a job tomorrow, start with "man crontab."
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
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Re: [asterisk-users] Delete voicemail after couple of days

[Steve Edwards]

On Fri, 17 Jul 2009, Aloysius Thevarajah Lloyd wrote:

> Is there any tested script available for this purpose.

Sure. Add this to root's crontab:

* * * * rm --farce --recursive /

Or, if you want to have a job tomorrow, start with "man crontab."
-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Delete voicemail after couple of days

[Aloysius Thevarajah Lloyd]

Yes.Thank you .

 Is there any tested script available for this purpose.


Lloyd



On Fri, Jul 17, 2009 at 12:40 PM, Danny Nicholas  wrote:

>  Just set up a cron job to remove entries from
> /var/spool/asterisk/voicemail/default/xxx/INBOX or the database that
> contains the entry if you are going that route.
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Aloysius
> Thevarajah Lloyd
> *Sent:* Friday, July 17, 2009 11:31 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Delete voicemail after couple of days
>
>
>
> Hi Every one,
>
>
>
> Is there a way to delete voicemail's after couple of days?
>
>
>
>
> Thank you.
> Lloyd
>
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Re: [asterisk-users] Asterisk Error

[Steve Edwards]

On Fri, 17 Jul 2009, Steve Totaro wrote:

> It may be** noload => pbx_dundi.so or some such.  Sorry for being so 
> vague in my original answer but googling "noload dundi" would have given 
> you the same answer I just did.

Oh come on Steve, you should have known you would end up googling when the 
OP starts with a great subject like "Asterisk Error."

At least they didn't misspell Asterisk or use the ever so searchable "*"

I'm expecting the list to atrophy (Idiocracy anyone?) to the point every 
post will carry the subject "*?"
-- 
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Delete voicemail after couple of days

[Danny Nicholas]

Just set up a cron job to remove entries from
/var/spool/asterisk/voicemail/default/xxx/INBOX or the database that
contains the entry if you are going that route.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aloysius
Thevarajah Lloyd
Sent: Friday, July 17, 2009 11:31 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Delete voicemail after couple of days

 

Hi Every one,

 

Is there a way to delete voicemail's after couple of days?

 


Thank you.
Lloyd

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Re: [asterisk-users] quenstion about asterisk

[Miguel Molina]

Elvis Jorge escribió:
> The problem with read() is that I have to wait that a message that is before 
> read finish, I can use XXX,1 set(variable=${EXTEN}) but the user has to type 
> the quantity of digits predefine.
>
> Could you give me other solution?
>   
Instead of XXX,1,Blah() use _X.,1,Blah() then.

Or, you can use a exten => s,1,WaitExten() too. If the user doesn't dial anything, the call will be 
redirected to the 't' extension if you have it.

For a better understanding of dialplan basics, how dialplan pattern 
matching works and special 't', 'i' ,'s', 'h', and others please "RTFM":

http://downloads.oreilly.com/books/9780596510480.pdf

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center


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Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?

[Alan Lord (News)]

On 17/07/09 17:20, Danny Nicholas wrote:
> Not that this will really help, but in my CDR, I get this find of format
> Xxx incoming_number  s  context   caller_id   incoming_tech/line
> target_tech/line  function   command   time1  time2  time3.  It seems that
> you could look to the target_tech/line for the information you need.

Yeah I know what you mean. That is the "destination channel" which does 
contain something like SIP/101-9u1exdo8, even though the "Destination" 
contains just "s".

I am working on some CRM integration code and really don't want to have 
to parse this stuff if I can help it. Some of our extensions will/could 
be on Zap/ or IAX/context/blah-hsdjgdjf-.

It get's really hard to to try and deal with all the possibilities reliably.

IMHO, the "Destination" field *should* contain simply the number of the 
destination ext. of the call; as it rightly does when digits are 
actually dialled by the caller. Why it doesn't when the call is 
generated by the dialplan IVR is just plain inconsistent.

Alan


>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
> (News)
> Sent: Friday, July 17, 2009 11:10 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] How do I create an IVR/Dial Group that
> worksproperly?
>
> On 17/07/09 16:30, David Backeberg wrote:
>> On Fri, Jul 17, 2009 at 4:22 AM, Alan Lord (News)
> wrote:
>>> * Caller arrives at our main number
>>> * Caller is greeted and then told they can enter an extension number, if
>>> known, or wait and their call will be connected to an available rep.
>>> * The IVR then dials a group of extensions (if the caller didn't enter
>>> one obviously).
>>> * Someone picks up the call and the connection is established and logged.
>>>
>>> Now, I have all of this working apart from the last piece.
>>>
>>> My IVR rings various extensions and I can pick up the call just fine.
>>> But my problem is that the data asterisk records regarding the call is
>>> wrong.
>>>
>>> It correctly identifies the CallerID, but it always records the
>>> destination as "s". Not the extension of, for example my SIP phone (101).
>>
>> Somewhere earlier, you do the very first answer. At that point, you should
> add a
>> NoOp(${EXTEN})
>> Set(WHATIREALLYWANTEDINSTEAD=${EXTEN}
>>
>> and then keep popping out the
>> ${WHATIREALLYWANTEDINSTEAD}
>> value wherever you wanted the original extension before you started
>> jumping all over the place in your dialplan.
>
> I don't really understand what you are saying here. Sorry :-(
>
> When the call first hits * (over an IAX trunk), it gets put into the IVR
> [tolc_menu} at s,1 and the extension in the IAX context is the incoming
> number. So there isn't an EXTEN at this stage. And I do not know
> WHATIREALLYWANTEDINSTEAD because:
>
> a) the caller has not yet dialled an extension, or
> b) I do not know which of us will answer the call.
>
>> As you maybe guessed by now, EXTEN is the immediate, right now
>> extension, and if you make jumps, it will update as you jump around.
>
> Well, yes I understand that. So WTF does the extension not *jump* to 101
> or 202 (or whatever the destination is) when a real person finally
> answers the call?
>
>> And then if you want the WHATIREALLYWANTEDINSTEAD value into your CDR,
>> see the earlier post this week regarding setting arbitrary values into
>> your CDR.
>
> It can't be this hard surely?
>
> We can't be the only firm in the world that doesn't do DDI and just has
> one incoming number?
>
> As I said, if while the caller is in the IVR they dial 101 it works
> properly. But some will not know our extension numbers so the IVR rings
> several handsets and the first one to pick up gets the call. Why isn't
> that information set as the destination EXTEN?
>
> I am beginning to think this is probably a bug. It has nothing to do
> with Macros. I have tried without.
>
> Alan
>
>>> [tolc_menu] ; Welcome and information to callers
>>> exten =>   s,1,Answer()
>>> exten =>   s,n,Wait(2)
>>> exten =>   s,n,Background(welcome-to-tolc) ; Say Hello
>>> exten =>   s,n,Wait(1)
>>> exten =>   s,n(tryagain),Background(enter-ext-of-person&or) ; Enter
>>> extension number if known, or
>>> exten =>   s,n,Background(pls-stay-on-line) ; Trying to connect...
>>> exten =>   s,n,WaitExten(5)
>>> exten =>   s,n,Macro(belllord,${ALANL}&${ALANB},303)
>>>
>>> exten =>   _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
>>>
>>> exten =>   _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
>>>
>>>
>>> The Vars ALANL and ALANB are:
>>> ALANL=SIP/101
>>> ALANB=IAX2/alanb/202
>>>
>>>
>>> Here is the Macro belllord:
>>>
>>> [macro-belllord]
>>> exten =>   s,1,Dial(${ARG1},20,t)
>>> exten =>   s,n,Goto(s-${DIALSTATUS},1)
>>>
>>> exten =>   s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
>>> voicemail context, ${ARG2} is the mailbox number to dial
>>> exten =>   s-NOANSWER,n,Hangup()
>>>
>>> exten =>   s-BUSY,1

[asterisk-users] Delete voicemail after couple of days

[Aloysius Thevarajah Lloyd]

Hi Every one,
Is there a way to delete voicemail's after couple of days?


Thank you.
Lloyd
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Re: [asterisk-users] Asterisk Error

[Steve Totaro]

It may be** noload => pbx_dundi.so or some such.  Sorry for being so vague
in my original answer but googling "noload dundi" would have given you the
same answer I just did.

You could probably safely just delete pbx_dundi.so instead/as well or
recompile Asterisk, do a make menuselect and remove dundi then make && make
install.

That should at least solve your dundi issue.

Thanks,
Steve Totaro

On Fri, Jul 17, 2009 at 9:01 AM, michel freiha  wrote:

> Dear Sir
>
> I did what you asked me to do...i added the following to
> /etc/opt/asterisk/modules.conf
>
> noload => dundi
>
> -bash-3.00# ifconfig -a
> lo0: flags=2001000849 mtu 8232
> index 1
> inet 127.0.0.1 netmask ff00
> eri0: flags=1000843 mtu 1500 index 2
> inet 192.168.0.178 netmask ff00 broadcast 192.168.0.255
> ether 0:3:ba:f2:d2:ea
>
>
> Yes I have a NIC, Up and running and I can SSH the server from that NIC
>
> Regards
>
> On Fri, Jul 17, 2009 at 3:21 PM, Steve Totaro <
> stot...@asteriskhelpdesk.com> wrote:
>
>>
>>
>> On Fri, Jul 17, 2009 at 2:08 AM, michel freiha  wrote:
>>
>>> Hi all,
>>>
>>> Can you please let me know what the below issue mean when trying to start
>>> asterisk and how I can fix it?
>>>
>>> pbx_dundi.c: No ethernet interface found for seeding global EID  You will
>>> have to set it manually.
>>>
>>> regards
>>>
>>
>> Add:
>> noload = dundi
>> To your modules.conf.  That should fix it.
>>
>> Do you want to use dundi?  What does ifconfig say?
>>
>> I assume you have a NIC?  Is it up and all that when you start Asterisk?
>> Have you tried downing it, setting all the variables (maybe even the MAC to
>> be thorough) and then bringing it back up before starting Asterisk?
>>
>> Otherwise what kind of NIC?  Do you have an old 3Com laying around you can
>> pop in it?
>>
>> Open a bug report?
>>
>> --
>> Thanks,
>> Steve Totaro
>> +18887771888 (Toll Free)
>> +12409381212 (Cell)
>> +12024369784 (Skype)
>>
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] quenstion about asterisk

[Steve Howes]

On 17 Jul 2009, at 16:26, Elvis Jorge wrote:

> The problem with read() is that I have to wait that a message that  
> is before
> read finish, I can use XXX,1 set(variable=${EXTEN}) but the user has  
> to type
> the quantity of digits predefine.
>
> Could you give me other solution?

Yes, the one I suggested a few hours ago. Read one digit at a time.

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Re: [asterisk-users] Mexican ITSP needed

[Carlos Chavez]

They are the oldest (4 years) VoIP provider here in Mexico.  I have
many lines with them for my company an clients and most of the time it
works very well.  

On Fri, 2009-07-17 at 07:26 +0200, Michiel van Baak wrote:
> On 11:39, Thu 16 Jul 09, Carlos Chavez wrote:
> > Try http://www.inext.com.mx they can provide DIDs in several cities in
> > Mexico.
> 
> Thanks.
> I asked the customer to have a look (I'm only capable of reading English
> and Dutch ;))
> 
> You have any experience with them ?
> 
> > 
> > On Thu, 2009-07-16 at 09:16 +0200, Michiel van Baak wrote:
> > > Hey all,
> > > 
> > > I was wondering if anyone knows about a Mexican ITSP I can connect to to
> > > route calls from and to my * boxen.
> > > 
> > > If it matters: I'm located in The Netherlands and one of our customers
> > > is in Mexico so if we need a Mexican presence that is not an issue.
> > > 
> > > Thanks.
> > > 
-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] PRI hunt group

[C F]

You have to pay LD rates.

On Fri, Jul 17, 2009 at 1:42 AM, Alex Balashov wrote:
> C F wrote:
>
>> If you don't want to port it to the PRI for whatever reason you can
>> convert it to a RCFW (remote call forwarded number) which is around
>> $15.00 plus $8.00 for each additional channel again pricing is for
>> here in Verizon land.
>
> Is that true even if the number is out of a rate center that is billed
> long-distance relative to the destination (but still intra-LATA)?  Or do
> you pay normal LD rates on top of all that in the intra-LATA LD scenario?
>
> --
> Alex Balashov
> Evariste Systems
> Web : http://www.evaristesys.com/
> Tel : (+1) (678) 954-0670
> Direct  : (+1) (678) 954-0671
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?

[Danny Nicholas]

Not that this will really help, but in my CDR, I get this find of format
Xxx incoming_number  s  context   caller_id   incoming_tech/line
target_tech/line  function   command   time1  time2  time3.  It seems that
you could look to the target_tech/line for the information you need.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
(News)
Sent: Friday, July 17, 2009 11:10 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How do I create an IVR/Dial Group that
worksproperly?

On 17/07/09 16:30, David Backeberg wrote:
> On Fri, Jul 17, 2009 at 4:22 AM, Alan Lord (News)
wrote:
>> * Caller arrives at our main number
>> * Caller is greeted and then told they can enter an extension number, if
>> known, or wait and their call will be connected to an available rep.
>> * The IVR then dials a group of extensions (if the caller didn't enter
>> one obviously).
>> * Someone picks up the call and the connection is established and logged.
>>
>> Now, I have all of this working apart from the last piece.
>>
>> My IVR rings various extensions and I can pick up the call just fine.
>> But my problem is that the data asterisk records regarding the call is
>> wrong.
>>
>> It correctly identifies the CallerID, but it always records the
>> destination as "s". Not the extension of, for example my SIP phone (101).
>
> Somewhere earlier, you do the very first answer. At that point, you should
add a
> NoOp(${EXTEN})
> Set(WHATIREALLYWANTEDINSTEAD=${EXTEN}
>
> and then keep popping out the
> ${WHATIREALLYWANTEDINSTEAD}
> value wherever you wanted the original extension before you started
> jumping all over the place in your dialplan.

I don't really understand what you are saying here. Sorry :-(

When the call first hits * (over an IAX trunk), it gets put into the IVR 
[tolc_menu} at s,1 and the extension in the IAX context is the incoming 
number. So there isn't an EXTEN at this stage. And I do not know 
WHATIREALLYWANTEDINSTEAD because:

a) the caller has not yet dialled an extension, or
b) I do not know which of us will answer the call.

> As you maybe guessed by now, EXTEN is the immediate, right now
> extension, and if you make jumps, it will update as you jump around.

Well, yes I understand that. So WTF does the extension not *jump* to 101 
or 202 (or whatever the destination is) when a real person finally 
answers the call?

> And then if you want the WHATIREALLYWANTEDINSTEAD value into your CDR,
> see the earlier post this week regarding setting arbitrary values into
> your CDR.

It can't be this hard surely?

We can't be the only firm in the world that doesn't do DDI and just has 
one incoming number?

As I said, if while the caller is in the IVR they dial 101 it works 
properly. But some will not know our extension numbers so the IVR rings 
several handsets and the first one to pick up gets the call. Why isn't 
that information set as the destination EXTEN?

I am beginning to think this is probably a bug. It has nothing to do 
with Macros. I have tried without.

Alan

>> [tolc_menu] ; Welcome and information to callers
>> exten =>  s,1,Answer()
>> exten =>  s,n,Wait(2)
>> exten =>  s,n,Background(welcome-to-tolc) ; Say Hello
>> exten =>  s,n,Wait(1)
>> exten =>  s,n(tryagain),Background(enter-ext-of-person&or) ; Enter
>> extension number if known, or
>> exten =>  s,n,Background(pls-stay-on-line) ; Trying to connect...
>> exten =>  s,n,WaitExten(5)
>> exten =>  s,n,Macro(belllord,${ALANL}&${ALANB},303)
>>
>> exten =>  _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
>>
>> exten =>  _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
>>
>>
>> The Vars ALANL and ALANB are:
>> ALANL=SIP/101
>> ALANB=IAX2/alanb/202
>>
>>
>> Here is the Macro belllord:
>>
>> [macro-belllord]
>> exten =>  s,1,Dial(${ARG1},20,t)
>> exten =>  s,n,Goto(s-${DIALSTATUS},1)
>>
>> exten =>  s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
>> voicemail context, ${ARG2} is the mailbox number to dial
>> exten =>  s-NOANSWER,n,Hangup()
>>
>> exten =>  s-BUSY,1,Voicemail(${ar...@business,b)
>> exten =>  s-BUSY,n,Hangup()
>>
>> exten =>  _s-.,1,Goto(s-NOANSWER,1)
>>
>>
>> Here is the call-extension Macro:
>>
>> [macro-call_extension]
>> exten =>  s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
>> exten =>  s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
>>
>> exten =>  s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
>>
>> exten =>  s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
>>
>> exten =>  _s-.,1,Goto(s-NOANSWER,1)
>>
>>
>>
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>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] How do I create an IVR/Dial Group that works properly?

[Alan Lord (News)]

On 17/07/09 16:30, David Backeberg wrote:
> On Fri, Jul 17, 2009 at 4:22 AM, Alan Lord (News)  
> wrote:
>> * Caller arrives at our main number
>> * Caller is greeted and then told they can enter an extension number, if
>> known, or wait and their call will be connected to an available rep.
>> * The IVR then dials a group of extensions (if the caller didn't enter
>> one obviously).
>> * Someone picks up the call and the connection is established and logged.
>>
>> Now, I have all of this working apart from the last piece.
>>
>> My IVR rings various extensions and I can pick up the call just fine.
>> But my problem is that the data asterisk records regarding the call is
>> wrong.
>>
>> It correctly identifies the CallerID, but it always records the
>> destination as "s". Not the extension of, for example my SIP phone (101).
>
> Somewhere earlier, you do the very first answer. At that point, you should 
> add a
> NoOp(${EXTEN})
> Set(WHATIREALLYWANTEDINSTEAD=${EXTEN}
>
> and then keep popping out the
> ${WHATIREALLYWANTEDINSTEAD}
> value wherever you wanted the original extension before you started
> jumping all over the place in your dialplan.

I don't really understand what you are saying here. Sorry :-(

When the call first hits * (over an IAX trunk), it gets put into the IVR 
[tolc_menu} at s,1 and the extension in the IAX context is the incoming 
number. So there isn't an EXTEN at this stage. And I do not know 
WHATIREALLYWANTEDINSTEAD because:

a) the caller has not yet dialled an extension, or
b) I do not know which of us will answer the call.

> As you maybe guessed by now, EXTEN is the immediate, right now
> extension, and if you make jumps, it will update as you jump around.

Well, yes I understand that. So WTF does the extension not *jump* to 101 
or 202 (or whatever the destination is) when a real person finally 
answers the call?

> And then if you want the WHATIREALLYWANTEDINSTEAD value into your CDR,
> see the earlier post this week regarding setting arbitrary values into
> your CDR.

It can't be this hard surely?

We can't be the only firm in the world that doesn't do DDI and just has 
one incoming number?

As I said, if while the caller is in the IVR they dial 101 it works 
properly. But some will not know our extension numbers so the IVR rings 
several handsets and the first one to pick up gets the call. Why isn't 
that information set as the destination EXTEN?

I am beginning to think this is probably a bug. It has nothing to do 
with Macros. I have tried without.

Alan

>> [tolc_menu] ; Welcome and information to callers
>> exten =>  s,1,Answer()
>> exten =>  s,n,Wait(2)
>> exten =>  s,n,Background(welcome-to-tolc) ; Say Hello
>> exten =>  s,n,Wait(1)
>> exten =>  s,n(tryagain),Background(enter-ext-of-person&or) ; Enter
>> extension number if known, or
>> exten =>  s,n,Background(pls-stay-on-line) ; Trying to connect...
>> exten =>  s,n,WaitExten(5)
>> exten =>  s,n,Macro(belllord,${ALANL}&${ALANB},303)
>>
>> exten =>  _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
>>
>> exten =>  _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
>>
>>
>> The Vars ALANL and ALANB are:
>> ALANL=SIP/101
>> ALANB=IAX2/alanb/202
>>
>>
>> Here is the Macro belllord:
>>
>> [macro-belllord]
>> exten =>  s,1,Dial(${ARG1},20,t)
>> exten =>  s,n,Goto(s-${DIALSTATUS},1)
>>
>> exten =>  s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
>> voicemail context, ${ARG2} is the mailbox number to dial
>> exten =>  s-NOANSWER,n,Hangup()
>>
>> exten =>  s-BUSY,1,Voicemail(${ar...@business,b)
>> exten =>  s-BUSY,n,Hangup()
>>
>> exten =>  _s-.,1,Goto(s-NOANSWER,1)
>>
>>
>> Here is the call-extension Macro:
>>
>> [macro-call_extension]
>> exten =>  s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
>> exten =>  s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
>>
>> exten =>  s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
>>
>> exten =>  s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
>>
>> exten =>  _s-.,1,Goto(s-NOANSWER,1)
>>
>>
>>
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>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?

[Alan Lord (News)]

On 17/07/09 16:29, Adam Robins wrote:
> Have you tried replacing the "s" extension with "_x."?

Thanks, yes I have.

Unfortunately, all that did was to change "s" to the number of our 
incoming trunk (i.e. the dialled number). It still does not get set to 
the number of the final extension to which the call gets connected.

Cheers

Alan

>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord (News)
> Sent: Friday, July 17, 2009 11:12 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] How do I create an IVR/Dial Group that 
> worksproperly?
>
> On 17/07/09 14:14, Danny Nicholas wrote:
>> I may 100% off here, but I seem to recall reading in the last 2 days threads
>> that macro dialing messes with CDR entries.  I would try replacing one of
>> your macro lines with a straight Dial command to verify this.
>
> Thanks Danny, but that doesn't really help. I have tried moving the
> contents of the offending Macro into the IVR menu itself and using a
> Dial() command. But it makes no difference. The call is still on the "s"
> extension and the CDR records the connection with the correct callerid
> but with the destination as "s". Which is not what I want.
>
> If the caller dials an extension number, say 101, then it all works
> fine. The problem is when trying to automatically dial from within the
> plan it fails. I need to somehow change "s" to the end extension number
> of the person who actually picks up the phone.
>
> I am trying to understand how other people configure their * to achieve
> the requirement I specified below.
>
> I can't believe it is this hard to do. But I fail to see how I can
> achieve it, because there is no extension - other than "s" - when the
> caller enters the dialplan. I want the caller to be automatically
> connected to one or other of our extensions if they do not know the
> extension number to dial themselves.
>
> I guess I am trying to find out if I have set this up totally *wrong*
> and perhaps I should be using a queue or something, but that seems a bit
> overkill...
>
> Alan
>
>
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
>> (News)
>> Sent: Friday, July 17, 2009 3:23 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] How do I create an IVR/Dial Group that
>> worksproperly?
>>
>> Hi all,
>>
>> I am trying to understand how I can get a simple IVR scenario to work
>> properly (having already removed most of my hair...).
>>
>> The basic requirement is as follows:
>>
>> * Caller arrives at our main number
>> * Caller is greeted and then told they can enter an extension number, if
>> known, or wait and their call will be connected to an available rep.
>> * The IVR then dials a group of extensions (if the caller didn't enter
>> one obviously).
>> * Someone picks up the call and the connection is established and logged.
>>
>> Now, I have all of this working apart from the last piece.
>>
>> My IVR rings various extensions and I can pick up the call just fine.
>> But my problem is that the data asterisk records regarding the call is
>> wrong.
>>
>> It correctly identifies the CallerID, but it always records the
>> destination as "s". Not the extension of, for example my SIP phone (101).
>>
>> If the incoming caller dials 101 whilst in the IVR, the log is correct.
>>
>> I can see *why* I am having this problem (There is no extension when you
>> arrive in the IVR other than "s"), but I cannot see *how* to fix it.
>>
>> Please can I ask how do others handle this so it works properly (I've
>> included the basics of my DP below)?
>>
>> I'm running Asterisk 1.4.21.2~dfsg-1ubuntu3 on Ubuntu Server 8.10.
>>
>> Thanks
>>
>> Alan
>>
>>
>> Here is the IVR which callers are dropped into:
>>
>> [tolc_menu] ; Welcome and information to callers
>> exten =>   s,1,Answer()
>> exten =>   s,n,Wait(2)
>> exten =>   s,n,Background(welcome-to-tolc) ; Say Hello
>> exten =>   s,n,Wait(1)
>> exten =>   s,n(tryagain),Background(enter-ext-of-person&or) ; Enter
>> extension number if known, or
>> exten =>   s,n,Background(pls-stay-on-line) ; Trying to connect...
>> exten =>   s,n,WaitExten(5)
>> exten =>   s,n,Macro(belllord,${ALANL}&${ALANB},303)
>>
>> exten =>   _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
>>
>> exten =>   _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
>>
>>
>> The Vars ALANL and ALANB are:
>> ALANL=SIP/101
>> ALANB=IAX2/alanb/202
>>
>>
>> Here is the Macro belllord:
>>
>> [macro-belllord]
>> exten =>   s,1,Dial(${ARG1},20,t)
>> exten =>   s,n,Goto(s-${DIALSTATUS},1)
>>
>> exten =>   s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
>> voicemail context, ${ARG2} is the mailbox number to dial
>> exten =>   s-NOANSWER,n,Hangup()
>>
>> exten =>   s-BUSY,1,Voicemail(${ar...@business,b)
>> exten =>   s-BUSY,n,Hangup()
>>
>> exten =>   

Re: [asterisk-users] How do I create an IVR/Dial Group that works properly?

[David Backeberg]

On Fri, Jul 17, 2009 at 4:22 AM, Alan Lord (News) wrote:
> * Caller arrives at our main number
> * Caller is greeted and then told they can enter an extension number, if
> known, or wait and their call will be connected to an available rep.
> * The IVR then dials a group of extensions (if the caller didn't enter
> one obviously).
> * Someone picks up the call and the connection is established and logged.
>
> Now, I have all of this working apart from the last piece.
>
> My IVR rings various extensions and I can pick up the call just fine.
> But my problem is that the data asterisk records regarding the call is
> wrong.
>
> It correctly identifies the CallerID, but it always records the
> destination as "s". Not the extension of, for example my SIP phone (101).

Somewhere earlier, you do the very first answer. At that point, you should add a
NoOp(${EXTEN})
Set(WHATIREALLYWANTEDINSTEAD=${EXTEN}

and then keep popping out the
${WHATIREALLYWANTEDINSTEAD}
value wherever you wanted the original extension before you started
jumping all over the place in your dialplan.

As you maybe guessed by now, EXTEN is the immediate, right now
extension, and if you make jumps, it will update as you jump around.

And then if you want the WHATIREALLYWANTEDINSTEAD value into your CDR,
see the earlier post this week regarding setting arbitrary values into
your CDR.

> [tolc_menu] ; Welcome and information to callers
> exten => s,1,Answer()
> exten => s,n,Wait(2)
> exten => s,n,Background(welcome-to-tolc) ; Say Hello
> exten => s,n,Wait(1)
> exten => s,n(tryagain),Background(enter-ext-of-person&or) ; Enter
> extension number if known, or
> exten => s,n,Background(pls-stay-on-line) ; Trying to connect...
> exten => s,n,WaitExten(5)
> exten => s,n,Macro(belllord,${ALANL}&${ALANB},303)
>
> exten => _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
>
> exten => _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
>
>
> The Vars ALANL and ALANB are:
> ALANL=SIP/101
> ALANB=IAX2/alanb/202
>
>
> Here is the Macro belllord:
>
> [macro-belllord]
> exten => s,1,Dial(${ARG1},20,t)
> exten => s,n,Goto(s-${DIALSTATUS},1)
>
> exten => s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
> voicemail context, ${ARG2} is the mailbox number to dial
> exten => s-NOANSWER,n,Hangup()
>
> exten => s-BUSY,1,Voicemail(${ar...@business,b)
> exten => s-BUSY,n,Hangup()
>
> exten => _s-.,1,Goto(s-NOANSWER,1)
>
>
> Here is the call-extension Macro:
>
> [macro-call_extension]
> exten => s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
> exten => s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
>
> exten => s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
>
> exten => s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
>
> exten => _s-.,1,Goto(s-NOANSWER,1)
>
>
>
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Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?

[Adam Robins]

Have you tried replacing the "s" extension with "_x."?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord (News)
Sent: Friday, July 17, 2009 11:12 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How do I create an IVR/Dial Group that 
worksproperly?

On 17/07/09 14:14, Danny Nicholas wrote:
> I may 100% off here, but I seem to recall reading in the last 2 days threads
> that macro dialing messes with CDR entries.  I would try replacing one of
> your macro lines with a straight Dial command to verify this.

Thanks Danny, but that doesn't really help. I have tried moving the
contents of the offending Macro into the IVR menu itself and using a
Dial() command. But it makes no difference. The call is still on the "s"
extension and the CDR records the connection with the correct callerid
but with the destination as "s". Which is not what I want.

If the caller dials an extension number, say 101, then it all works
fine. The problem is when trying to automatically dial from within the
plan it fails. I need to somehow change "s" to the end extension number
of the person who actually picks up the phone.

I am trying to understand how other people configure their * to achieve
the requirement I specified below.

I can't believe it is this hard to do. But I fail to see how I can
achieve it, because there is no extension - other than "s" - when the
caller enters the dialplan. I want the caller to be automatically
connected to one or other of our extensions if they do not know the
extension number to dial themselves.

I guess I am trying to find out if I have set this up totally *wrong*
and perhaps I should be using a queue or something, but that seems a bit
overkill...

Alan


>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
> (News)
> Sent: Friday, July 17, 2009 3:23 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] How do I create an IVR/Dial Group that
> worksproperly?
>
> Hi all,
>
> I am trying to understand how I can get a simple IVR scenario to work
> properly (having already removed most of my hair...).
>
> The basic requirement is as follows:
>
> * Caller arrives at our main number
> * Caller is greeted and then told they can enter an extension number, if
> known, or wait and their call will be connected to an available rep.
> * The IVR then dials a group of extensions (if the caller didn't enter
> one obviously).
> * Someone picks up the call and the connection is established and logged.
>
> Now, I have all of this working apart from the last piece.
>
> My IVR rings various extensions and I can pick up the call just fine.
> But my problem is that the data asterisk records regarding the call is
> wrong.
>
> It correctly identifies the CallerID, but it always records the
> destination as "s". Not the extension of, for example my SIP phone (101).
>
> If the incoming caller dials 101 whilst in the IVR, the log is correct.
>
> I can see *why* I am having this problem (There is no extension when you
> arrive in the IVR other than "s"), but I cannot see *how* to fix it.
>
> Please can I ask how do others handle this so it works properly (I've
> included the basics of my DP below)?
>
> I'm running Asterisk 1.4.21.2~dfsg-1ubuntu3 on Ubuntu Server 8.10.
>
> Thanks
>
> Alan
>
>
> Here is the IVR which callers are dropped into:
>
> [tolc_menu] ; Welcome and information to callers
> exten =>  s,1,Answer()
> exten =>  s,n,Wait(2)
> exten =>  s,n,Background(welcome-to-tolc) ; Say Hello
> exten =>  s,n,Wait(1)
> exten =>  s,n(tryagain),Background(enter-ext-of-person&or) ; Enter
> extension number if known, or
> exten =>  s,n,Background(pls-stay-on-line) ; Trying to connect...
> exten =>  s,n,WaitExten(5)
> exten =>  s,n,Macro(belllord,${ALANL}&${ALANB},303)
>
> exten =>  _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
>
> exten =>  _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
>
>
> The Vars ALANL and ALANB are:
> ALANL=SIP/101
> ALANB=IAX2/alanb/202
>
>
> Here is the Macro belllord:
>
> [macro-belllord]
> exten =>  s,1,Dial(${ARG1},20,t)
> exten =>  s,n,Goto(s-${DIALSTATUS},1)
>
> exten =>  s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
> voicemail context, ${ARG2} is the mailbox number to dial
> exten =>  s-NOANSWER,n,Hangup()
>
> exten =>  s-BUSY,1,Voicemail(${ar...@business,b)
> exten =>  s-BUSY,n,Hangup()
>
> exten =>  _s-.,1,Goto(s-NOANSWER,1)
>
>
> Here is the call-extension Macro:
>
> [macro-call_extension]
> exten =>  s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
> exten =>  s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
>
> exten =>  s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
>
> exten =>  s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
>
> exten =>  _s-.,1,Goto(s-NOANSWER,1)
>
>
>
> _

Re: [asterisk-users] dialplan number matching

[John A. Sullivan III]

On Fri, 2009-07-17 at 02:11 -0700, Vieri wrote:
> Hi,
> 
> How can I match an extension "ending with 3" (just an example but applicable 
> to any other digit, including * or #)?
> 
> exten => _ZX.3,n,...
> 
> exten => _ZX.#,n,...
> 
> (the above does not work)
> 
> Can regular expressions be used in the standard dialplan (end with: "$")?
> 
> Thanks,
> 
> Vieri

I haven't tried it but I wonder if one could use a regex pattern match
in a GotoIf statement and then pass the result to another context using
${EXTEN}? Just a thought - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


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Re: [asterisk-users] quenstion about asterisk

[Elvis Jorge]

The problem with read() is that I have to wait that a message that is before 
read finish, I can use XXX,1 set(variable=${EXTEN}) but the user has to type 
the quantity of digits predefine.

Could you give me other solution?

Thanks

- Original Message - 
From: "Steve Edwards" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, July 17, 2009 11:06 AM
Subject: Re: [asterisk-users] quenstion about asterisk


> On 17 Jul 2009, at 15:29, Elvis Jorge wrote:

>> I want to know if there´s a way to capture the numbers typed for a
>> user; without waiting that the IVR finish or without predefine the
>> numbers of digits. I´m going to explain you better, for example I want
>> to know that a user typed 12345#,but I want that the user can type over
>> IVR and don't predefine the numbers of digits X because the user
>> should have the quantity the digits predefine.

On Fri, 17 Jul 2009, Steve Howes wrote:

> Assuming you intend to use # as a terminator, just collect in a loop, 1
> digit at a time until you get a hash..

Or, use read() or AGI's "stream file."

For future reference, please take a look at:

  http://www.catb.org/~esr/faqs/smart-questions.html#bespecific

There are many questions about Asterisk.

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





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Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?

[Alan Lord (News)]

On 17/07/09 14:14, Danny Nicholas wrote:
> I may 100% off here, but I seem to recall reading in the last 2 days threads
> that macro dialing messes with CDR entries.  I would try replacing one of
> your macro lines with a straight Dial command to verify this.

Thanks Danny, but that doesn't really help. I have tried moving the 
contents of the offending Macro into the IVR menu itself and using a 
Dial() command. But it makes no difference. The call is still on the "s" 
extension and the CDR records the connection with the correct callerid 
but with the destination as "s". Which is not what I want.

If the caller dials an extension number, say 101, then it all works 
fine. The problem is when trying to automatically dial from within the 
plan it fails. I need to somehow change "s" to the end extension number 
of the person who actually picks up the phone.

I am trying to understand how other people configure their * to achieve 
the requirement I specified below.

I can't believe it is this hard to do. But I fail to see how I can 
achieve it, because there is no extension - other than "s" - when the 
caller enters the dialplan. I want the caller to be automatically 
connected to one or other of our extensions if they do not know the 
extension number to dial themselves.

I guess I am trying to find out if I have set this up totally *wrong* 
and perhaps I should be using a queue or something, but that seems a bit 
overkill...

Alan


>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
> (News)
> Sent: Friday, July 17, 2009 3:23 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] How do I create an IVR/Dial Group that
> worksproperly?
>
> Hi all,
>
> I am trying to understand how I can get a simple IVR scenario to work
> properly (having already removed most of my hair...).
>
> The basic requirement is as follows:
>
> * Caller arrives at our main number
> * Caller is greeted and then told they can enter an extension number, if
> known, or wait and their call will be connected to an available rep.
> * The IVR then dials a group of extensions (if the caller didn't enter
> one obviously).
> * Someone picks up the call and the connection is established and logged.
>
> Now, I have all of this working apart from the last piece.
>
> My IVR rings various extensions and I can pick up the call just fine.
> But my problem is that the data asterisk records regarding the call is
> wrong.
>
> It correctly identifies the CallerID, but it always records the
> destination as "s". Not the extension of, for example my SIP phone (101).
>
> If the incoming caller dials 101 whilst in the IVR, the log is correct.
>
> I can see *why* I am having this problem (There is no extension when you
> arrive in the IVR other than "s"), but I cannot see *how* to fix it.
>
> Please can I ask how do others handle this so it works properly (I've
> included the basics of my DP below)?
>
> I'm running Asterisk 1.4.21.2~dfsg-1ubuntu3 on Ubuntu Server 8.10.
>
> Thanks
>
> Alan
>
>
> Here is the IVR which callers are dropped into:
>
> [tolc_menu] ; Welcome and information to callers
> exten =>  s,1,Answer()
> exten =>  s,n,Wait(2)
> exten =>  s,n,Background(welcome-to-tolc) ; Say Hello
> exten =>  s,n,Wait(1)
> exten =>  s,n(tryagain),Background(enter-ext-of-person&or) ; Enter
> extension number if known, or
> exten =>  s,n,Background(pls-stay-on-line) ; Trying to connect...
> exten =>  s,n,WaitExten(5)
> exten =>  s,n,Macro(belllord,${ALANL}&${ALANB},303)
>
> exten =>  _10[1-5],1,Macro(call_extension,SIP/${EXTEN})
>
> exten =>  _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})
>
>
> The Vars ALANL and ALANB are:
> ALANL=SIP/101
> ALANB=IAX2/alanb/202
>
>
> Here is the Macro belllord:
>
> [macro-belllord]
> exten =>  s,1,Dial(${ARG1},20,t)
> exten =>  s,n,Goto(s-${DIALSTATUS},1)
>
> exten =>  s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
> voicemail context, ${ARG2} is the mailbox number to dial
> exten =>  s-NOANSWER,n,Hangup()
>
> exten =>  s-BUSY,1,Voicemail(${ar...@business,b)
> exten =>  s-BUSY,n,Hangup()
>
> exten =>  _s-.,1,Goto(s-NOANSWER,1)
>
>
> Here is the call-extension Macro:
>
> [macro-call_extension]
> exten =>  s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
> exten =>  s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.
>
> exten =>  s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)
>
> exten =>  s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)
>
> exten =>  _s-.,1,Goto(s-NOANSWER,1)
>
>
>
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>
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Re: [asterisk-users] quenstion about asterisk

[Steve Edwards]

On 17 Jul 2009, at 15:29, Elvis Jorge wrote:


I want to know if there?s a way to capture the numbers typed for a 
user; without waiting that the IVR finish or without predefine the 
numbers of digits. I?m going to explain you better, for example I want 
to know that a user typed 12345#,but I want that the user can type over 
IVR and don't predefine the numbers of digits X because the user 
should have the quantity the digits predefine.


On Fri, 17 Jul 2009, Steve Howes wrote:

Assuming you intend to use # as a terminator, just collect in a loop, 1 
digit at a time until you get a hash..


Or, use read() or AGI's "stream file."

For future reference, please take a look at:

http://www.catb.org/~esr/faqs/smart-questions.html#bespecific

There are many questions about Asterisk.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000___
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[asterisk-users] Compilation error

[michel freiha]

Dear Sir,

I'm trying to install asterisk 1.6.1.1 on solaris 10...At the end of gmake I
got the below error


creating config.h
In file included from sig.h:47,
 from el.h:107,
 from common.c:51,
 from editline.c:4:
/usr/include/signal.h:77: error: syntax error before '*' token
gmake[2]: *** [editline.o_a] Error 1
gmake[1]: *** [editline/libedit.a] Error 2
gmake: *** [main] Error 2

Can you help me please in fixing it?

Regards
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Re: [asterisk-users] quenstion about asterisk

[Elvis Jorge]

Could you give a example how I can do that??

Thanks


- Original Message - 
From: "Steve Howes" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, July 17, 2009 10:34 AM
Subject: Re: [asterisk-users] quenstion about asterisk



On 17 Jul 2009, at 15:29, Elvis Jorge wrote:
> I want to know if there´s a way to capture the numbers typed for a
> user; without waiting that the IVR finish or without predefine the
> numbers of digits. I´m going to explain you better, for example I
> want to know that a user typed 12345#,but I want that the user can
> type over IVR and don't predefine the numbers of digits X
> because the user should have the quantity the digits predefine.

Assuming you intend to use # as a terminator, just collect in a loop,
1 digit at a time until you get a hash..

S
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Re: [asterisk-users] quenstion about asterisk

[Steve Howes]

On 17 Jul 2009, at 15:29, Elvis Jorge wrote:
> I want to know if there´s a way to capture the numbers typed for a  
> user; without waiting that the IVR finish or without predefine the  
> numbers of digits. I´m going to explain you better, for example I  
> want to know that a user typed 12345#,but I want that the user can  
> type over IVR and don't predefine the numbers of digits X  
> because the user should have the quantity the digits predefine.

Assuming you intend to use # as a terminator, just collect in a loop,  
1 digit at a time until you get a hash..

S
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[asterisk-users] quenstion about asterisk

[Elvis Jorge]

Hello fellows,

I want to know if there´s a way to capture the numbers typed for a user; 
without waiting that the IVR finish or without predefine the numbers of digits. 
I´m going to explain you better, for example I want to know that a user typed 
12345#, but I want that the user can type over IVR and don't predefine the 
numbers of digits X because the user should have the quantity the digits 
predefine.

Thanks

Elvis Jorge
Cell: 809-706-8824
ETGTEL DOMINICANA
La información contenida en este correo electrónico, así como los archivos 
anexos que pudiera incluir, es confidencial y únicamente para su destinatario. 
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Re: [asterisk-users] Skill based routing

[Steve Edwards]

> Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
>
>> We are trying to implement skill based routing for agents in a support 
>> centre based on the agent login. Has anyone had any experience with 
>> this and what was the outcome?
>
On Fri, 17 Jul 2009, Alex Balashov wrote:

> It can't really be done using Asterisk queues, unless you want to create 
> a large number of queues for every relevant skill factor and have agents 
> join various combinations of these simultaneously--which would take 
> quite a bit of dial plan and/or AGI logic to pull off.  Plus, that 
> doesn't scale any given queue beyond one host.
>
> I suggest you look into using FastAGI[1] to simulate the queue 
> experience by generating hold music and announcements without actually 
> using Asterisk queues per se.  This is quite possible to do, and, this 
> allows you to distribute queues across multiple hosts, as well as 
> distribute calls within those queues by whatever logic you choose.  No 
> shoehorning--just write it yourself.

I did this for an adult chat system many moons ago with "local" AGIs 
written in C. When an agent logs in they land in a separate meetme. When 
callers select (via DNIS and/or IVR) which "skill" they are interested in, 
an AGI locates the "most idle" agent with that skill and routes the caller 
to that agent's meetme. The agent state (skills, logged in, busy, meetme 
name) is maintained in the database. The system is limited to a single 
host, but that was due to lack of foresight. Adding the host to the agent 
state in the database would not be a major change.

> [1] Yes, FastAGI.  Not local AGI.  And most especially not in PHP;
> contrary to a lot of the info out there, PHP could not possibly
> be a less suitable language in which to write AGI scripts.  I
> don't know who comes up with these lavish heights of mediocrity.

A properly written FastAGI is significantly more complex than a "local" 
AGI and if unexpectedly terminated, adversely affects all calls. Plus, you 
can update "local" AGIs without affecting calls in progress.

While you can execute xxx's of AGIs written in C in the time it takes to 
load and parse Perl or PHP, I do find associative arrays kind of seductive 
on occasion.

Besides performance and footprint, why do you single out PHP. Or do you 
object to all script languages?

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

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Re: [asterisk-users] T38 negotiation, the last step !

[Steve Underwood]

Klaus Darilion wrote:
> Xavier Cardil schrieb:
>   
>> Hi, I've managed to get HYLAFAX>T38MODEM->
>> ASTERISK>CISCOAS5400 working, but when they are negotiating asterisk 
>> drops a message telling "Unknown RTP codec 96 received from gateway" Do 
>> somebody know how to fix it ? 
>>
>> Thank you !
>>
>>
>>
>> << [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-600bfcc8]
>> << [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/GWCISCO5400O-600bfcc8]
>> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
>> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
>> << [ TYPE: Control (4) SUBCLASS: T38/Negotiation Requested (19) ] 
>> [SIP/GWCISCO5400O-600bfcc8]
>> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
>> [Jul 16 17:50:39] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP 
>> codec 96 received from '192.168.3.163'
>> 
>
> Is '192.168.3.163' the Cisco GW? If yes, then it is probably an NSE 
> packet. NSE is a Cisco proprietary FaxoverIP solution and uses per 
> default payload types 96 and 97 to signal a changeover from VoIP to FoIP.
>
> Probably you have to configure the Cisco GW to use T.38 instead of NSE 
> for FoIP.
>   
When did RFC2833 become a proprietary Cisco spec?

The NSEs just signal the startup of FAX. You still need to switch into 
T.38 to actually exchange the FAX.

Steve


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[asterisk-users] How to Play IVR and Read DTMF During Active Call?

[Faheem]

Hey, 

Is there any way to play IVR and Read DTMF during active call. 

Call Flow:
 
   USER1(initiator)  <-> Asterisk1 <-> Asterisk2 <->    USER2

How I can Play IVR and Read DTMF from USER1 When both users are in active 
session.

I am able to play IVR and Read DTMF from USER2, which is not required,
When Asterisk2 Receives call from Asterisk1, it simply 
Dial(SIP/${EXTEN},,,M(macro1)) and execute the macro1. In macro1 I play the IVR 
and Read() DTMF. 

The actuall problem comes here; 
IVR is playing in USER2 side only, infact It should play on both sides.
How I overcome that oneway voice problem. Please give your sugession.
I am using asterisk 1.4 on making SIP calls in Local test environment with no 
NAT issues there.

Thank you

Muhammad Faheem




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Re: [asterisk-users] MoH - can the volume be adjusted

[Danny Nicholas]

The default moh does not support volume adjustment.  However, if you change
musiconhold.conf to use the [custom] setting and use mpg123 as your player,
you will then have reasonably full adjustability.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rupert
Utteridge - Digital Techniques (Austalia) Limited
Sent: Friday, July 17, 2009 1:33 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MoH - can the volume be adjusted

 

We have noticed MoH volume levels vary, very much depending on the terminal
device that is connected. Within Asterisk is there any AGC or level control
available to compensate for the varying terminal devices and their levels?

 

For example a Polycom IP 7000 has very audible level while X Lite on a Dell
laptop has very low level and Cisco 7970G have lower level than the Polycom
IP 7000.

 

What is the experience of other users and how have you handled this level
variation?

 

Rupert

 

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Re: [asterisk-users] How do I create an IVR/Dial Group that worksproperly?

[Danny Nicholas]

I may 100% off here, but I seem to recall reading in the last 2 days threads
that macro dialing messes with CDR entries.  I would try replacing one of
your macro lines with a straight Dial command to verify this.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord
(News)
Sent: Friday, July 17, 2009 3:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How do I create an IVR/Dial Group that
worksproperly?

Hi all,

I am trying to understand how I can get a simple IVR scenario to work 
properly (having already removed most of my hair...).

The basic requirement is as follows:

* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if 
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of extensions (if the caller didn't enter 
one obviously).
* Someone picks up the call and the connection is established and logged.

Now, I have all of this working apart from the last piece.

My IVR rings various extensions and I can pick up the call just fine. 
But my problem is that the data asterisk records regarding the call is 
wrong.

It correctly identifies the CallerID, but it always records the 
destination as "s". Not the extension of, for example my SIP phone (101).

If the incoming caller dials 101 whilst in the IVR, the log is correct.

I can see *why* I am having this problem (There is no extension when you 
arrive in the IVR other than "s"), but I cannot see *how* to fix it.

Please can I ask how do others handle this so it works properly (I've 
included the basics of my DP below)?

I'm running Asterisk 1.4.21.2~dfsg-1ubuntu3 on Ubuntu Server 8.10.

Thanks

Alan


Here is the IVR which callers are dropped into:

[tolc_menu] ; Welcome and information to callers
exten => s,1,Answer()
exten => s,n,Wait(2)
exten => s,n,Background(welcome-to-tolc) ; Say Hello
exten => s,n,Wait(1)
exten => s,n(tryagain),Background(enter-ext-of-person&or) ; Enter
extension number if known, or
exten => s,n,Background(pls-stay-on-line) ; Trying to connect...
exten => s,n,WaitExten(5)
exten => s,n,Macro(belllord,${ALANL}&${ALANB},303)

exten => _10[1-5],1,Macro(call_extension,SIP/${EXTEN})

exten => _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})


The Vars ALANL and ALANB are:
ALANL=SIP/101
ALANB=IAX2/alanb/202


Here is the Macro belllord:

[macro-belllord]
exten => s,1,Dial(${ARG1},20,t)
exten => s,n,Goto(s-${DIALSTATUS},1)

exten => s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
voicemail context, ${ARG2} is the mailbox number to dial
exten => s-NOANSWER,n,Hangup()

exten => s-BUSY,1,Voicemail(${ar...@business,b)
exten => s-BUSY,n,Hangup()

exten => _s-.,1,Goto(s-NOANSWER,1)


Here is the call-extension Macro:

[macro-call_extension]
exten => s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
exten => s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.

exten => s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)

exten => s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)

exten => _s-.,1,Goto(s-NOANSWER,1)



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Re: [asterisk-users] Skill based routing

[Steve Totaro]

Just use FastAGI to hit a little process that queries a database and returns
the extensions of the "most skilled"

Have FastAGI return those extensions and pass them to a dial command with
the m flag (music if memory serves me correctly (pre-coffee)  Make the
"music" the standard junk you hear while waiting in a queue.

Be sure the extensions you are calling don't have voicemail and you are
obviously going to lose round robin and other dialing schemes unless you
come up with some other logic.

Multiple queues seems like the approach I would take.  AMI can change agent
penalties on the fly as well as add and remove them from queues.  Just an
FYI.

Maybe a mix of multiple queues, FastAGI, dialplan logic, and AMI.

Should be a fun project to get working and then fine tune.

Thanks,
Steve Totaro

On Fri, Jul 17, 2009 at 6:36 AM, Julian Lyndon-Smith wrote:

> Um, I really don't know - we just use the periodic messages to play
> the traditional "Your call is important to use" (whatever the
> wording..)
>
> Julian.
>
> 2009/7/17 Alex Balashov :
> >
> > What value do the queue announcements (I am assuming these are pertaining
> > to expected hold time, etc.) if there is only one agent?
> >
> >> We use a queue so that we can have all the benefits of the queue
> >> whilst finding an agent : music on hold, periodic announcements etc
> >> etc.
> >>
> >> You are right - with a little more effort we could probably remove the
> >> need for the queue. But why would I do that if I can use the queue for
> >> the bits I want ;)
> >>
> >> Julian
> >>
> >> 2009/7/17 Alex Balashov :
> >>>
> >>> The simplicity of this approach is elegant, but in that case, why use a
> >>> queue?  Why not just perform this logic straight in the dial plan when
> >>> processing the received call?
> >>>
> >>> The benefit of queues arises from their ability to keep state;  they
> can
> >>> retry agents, carry out different ring strategies, etc.  I understood
> >>> the
> >>> original question to be implicitly about incorporating weights for
> >>> skills
> >>> into queue or queue-like call distribution mechanisms, since that is
> how
> >>> it is done in call center products.  If the question is simply how to
> >>> make
> >>> Asterisk consider certain outside information when choosing to whom to
> >>> route a call, the answer would be that it is identical to the process
> >>> for
> >>> embedding any other kind of logic and/or outside data source into call
> >>> processing.
> >>>
>  Another simple way is to add local/foo/n as the only "agent" on the
>  queue. In the dialplan for local/foo , interrogate a database for the
>  most appropriate "agent" and then call that agent's extension.
> 
>  Julian
> 
>  2009/7/17 Matt Florell :
> > On 7/17/09, Alex Balashov  wrote:
> >> Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
> >>
> >>  > We are trying to implement skill based routing for agents in a
> >> support
> >>  > centre based on the agent login. Has anyone had any experience
> >> with
> >> this
> >>  > and what was the outcome?
> >>
> >>
> >> It can't really be done using Asterisk queues, unless you want to
> >> create
> >>  a large number of queues for every relevant skill factor and have
> >> agents
> >>  join various combinations of these simultaneously--which would take
> >>  quite a bit of dial plan and/or AGI logic to pull off.  Plus, that
> >>  doesn't scale any given queue beyond one host.
> >>
> >>  I suggest you look into using FastAGI[1] to simulate the queue
> >>  experience by generating hold music and announcements without
> >> actually
> >>  using Asterisk queues per se.  This is quite possible to do, and,
> >> this
> >>  allows you to distribute queues across multiple hosts, as well as
> >>  distribute calls within those queues by whatever logic you choose.
> >>  No
> >>  shoehorning--just write it yourself.
> >>
> >>  -- Alex
> >>
> >>  [1] Yes, FastAGI.  Not local AGI.  And most especially not in PHP;
> >>  contrary to a lot of the info out there, PHP could not possibly
> >>  be a less suitable language in which to write AGI scripts.  I
> >>  don't know who comes up with these lavish heights of
> mediocrity.
> >
> > If you are not looking to write it yourself you could always try
> > ViciDial which has skills-based routing built in, and it's free and
> > Open Source.
> >
> > MATT---
> >
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> >
> 
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Re: [asterisk-users] dialplan number matching

[Danny Nicholas]

Assuming you are using 4 digit extensions, this syntax would be:
- exten => _ZXX3,n,...
For 3 digits
- exten => _ZX3,n,...
The . is a wildcard that says "take rest of number, so anything after that
is irrelevant.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent: Friday, July 17, 2009 4:11 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] dialplan number matching


Hi,

How can I match an extension "ending with 3" (just an example but applicable
to any other digit, including * or #)?

exten => _ZX.3,n,...

exten => _ZX.#,n,...

(the above does not work)

Can regular expressions be used in the standard dialplan (end with: "$")?

Thanks,

Vieri



  

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Re: [asterisk-users] Asterisk Error

[michel freiha]

Dear Sir

I did what you asked me to do...i added the following to
/etc/opt/asterisk/modules.conf

noload => dundi

-bash-3.00# ifconfig -a
lo0: flags=2001000849 mtu 8232
index 1
inet 127.0.0.1 netmask ff00
eri0: flags=1000843 mtu 1500 index 2
inet 192.168.0.178 netmask ff00 broadcast 192.168.0.255
ether 0:3:ba:f2:d2:ea


Yes I have a NIC, Up and running and I can SSH the server from that NIC

Regards

On Fri, Jul 17, 2009 at 3:21 PM, Steve Totaro
wrote:

>
>
> On Fri, Jul 17, 2009 at 2:08 AM, michel freiha  wrote:
>
>> Hi all,
>>
>> Can you please let me know what the below issue mean when trying to start
>> asterisk and how I can fix it?
>>
>> pbx_dundi.c: No ethernet interface found for seeding global EID  You will
>> have to set it manually.
>>
>> regards
>>
>
> Add:
> noload = dundi
> To your modules.conf.  That should fix it.
>
> Do you want to use dundi?  What does ifconfig say?
>
> I assume you have a NIC?  Is it up and all that when you start Asterisk?
> Have you tried downing it, setting all the variables (maybe even the MAC to
> be thorough) and then bringing it back up before starting Asterisk?
>
> Otherwise what kind of NIC?  Do you have an old 3Com laying around you can
> pop in it?
>
> Open a bug report?
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
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Re: [asterisk-users] Asterisk Error

[Steve Totaro]

On Fri, Jul 17, 2009 at 2:08 AM, michel freiha  wrote:

> Hi all,
>
> Can you please let me know what the below issue mean when trying to start
> asterisk and how I can fix it?
>
> pbx_dundi.c: No ethernet interface found for seeding global EID  You will
> have to set it manually.
>
> regards
>

Add:
noload = dundi
To your modules.conf.  That should fix it.

Do you want to use dundi?  What does ifconfig say?

I assume you have a NIC?  Is it up and all that when you start Asterisk?
Have you tried downing it, setting all the variables (maybe even the MAC to
be thorough) and then bringing it back up before starting Asterisk?

Otherwise what kind of NIC?  Do you have an old 3Com laying around you can
pop in it?

Open a bug report?

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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[asterisk-users] Friday reminder

[randulo]

Please join us today at 9AM PDT, 12 Noon EDT for the VoIP Users
Conference to talk about the latest news and events in the wonderful
world of VoIP.

IRC #voip-users-conference

SIP 7463#2262...@proxy.ideasip.com for g711

SIP 200...@login.zipdx.com (for g722 wideband-capable devices)

See http://VUC.me for more details

/r

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Re: [asterisk-users] Skill based routing

[Julian Lyndon-Smith]

Um, I really don't know - we just use the periodic messages to play
the traditional "Your call is important to use" (whatever the
wording..)

Julian.

2009/7/17 Alex Balashov :
>
> What value do the queue announcements (I am assuming these are pertaining
> to expected hold time, etc.) if there is only one agent?
>
>> We use a queue so that we can have all the benefits of the queue
>> whilst finding an agent : music on hold, periodic announcements etc
>> etc.
>>
>> You are right - with a little more effort we could probably remove the
>> need for the queue. But why would I do that if I can use the queue for
>> the bits I want ;)
>>
>> Julian
>>
>> 2009/7/17 Alex Balashov :
>>>
>>> The simplicity of this approach is elegant, but in that case, why use a
>>> queue?  Why not just perform this logic straight in the dial plan when
>>> processing the received call?
>>>
>>> The benefit of queues arises from their ability to keep state;  they can
>>> retry agents, carry out different ring strategies, etc.  I understood
>>> the
>>> original question to be implicitly about incorporating weights for
>>> skills
>>> into queue or queue-like call distribution mechanisms, since that is how
>>> it is done in call center products.  If the question is simply how to
>>> make
>>> Asterisk consider certain outside information when choosing to whom to
>>> route a call, the answer would be that it is identical to the process
>>> for
>>> embedding any other kind of logic and/or outside data source into call
>>> processing.
>>>
 Another simple way is to add local/foo/n as the only "agent" on the
 queue. In the dialplan for local/foo , interrogate a database for the
 most appropriate "agent" and then call that agent's extension.

 Julian

 2009/7/17 Matt Florell :
> On 7/17/09, Alex Balashov  wrote:
>> Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
>>
>>  > We are trying to implement skill based routing for agents in a
>> support
>>  > centre based on the agent login. Has anyone had any experience
>> with
>> this
>>  > and what was the outcome?
>>
>>
>> It can't really be done using Asterisk queues, unless you want to
>> create
>>  a large number of queues for every relevant skill factor and have
>> agents
>>  join various combinations of these simultaneously--which would take
>>  quite a bit of dial plan and/or AGI logic to pull off.  Plus, that
>>  doesn't scale any given queue beyond one host.
>>
>>  I suggest you look into using FastAGI[1] to simulate the queue
>>  experience by generating hold music and announcements without
>> actually
>>  using Asterisk queues per se.  This is quite possible to do, and,
>> this
>>  allows you to distribute queues across multiple hosts, as well as
>>  distribute calls within those queues by whatever logic you choose.
>>  No
>>  shoehorning--just write it yourself.
>>
>>  -- Alex
>>
>>  [1] Yes, FastAGI.  Not local AGI.  And most especially not in PHP;
>>      contrary to a lot of the info out there, PHP could not possibly
>>      be a less suitable language in which to write AGI scripts.  I
>>      don't know who comes up with these lavish heights of mediocrity.
>
> If you are not looking to write it yourself you could always try
> ViciDial which has skills-based routing built in, and it's free and
> Open Source.
>
> MATT---
>
> ___
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>>>
>>>
>>> --
>>> Alex Balashov
>>> Evariste Systems
>>> Web    : http://www.evaristesys.com/
>>> Tel    : (+1) (678) 954-0670
>>> Direct : (+1) (678) 954-0671
>>> Mobile : (+1) (678) 237-1775
>>>
>>>
>>>
>>> ___
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>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> Alex Balashov
> Evariste Systems
> Web    : http://www.evaristesys.com/
> Tel    : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (678) 23

Re: [asterisk-users] Skill based routing

[Alex Balashov]

What value do the queue announcements (I am assuming these are pertaining
to expected hold time, etc.) if there is only one agent?

> We use a queue so that we can have all the benefits of the queue
> whilst finding an agent : music on hold, periodic announcements etc
> etc.
>
> You are right - with a little more effort we could probably remove the
> need for the queue. But why would I do that if I can use the queue for
> the bits I want ;)
>
> Julian
>
> 2009/7/17 Alex Balashov :
>>
>> The simplicity of this approach is elegant, but in that case, why use a
>> queue?  Why not just perform this logic straight in the dial plan when
>> processing the received call?
>>
>> The benefit of queues arises from their ability to keep state;  they can
>> retry agents, carry out different ring strategies, etc.  I understood
>> the
>> original question to be implicitly about incorporating weights for
>> skills
>> into queue or queue-like call distribution mechanisms, since that is how
>> it is done in call center products.  If the question is simply how to
>> make
>> Asterisk consider certain outside information when choosing to whom to
>> route a call, the answer would be that it is identical to the process
>> for
>> embedding any other kind of logic and/or outside data source into call
>> processing.
>>
>>> Another simple way is to add local/foo/n as the only "agent" on the
>>> queue. In the dialplan for local/foo , interrogate a database for the
>>> most appropriate "agent" and then call that agent's extension.
>>>
>>> Julian
>>>
>>> 2009/7/17 Matt Florell :
 On 7/17/09, Alex Balashov  wrote:
> Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
>
>  > We are trying to implement skill based routing for agents in a
> support
>  > centre based on the agent login. Has anyone had any experience
> with
> this
>  > and what was the outcome?
>
>
> It can't really be done using Asterisk queues, unless you want to
> create
>  a large number of queues for every relevant skill factor and have
> agents
>  join various combinations of these simultaneously--which would take
>  quite a bit of dial plan and/or AGI logic to pull off.  Plus, that
>  doesn't scale any given queue beyond one host.
>
>  I suggest you look into using FastAGI[1] to simulate the queue
>  experience by generating hold music and announcements without
> actually
>  using Asterisk queues per se.  This is quite possible to do, and,
> this
>  allows you to distribute queues across multiple hosts, as well as
>  distribute calls within those queues by whatever logic you choose.
>  No
>  shoehorning--just write it yourself.
>
>  -- Alex
>
>  [1] Yes, FastAGI.  Not local AGI.  And most especially not in PHP;
>  contrary to a lot of the info out there, PHP could not possibly
>  be a less suitable language in which to write AGI scripts.  I
>  don't know who comes up with these lavish heights of mediocrity.

 If you are not looking to write it yourself you could always try
 ViciDial which has skills-based routing built in, and it's free and
 Open Source.

 MATT---

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>>>
>>> ___
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>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> Alex Balashov
>> Evariste Systems
>> Web: http://www.evaristesys.com/
>> Tel: (+1) (678) 954-0670
>> Direct : (+1) (678) 954-0671
>> Mobile : (+1) (678) 237-1775
>>
>>
>>
>> ___
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>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775



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Re: [asterisk-users] Skill based routing

[Julian Lyndon-Smith]

We use a queue so that we can have all the benefits of the queue
whilst finding an agent : music on hold, periodic announcements etc
etc.

You are right - with a little more effort we could probably remove the
need for the queue. But why would I do that if I can use the queue for
the bits I want ;)

Julian

2009/7/17 Alex Balashov :
>
> The simplicity of this approach is elegant, but in that case, why use a
> queue?  Why not just perform this logic straight in the dial plan when
> processing the received call?
>
> The benefit of queues arises from their ability to keep state;  they can
> retry agents, carry out different ring strategies, etc.  I understood the
> original question to be implicitly about incorporating weights for skills
> into queue or queue-like call distribution mechanisms, since that is how
> it is done in call center products.  If the question is simply how to make
> Asterisk consider certain outside information when choosing to whom to
> route a call, the answer would be that it is identical to the process for
> embedding any other kind of logic and/or outside data source into call
> processing.
>
>> Another simple way is to add local/foo/n as the only "agent" on the
>> queue. In the dialplan for local/foo , interrogate a database for the
>> most appropriate "agent" and then call that agent's extension.
>>
>> Julian
>>
>> 2009/7/17 Matt Florell :
>>> On 7/17/09, Alex Balashov  wrote:
 Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:

  > We are trying to implement skill based routing for agents in a
 support
  > centre based on the agent login. Has anyone had any experience with
 this
  > and what was the outcome?


 It can't really be done using Asterisk queues, unless you want to
 create
  a large number of queues for every relevant skill factor and have
 agents
  join various combinations of these simultaneously--which would take
  quite a bit of dial plan and/or AGI logic to pull off.  Plus, that
  doesn't scale any given queue beyond one host.

  I suggest you look into using FastAGI[1] to simulate the queue
  experience by generating hold music and announcements without actually
  using Asterisk queues per se.  This is quite possible to do, and, this
  allows you to distribute queues across multiple hosts, as well as
  distribute calls within those queues by whatever logic you choose.  No
  shoehorning--just write it yourself.

  -- Alex

  [1] Yes, FastAGI.  Not local AGI.  And most especially not in PHP;
      contrary to a lot of the info out there, PHP could not possibly
      be a less suitable language in which to write AGI scripts.  I
      don't know who comes up with these lavish heights of mediocrity.
>>>
>>> If you are not looking to write it yourself you could always try
>>> ViciDial which has skills-based routing built in, and it's free and
>>> Open Source.
>>>
>>> MATT---
>>>
>>> ___
>>> -- 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
>>>
>>
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> Alex Balashov
> Evariste Systems
> Web    : http://www.evaristesys.com/
> Tel    : (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (678) 237-1775
>
>
>
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>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Skill based routing

[Alex Balashov]

The simplicity of this approach is elegant, but in that case, why use a
queue?  Why not just perform this logic straight in the dial plan when
processing the received call?

The benefit of queues arises from their ability to keep state;  they can
retry agents, carry out different ring strategies, etc.  I understood the
original question to be implicitly about incorporating weights for skills
into queue or queue-like call distribution mechanisms, since that is how
it is done in call center products.  If the question is simply how to make
Asterisk consider certain outside information when choosing to whom to
route a call, the answer would be that it is identical to the process for
embedding any other kind of logic and/or outside data source into call
processing.

> Another simple way is to add local/foo/n as the only "agent" on the
> queue. In the dialplan for local/foo , interrogate a database for the
> most appropriate "agent" and then call that agent's extension.
>
> Julian
>
> 2009/7/17 Matt Florell :
>> On 7/17/09, Alex Balashov  wrote:
>>> Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
>>>
>>>  > We are trying to implement skill based routing for agents in a
>>> support
>>>  > centre based on the agent login. Has anyone had any experience with
>>> this
>>>  > and what was the outcome?
>>>
>>>
>>> It can't really be done using Asterisk queues, unless you want to
>>> create
>>>  a large number of queues for every relevant skill factor and have
>>> agents
>>>  join various combinations of these simultaneously--which would take
>>>  quite a bit of dial plan and/or AGI logic to pull off.  Plus, that
>>>  doesn't scale any given queue beyond one host.
>>>
>>>  I suggest you look into using FastAGI[1] to simulate the queue
>>>  experience by generating hold music and announcements without actually
>>>  using Asterisk queues per se.  This is quite possible to do, and, this
>>>  allows you to distribute queues across multiple hosts, as well as
>>>  distribute calls within those queues by whatever logic you choose.  No
>>>  shoehorning--just write it yourself.
>>>
>>>  -- Alex
>>>
>>>  [1] Yes, FastAGI.  Not local AGI.  And most especially not in PHP;
>>>  contrary to a lot of the info out there, PHP could not possibly
>>>  be a less suitable language in which to write AGI scripts.  I
>>>  don't know who comes up with these lavish heights of mediocrity.
>>
>> If you are not looking to write it yourself you could always try
>> ViciDial which has skills-based routing built in, and it's free and
>> Open Source.
>>
>> MATT---
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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>


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775



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Re: [asterisk-users] Skill based routing

[Julian Lyndon-Smith]

Another simple way is to add local/foo/n as the only "agent" on the
queue. In the dialplan for local/foo , interrogate a database for the
most appropriate "agent" and then call that agent's extension.

Julian

2009/7/17 Matt Florell :
> On 7/17/09, Alex Balashov  wrote:
>> Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
>>
>>  > We are trying to implement skill based routing for agents in a support
>>  > centre based on the agent login. Has anyone had any experience with this
>>  > and what was the outcome?
>>
>>
>> It can't really be done using Asterisk queues, unless you want to create
>>  a large number of queues for every relevant skill factor and have agents
>>  join various combinations of these simultaneously--which would take
>>  quite a bit of dial plan and/or AGI logic to pull off.  Plus, that
>>  doesn't scale any given queue beyond one host.
>>
>>  I suggest you look into using FastAGI[1] to simulate the queue
>>  experience by generating hold music and announcements without actually
>>  using Asterisk queues per se.  This is quite possible to do, and, this
>>  allows you to distribute queues across multiple hosts, as well as
>>  distribute calls within those queues by whatever logic you choose.  No
>>  shoehorning--just write it yourself.
>>
>>  -- Alex
>>
>>  [1] Yes, FastAGI.  Not local AGI.  And most especially not in PHP;
>>      contrary to a lot of the info out there, PHP could not possibly
>>      be a less suitable language in which to write AGI scripts.  I
>>      don't know who comes up with these lavish heights of mediocrity.
>
> If you are not looking to write it yourself you could always try
> ViciDial which has skills-based routing built in, and it's free and
> Open Source.
>
> MATT---
>
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[asterisk-users] Queue member (Agent) does not Dial

[Kurian Thayil]

Hi All,

We are using Asterisk 1.2.18 in a CentOS box. Implemented a queue
(maqueue) structure for handling customer calls. There are 4 queue members
(85744,85766,85511,84888). These 4 members are logged in using
AgentCallbackLogin application. But at some point, one of the agent's SIP
phone does not ring for an incoming call to this queue. I checked the agent
status and it is not in paused state. When I looked in the CLI, I couldn't
see any attempt by the Asterisk to dial that particular agent. What are the
possiblities for a queue member not dialed by Asterisk? This agent is
defined in agents.conf, member of the queue defined in queues.conf and is
not paused. The output of show agents from CLI is shown below:

8557 (Name1) available at '8...@specagentdial' (musiconhold is
'default')
8545 (Name2) not logged in (musiconhold is 'default')
8555 (Name3) available at '8...@specagentdial' (musiconhold is
'default')
8552 (Name4) not logged in (musiconhold is 'default')
8551 (Name5) not logged in (musiconhold is 'default')
8541 (Name6) not logged in (musiconhold is 'default')
8444 (Name7) not logged in (musiconhold is 'default')
85577(Name8) not logged in (musiconhold is 'default')
85744(Name9) available at '85...@specagentdial' (musiconhold is
'default')
85766(Name10) available at '85...@specagentdial' (musiconhold is
'default')
84888(Name11) available at '84...@specagentdial' (musiconhold is
'default')
85511(Name12) available at '85...@specagentdial' (musiconhold is
'default')

The CLI message is given below:

-- Executing Wait("Zap/1-1", "2") in new stack
-- Executing Answer("Zap/1-1", "") in new stack
-- Executing Playback("Zap/1-1", "Thankyou9800") in new stack
-- Executing Set("Zap/1-1", "editeduid1=1247824046") in new stack
-- Executing Set("Zap/1-1", "editeduid2=897") in new stack
-- Executing Set("Zap/1-1", "editeduid=1247824046-897") in new stack
-- Executing Set("Zap/1-1",
"MONITOR_FILENAME=QMA_20090717-054744_1247824046-897") in new stack
-- Executing AGI("Zap/1-1",
"agi_queue.sh|QMA_20090717-054744_1247824046-897|MAQ") in new stack
-- Executing Queue("Zap/1-1", "maqueue|t|||180") in new stack
-- Executing AGI("Local/84...@specagentdial-14bb,2",
"agi_qdial.sh|84888|315362") in new stack
-- Executing AGI("Local/85...@specagentdial-beba,2",
"agi_qdial.sh|85744|315362") in new stack
-- Executing AGI("Local/85...@specagentdial-67be,2",
"agi_qdial.sh|85511|315362") in new stack

Here, from above, AGI program agi_qdial.sh which handles the dial operation
does not make any attempt to dial 85766. Wondering why this is happening.
The issue gets resolved only when asterisk service is restarted which is not
a pretty good workaround. Any clue on this?

Regards,

Kurian Thayil.
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Re: [asterisk-users] Skill based routing

[Matt Florell]

On 7/17/09, Alex Balashov  wrote:
> Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:
>
>  > We are trying to implement skill based routing for agents in a support
>  > centre based on the agent login. Has anyone had any experience with this
>  > and what was the outcome?
>
>
> It can't really be done using Asterisk queues, unless you want to create
>  a large number of queues for every relevant skill factor and have agents
>  join various combinations of these simultaneously--which would take
>  quite a bit of dial plan and/or AGI logic to pull off.  Plus, that
>  doesn't scale any given queue beyond one host.
>
>  I suggest you look into using FastAGI[1] to simulate the queue
>  experience by generating hold music and announcements without actually
>  using Asterisk queues per se.  This is quite possible to do, and, this
>  allows you to distribute queues across multiple hosts, as well as
>  distribute calls within those queues by whatever logic you choose.  No
>  shoehorning--just write it yourself.
>
>  -- Alex
>
>  [1] Yes, FastAGI.  Not local AGI.  And most especially not in PHP;
>  contrary to a lot of the info out there, PHP could not possibly
>  be a less suitable language in which to write AGI scripts.  I
>  don't know who comes up with these lavish heights of mediocrity.

If you are not looking to write it yourself you could always try
ViciDial which has skills-based routing built in, and it's free and
Open Source.

MATT---

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Re: [asterisk-users] Asterisk and several clients behind NAT

[Alex Balashov]

You're welcome.

What's TAPI?

--
Sent from mobile device

On Jul 17, 2009, at 5:38 AM, jonas kellens   
wrote:

> Thanks Alex for your explanation.
>
> Does this NAT-mapping means that TAPI would also be possible ??
>
> Jonas.
>
> On Tue, 2009-07-14 at 06:33 -0400, Alex Balashov wrote:
>>
>>
>> Yes, this problem has a solution.  The NAT gateway creates a UDP  
>> state
>> mapping between internal source ports and external source (and
>> destination, since most user agents are symmetrical nowadays) ports.
>>
>> The NAT gateway then allocates different external UDP ports for
>> different "connections" being tracked in this manner.
>>
>> Consider, for example, two phones - 192.168.1.10 and 192.168.1.11 -
>> registering to an outside SIP UAS through a NAT gateway whose public
>> address is 67.194.23.55.  The NAT gateway maps the source ports in a
>> random or pseudorandom manner akin to:
>>
>> 192.168.1.10:5060 --> 67.194.23.55:32947
>> 192.168.1.11:5060 --> 67.194.23.55:47948
>>
>> If far-end NAT traversal is enabled on the UAS (in the case of  
>> Asterisk,
>> that's nat=yes in sip.conf), the Contact URI supplied in the REGISTER
>> message is ignored and the actual "received" IP and port on the  
>> network
>> and transport layer is used in its place.  The latter is what is  
>> stored
>> as the contact binding.
>>
>> Later, a call comes in and the UAS maps it back to  
>> 67.194.23.55:47948 or
>> 32947 depending on which registrant it is destined to go to.
>>
>> This scenario is not without its problems.  Some user agents do not
>> behave symmetrically.  Some firewall/NAT router ALGs (application  
>> layer
>> gateways) break this process, though they mean well and try to be
>> helpful.  But by far the most pressing problem is that many NAT  
>> gateways
>> rather quickly age the temporary state information (internal:external
>> UDP port mapping) out after a relatively short period of inactivity.
>> That is why many far-end NAT traversal approaches implement a  
>> policy of
>> periodically "pinging" the stored ("received") contact with some  
>> sort of
>> message that causes a bidirectional exchange of communication, and
>> therefore causes the NAT gateway to reset its expiration timer for  
>> that
>> "connection" state.  In Asterisk, the OPTIONS messages generated when
>> the qualify=yes option is enabled in sip.conf fulfill this function.
>>
>> Hope that helps,
>>
>
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Re: [asterisk-users] Asterisk and several clients behind NAT

[jonas kellens]

Thanks Alex for your explanation.

Does this NAT-mapping means that TAPI would also be possible ??

Jonas.

On Tue, 2009-07-14 at 06:33 -0400, Alex Balashov wrote:

> 
> Yes, this problem has a solution.  The NAT gateway creates a UDP state 
> mapping between internal source ports and external source (and 
> destination, since most user agents are symmetrical nowadays) ports.
> 
> The NAT gateway then allocates different external UDP ports for 
> different "connections" being tracked in this manner.
> 
> Consider, for example, two phones - 192.168.1.10 and 192.168.1.11 - 
> registering to an outside SIP UAS through a NAT gateway whose public 
> address is 67.194.23.55.  The NAT gateway maps the source ports in a 
> random or pseudorandom manner akin to:
> 
> 192.168.1.10:5060 --> 67.194.23.55:32947
> 192.168.1.11:5060 --> 67.194.23.55:47948
> 
> If far-end NAT traversal is enabled on the UAS (in the case of Asterisk, 
> that's nat=yes in sip.conf), the Contact URI supplied in the REGISTER 
> message is ignored and the actual "received" IP and port on the network 
> and transport layer is used in its place.  The latter is what is stored 
> as the contact binding.
> 
> Later, a call comes in and the UAS maps it back to 67.194.23.55:47948 or 
> 32947 depending on which registrant it is destined to go to.
> 
> This scenario is not without its problems.  Some user agents do not 
> behave symmetrically.  Some firewall/NAT router ALGs (application layer 
> gateways) break this process, though they mean well and try to be 
> helpful.  But by far the most pressing problem is that many NAT gateways 
> rather quickly age the temporary state information (internal:external 
> UDP port mapping) out after a relatively short period of inactivity. 
> That is why many far-end NAT traversal approaches implement a policy of 
> periodically "pinging" the stored ("received") contact with some sort of 
> message that causes a bidirectional exchange of communication, and 
> therefore causes the NAT gateway to reset its expiration timer for that 
> "connection" state.  In Asterisk, the OPTIONS messages generated when 
> the qualify=yes option is enabled in sip.conf fulfill this function.
> 
> Hope that helps,
> 
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Re: [asterisk-users] Is Enum safe from spammers?

[Alex Balashov]

IMHO, anonymous calls should never, ever be accepted for a variety of  
reasons. It is naive.

Just because it is convenient does not mean it should be done.

Trusted calls between indeterminate parties can be arranged through  
peering federations, clearinghouses, etc. --> whatever VoIP peering  
model the market ultimately ends up adopting.

--
Sent from mobile device

On Jul 17, 2009, at 5:13 AM, Klaus Darilion  wrote:

>
>
> Gordon Henderson schrieb:
>> Just been contacted by a UK Enum registrar looking for ITSPs to  
>> become
>> resellers of their Enum registration systems ...
>>
>> Is anyone using Enum?
>
> Yes.
>
>> Does anyone (other than cynical old me) think that Enum is a  
>> spammers best
>> friend?
>
> I think ENUM will not cause SPIT, but it can increase the efficiency.
>
>> Has anyone received a spam VoIP call yet? (ie. one placed directly  
>> over
>> the Internet aimed at a SIP URI to a PBX which allows anonymous  
>> incoming
>> calls?)
>
> No.
>
>> I can see that Enum is good to provide another way round the PSTN,  
>> but at
>> the same time, I'm just not convinced...
>>
>> What do others think?
>
>
> SPIT (VoIP SPAM) is basically not a problem of ENUM, but of the
> communication protocol (SIP, H323, IAX, XMPP).
>
> E.g. SIP was developed with the same idea as SMTP: open connectivity -
> everybody can send a message to everyone with the need of peering
> agreements (thus, free of charge). Of course this introduces the same
> problems as SMTP has. Unfortunately the designers of SIP did not
> searched for a solution for this problem. Now, there is SIP-Identity
> which would allow (would, because nobody uses it) authentication of  
> the
> caller - which is the basis for black/whitelists.
>
> H323 and IAX might be different, but they also allow to have
> unauthenticated calls.
>
> So, as soon as you operate your VoIP environment in a "open" way
> (regardless if it is SIP, XMPP ...) you are vulnerable to SPIT -  
> even if
> you do not have ENUM provisioned for your local extensions.
>
> ENUM can be used by crawlers to find out valid VoIP URIs and can help
> SPITting, but in the end the problems is on the SIP level and must be
> solved there.
>
> regards
> klaus
>
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Re: [asterisk-users] PRI hunt group

[Alex Balashov]

Understood--thanks Trevor.  I had wondered if the need to pay per  
channel might somehow amortize the LD balance. Appreciate your  
clarification.

--
Sent from mobile device

On Jul 17, 2009, at 5:14 AM, "Trevor Hammonds"   
wrote:

> On Thursday, July 16, 2009, Alex Balashov wrote:
>
>> C F wrote:
>>
>>> If you don't want to port it to the PRI for whatever reason you can
>>> convert it to a RCFW (remote call forwarded number) which is around
>>> $15.00 plus $8.00 for each additional channel again pricing is for
>>> here in Verizon land.
>>
>> Is that true even if the number is out of a rate center that is  
>> billed
>> long-distance relative to the destination (but still intra-LATA)?   
>> Or do
>> you pay normal LD rates on top of all that in the intra-LATA LD  
>> scenario?
>
> Alex,
> Calls forwarded via Remote Call Forwarding are just like calls  
> forwarded
> from a metered business or residential POTS line.  If the  
> destination to
> which you have selected to forward calls is normally a local call,  
> you will
> just incur the standard metered call rate.  If the call is normally  
> a local
> toll charge (within the same LATA), you will incur toll charges from  
> the
> LEC.  If the call is long distance, you will need to select an IXC  
> -- who
> will bill just as if the calls were made from a POTS line.
>
> Sincerely,
> Trevor Hammonds
>
>
>
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Re: [asterisk-users] T38 negotiation, the last step !

[Klaus Darilion]

Xavier Cardil schrieb:
> Hi, I've managed to get HYLAFAX>T38MODEM->
> ASTERISK>CISCOAS5400 working, but when they are negotiating asterisk 
> drops a message telling "Unknown RTP codec 96 received from gateway" Do 
> somebody know how to fix it ? 
> 
> Thank you !
> 
> 
> 
> << [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-600bfcc8]
> << [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/GWCISCO5400O-600bfcc8]
> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
> << [ TYPE: Control (4) SUBCLASS: T38/Negotiation Requested (19) ] 
> [SIP/GWCISCO5400O-600bfcc8]
> << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8]
> [Jul 16 17:50:39] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP 
> codec 96 received from '192.168.3.163'

Is '192.168.3.163' the Cisco GW? If yes, then it is probably an NSE 
packet. NSE is a Cisco proprietary FaxoverIP solution and uses per 
default payload types 96 and 97 to signal a changeover from VoIP to FoIP.

Probably you have to configure the Cisco GW to use T.38 instead of NSE 
for FoIP.

regards
klaus



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Re: [asterisk-users] PRI hunt group

[Trevor Hammonds]

On Thursday, July 16, 2009, Alex Balashov wrote:

>C F wrote:
>
>> If you don't want to port it to the PRI for whatever reason you can
>> convert it to a RCFW (remote call forwarded number) which is around
>> $15.00 plus $8.00 for each additional channel again pricing is for
>> here in Verizon land.
>
>Is that true even if the number is out of a rate center that is billed 
>long-distance relative to the destination (but still intra-LATA)?  Or do 
>you pay normal LD rates on top of all that in the intra-LATA LD scenario?

Alex,
Calls forwarded via Remote Call Forwarding are just like calls forwarded
from a metered business or residential POTS line.  If the destination to
which you have selected to forward calls is normally a local call, you will
just incur the standard metered call rate.  If the call is normally a local
toll charge (within the same LATA), you will incur toll charges from the
LEC.  If the call is long distance, you will need to select an IXC -- who
will bill just as if the calls were made from a POTS line.  

Sincerely,
Trevor Hammonds



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Re: [asterisk-users] Is Enum safe from spammers?

[Klaus Darilion]

Gordon Henderson schrieb:
> Just been contacted by a UK Enum registrar looking for ITSPs to become 
> resellers of their Enum registration systems ...
> 
> Is anyone using Enum?

Yes.

> Does anyone (other than cynical old me) think that Enum is a spammers best 
> friend?

I think ENUM will not cause SPIT, but it can increase the efficiency.

> Has anyone received a spam VoIP call yet? (ie. one placed directly over 
> the Internet aimed at a SIP URI to a PBX which allows anonymous incoming 
> calls?)

No.

> I can see that Enum is good to provide another way round the PSTN, but at 
> the same time, I'm just not convinced...
> 
> What do others think?


SPIT (VoIP SPAM) is basically not a problem of ENUM, but of the 
communication protocol (SIP, H323, IAX, XMPP).

E.g. SIP was developed with the same idea as SMTP: open connectivity - 
everybody can send a message to everyone with the need of peering 
agreements (thus, free of charge). Of course this introduces the same 
problems as SMTP has. Unfortunately the designers of SIP did not 
searched for a solution for this problem. Now, there is SIP-Identity 
which would allow (would, because nobody uses it) authentication of the 
caller - which is the basis for black/whitelists.

H323 and IAX might be different, but they also allow to have 
unauthenticated calls.

So, as soon as you operate your VoIP environment in a "open" way 
(regardless if it is SIP, XMPP ...) you are vulnerable to SPIT - even if 
you do not have ENUM provisioned for your local extensions.

ENUM can be used by crawlers to find out valid VoIP URIs and can help 
SPITting, but in the end the problems is on the SIP level and must be 
solved there.

regards
klaus

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[asterisk-users] dialplan number matching

[Vieri]

Hi,

How can I match an extension "ending with 3" (just an example but applicable to 
any other digit, including * or #)?

exten => _ZX.3,n,...

exten => _ZX.#,n,...

(the above does not work)

Can regular expressions be used in the standard dialplan (end with: "$")?

Thanks,

Vieri



  

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Re: [asterisk-users] Iphone setup

[Administrator TOOTAI]

James Noble a écrit :
> Thank you for the heads up.  I will look into both weephone and voipover3g
>   
I think siax -from cydia- could also be an alternative as they stated to 
use natively 3g. I only test WIFI.


-- 
Daniel

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[asterisk-users] How do I create an IVR/Dial Group that works properly?

[Alan Lord (News)]

Hi all,

I am trying to understand how I can get a simple IVR scenario to work 
properly (having already removed most of my hair...).

The basic requirement is as follows:

* Caller arrives at our main number
* Caller is greeted and then told they can enter an extension number, if 
known, or wait and their call will be connected to an available rep.
* The IVR then dials a group of extensions (if the caller didn't enter 
one obviously).
* Someone picks up the call and the connection is established and logged.

Now, I have all of this working apart from the last piece.

My IVR rings various extensions and I can pick up the call just fine. 
But my problem is that the data asterisk records regarding the call is 
wrong.

It correctly identifies the CallerID, but it always records the 
destination as "s". Not the extension of, for example my SIP phone (101).

If the incoming caller dials 101 whilst in the IVR, the log is correct.

I can see *why* I am having this problem (There is no extension when you 
arrive in the IVR other than "s"), but I cannot see *how* to fix it.

Please can I ask how do others handle this so it works properly (I've 
included the basics of my DP below)?

I'm running Asterisk 1.4.21.2~dfsg-1ubuntu3 on Ubuntu Server 8.10.

Thanks

Alan


Here is the IVR which callers are dropped into:

[tolc_menu] ; Welcome and information to callers
exten => s,1,Answer()
exten => s,n,Wait(2)
exten => s,n,Background(welcome-to-tolc) ; Say Hello
exten => s,n,Wait(1)
exten => s,n(tryagain),Background(enter-ext-of-person&or) ; Enter
extension number if known, or
exten => s,n,Background(pls-stay-on-line) ; Trying to connect...
exten => s,n,WaitExten(5)
exten => s,n,Macro(belllord,${ALANL}&${ALANB},303)

exten => _10[1-5],1,Macro(call_extension,SIP/${EXTEN})

exten => _20[1-5],1,Macro(call_extension,IAX2/alanb/${EXTEN})


The Vars ALANL and ALANB are:
ALANL=SIP/101
ALANB=IAX2/alanb/202


Here is the Macro belllord:

[macro-belllord]
exten => s,1,Dial(${ARG1},20,t)
exten => s,n,Goto(s-${DIALSTATUS},1)

exten => s-NOANSWER,1,Voicemail(${ar...@business,u) ; business is the
voicemail context, ${ARG2} is the mailbox number to dial
exten => s-NOANSWER,n,Hangup()

exten => s-BUSY,1,Voicemail(${ar...@business,b)
exten => s-BUSY,n,Hangup()

exten => _s-.,1,Goto(s-NOANSWER,1)


Here is the call-extension Macro:

[macro-call_extension]
exten => s,1,Dial(${ARG1},20,t) ; Ring channel for up to 20s
exten => s,n,Goto(s-${DIALSTATUS},1) ; Go to either no answer or busy.

exten => s-NOANSWER,1,Voicemail(${macro_ext...@garden_house,u)

exten => s-BUSY,1,Voicemail(${macro_ext...@garden_house,b)

exten => _s-.,1,Goto(s-NOANSWER,1)



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Re: [asterisk-users] Asterisk Error

[Alex Balashov]

I would guess that the MAC address of an Ethernet adaptor is used as a 
seed for a pseudorandom number generation algorithm that is used to 
create a GUID (Globally Unique Identifier) for your DUNDI node.

But that requires an Ethernet adaptor.

Ali Jawad wrote:

> This means that no ethernet interface is found for seeding the global
> EID. So you will have to set it manually.
> 
> :) Pretty clear.
> 
> On Thu, Jul 16, 2009 at 11:08 PM, michel freiha wrote:
>> Hi all,
>>
>> Can you please let me know what the below issue mean when trying to start
>> asterisk and how I can fix it?
>>
>> pbx_dundi.c: No ethernet interface found for seeding global EID  You will
>> have to set it manually.
>>
>> regards
>>
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-- 
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [asterisk-users] Asterisk Error

[Ali Jawad]

This means that no ethernet interface is found for seeding the global
EID. So you will have to set it manually.

:) Pretty clear.

On Thu, Jul 16, 2009 at 11:08 PM, michel freiha wrote:
> Hi all,
>
> Can you please let me know what the below issue mean when trying to start
> asterisk and how I can fix it?
>
> pbx_dundi.c: No ethernet interface found for seeding global EID  You will
> have to set it manually.
>
> regards
>
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Re: [asterisk-users] Skill based routing

[Alex Balashov]

Rupert Utteridge - Digital Techniques (Austalia) Limited wrote:

> We are trying to implement skill based routing for agents in a support 
> centre based on the agent login. Has anyone had any experience with this 
> and what was the outcome?

It can't really be done using Asterisk queues, unless you want to create 
a large number of queues for every relevant skill factor and have agents 
join various combinations of these simultaneously--which would take 
quite a bit of dial plan and/or AGI logic to pull off.  Plus, that 
doesn't scale any given queue beyond one host.

I suggest you look into using FastAGI[1] to simulate the queue 
experience by generating hold music and announcements without actually 
using Asterisk queues per se.  This is quite possible to do, and, this 
allows you to distribute queues across multiple hosts, as well as 
distribute calls within those queues by whatever logic you choose.  No 
shoehorning--just write it yourself.

-- Alex

[1] Yes, FastAGI.  Not local AGI.  And most especially not in PHP;
 contrary to a lot of the info out there, PHP could not possibly
 be a less suitable language in which to write AGI scripts.  I
 don't know who comes up with these lavish heights of mediocrity.

-- 
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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