[asterisk-users] Testing

[Joshua C. Colp]

This is just a test of the asterisk-users mailing list.

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Testing for real from a non-digium email

[Matthew Fredrickson]

Here we go!

Matt

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

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Testing...

[Matt Fredrickson]

Test from non-digium email.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] testing users list

[Matt Ball]

testing users list
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] Testing 911 call

[David Wessell]

Quite a few SIP providers will have 911 testing functionality. Our main 911 
provider lets you dial 933. Than they read back to you the address information 
that is transmitted with the 911 call.

--
Ringfree Communications
David Wessell
828-575-0030 x101

From: James Miller mailto:paramedi...@gmail.com>>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com>>
Date: Sunday, May 5, 2013 8:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com>>
Subject: Re: [asterisk-users] Testing 911 call

I actually work in a 911 center.  Please do not dial blindly to do a test call. 
 Please call the non-emergency dispatch number, ask if it would be ok to make 
one or two test calls.  If they give you the ok, please complete those calls as 
quickly as possible as conditions change in an instant.  If they give you 
permission at 9am, don't wait until 5pm to do the test.  Further, most 911/PSAP 
centers are not busy after 10pm local time, please consider doing your testing 
in the overnight hours.  Again, please check with your local 911/PSAP to 
confirm when their peak times are and try to avoid them.

Here is a good script to read when speaking to the 911 call taker:

Hello my name is (your name) with (company name).  We are performing a test 911 
call and would like to confirm some information if you have a moment.  (if they 
answer go ahead, continue with the script.  If they advise now is not a good 
time, say thank you and disconnect.  In a 30 to 60 minutes call the 
non-emergency number and ask if you may make another test call)

(continuing the script)  Can you please confirm the address that shows up on 
your phone system please? (wait and confirm the info)  Great thank you.  If you 
can, please tell me the number you show we are calling from? (wait and confirm) 
 Can you confirm for me which 911/PSAP center we have reached? ( wait and 
confirm this is the proper 911/PSAP you need)  (if this is the first of several 
calls:)  Thank you very much for your time, we will be making (xx number) of 
calls in the next few minutes.  (if this is the end of your testing:)  Thank 
you very much for your time, this concludes our testing.


Good luck with your phone testing.

Regards,
James

"I see blindness, not as a disability, but more of an ability.  And Sight 
actually, more of a disability because some people with sight tend to judge 
others by what they see on the outside, whereas I don't see that. I just see 
that which is in a person."  Patrick Henry Hughes, Louisville Kentucky,2008


On Sun, May 5, 2013 at 8:00 PM, Dale Noll 
mailto:dn...@wi.rr.com>> wrote:
If there is a non-emergency number you can call and let them know you would 
like to do some test calls. This also allows you to schedule a time for testing 
when the PSAP is not as busy allowing for real calls to be handled.


On Sun, May 5, 2013 at 11:15 AM, Mark Engelhardt 
mailto:ma...@intuitiveengineering.com>> wrote:
Joseph,

I have made a quite a few test calls to 911.  They don't charge you and they 
don't get upset.

Just let them know right away it is a non-emergency test call, and then let 
them know who you are and what you need to verify on their information screen.

Mark Engelhardt


On May 5, 2013, at 11:07 AM, Joseph wrote:

> How to test 911 call?
>
> I'm using Audiocodes and it setup to strip the first number but I've never 
> tested the 911 call.  I don't want to go live as they might charge me.
>
> --
> Joseph

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing 911 call

[James Miller]

I actually work in a 911 center.  Please do not dial blindly to do a test
call.  Please call the non-emergency dispatch number, ask if it would be ok
to make one or two test calls.  If they give you the ok, please complete
those calls as quickly as possible as conditions change in an instant.  If
they give you permission at 9am, don't wait until 5pm to do the test.
 Further, most 911/PSAP centers are not busy after 10pm local time, please
consider doing your testing in the overnight hours.  Again, please check
with your local 911/PSAP to confirm when their peak times are and try to
avoid them.

Here is a good script to read when speaking to the 911 call taker:

Hello my name is (your name) with (company name).  We are performing a test
911 call and would like to confirm some information if you have a moment.
 (if they answer go ahead, continue with the script.  If they advise now is
not a good time, say thank you and disconnect.  In a 30 to 60 minutes call
the non-emergency number and ask if you may make another test call)

(continuing the script)  Can you please confirm the address that shows up
on your phone system please? (wait and confirm the info)  Great thank you.
 If you can, please tell me the number you show we are calling from? (wait
and confirm)  Can you confirm for me which 911/PSAP center we have reached?
( wait and confirm this is the proper 911/PSAP you need)  (if this is the
first of several calls:)  Thank you very much for your time, we will be
making (xx number) of calls in the next few minutes.  (if this is the end
of your testing:)  Thank you very much for your time, this concludes our
testing.


Good luck with your phone testing.

Regards,
James

"I see blindness, not as a disability, but more of an ability.  And Sight
actually, more of a disability because some people with sight tend to judge
others by what they see on the outside, whereas I don't see that. I just
see that which is in a person."  Patrick Henry Hughes, Louisville
Kentucky,2008


On Sun, May 5, 2013 at 8:00 PM, Dale Noll  wrote:

> If there is a non-emergency number you can call and let them know you
> would like to do some test calls. This also allows you to schedule a time
> for testing when the PSAP is not as busy allowing for real calls to be
> handled.
>
>
> On Sun, May 5, 2013 at 11:15 AM, Mark Engelhardt <
> ma...@intuitiveengineering.com> wrote:
>
>> Joseph,
>>
>> I have made a quite a few test calls to 911.  They don't charge you and
>> they don't get upset.
>>
>> Just let them know right away it is a non-emergency test call, and then
>> let them know who you are and what you need to verify on their information
>> screen.
>>
>> Mark Engelhardt
>>
>>
>> On May 5, 2013, at 11:07 AM, Joseph wrote:
>>
>> > How to test 911 call?
>> >
>> > I'm using Audiocodes and it setup to strip the first number but I've
>> never tested the 911 call.  I don't want to go live as they might charge me.
>> >
>> > --
>> > Joseph
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing 911 call

[Dale Noll]

If there is a non-emergency number you can call and let them know you would
like to do some test calls. This also allows you to schedule a time for
testing when the PSAP is not as busy allowing for real calls to be handled.


On Sun, May 5, 2013 at 11:15 AM, Mark Engelhardt <
ma...@intuitiveengineering.com> wrote:

> Joseph,
>
> I have made a quite a few test calls to 911.  They don't charge you and
> they don't get upset.
>
> Just let them know right away it is a non-emergency test call, and then
> let them know who you are and what you need to verify on their information
> screen.
>
> Mark Engelhardt
>
>
> On May 5, 2013, at 11:07 AM, Joseph wrote:
>
> > How to test 911 call?
> >
> > I'm using Audiocodes and it setup to strip the first number but I've
> never tested the 911 call.  I don't want to go live as they might charge me.
> >
> > --
> > Joseph
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing 911 call

[Mark Engelhardt]

Joseph,

I have made a quite a few test calls to 911.  They don't charge you and they 
don't get upset. 

Just let them know right away it is a non-emergency test call, and then let 
them know who you are and what you need to verify on their information screen. 

Mark Engelhardt


On May 5, 2013, at 11:07 AM, Joseph wrote:

> How to test 911 call?
> 
> I'm using Audiocodes and it setup to strip the first number but I've never 
> tested the 911 call.  I don't want to go live as they might charge me.
> 
> -- 
> Joseph

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] Testing 911 call

[Joseph]

How to test 911 call?

I'm using Audiocodes and it setup to strip the first number but I've never tested the 911 call.  
I don't want to go live as they might charge me.


--
Joseph

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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

[asterisk-users] testing asterisk11 on single machine

[alok srivastava]

can i test my asterisk11 on a single machine on which asterisk is installed
that sounds are working from both end properly.
i have installed asterisk 11 on  ubuntu12.04 with twinkle soft phone.
regards
abhi
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing for media?

[Arstan Jusupov]

tcpdump and wireshark would help I guess. Just sniff for sip traffic and look 
out for what's happening there. My 2 cents

Sent from my iPhone

On May 19, 2012, at 8:33 PM, David Wessell  wrote:

> I'm in the process of setting up an asterisk box that will stand
> between PBX's and the SIP providers. So a trunking server.
> 
> How can I 'test' to see if this trunking server is stepping out of the
> media path during calls?
> 
> Thanks
> David
> 
> -- 
> --
> www.ringfree.biz
> 828-575-0030
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] Testing for media?

[David Wessell]

I'm in the process of setting up an asterisk box that will stand
between PBX's and the SIP providers. So a trunking server.

How can I 'test' to see if this trunking server is stepping out of the
media path during calls?

Thanks
David

-- 
--
www.ringfree.biz
828-575-0030

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] testing simultaneous calls

[Stefan Schmidt]

Am 15.09.2011 21:18, schrieb ERIC HERRON:
>  
> 
> Asterisk 1.4.26 keeps randomly crashing then restarting itself on my
> live production.
> 
>  
> 
> I cannot run valgrind and I do not have the right flags set in menuselect.
> 
>  
> 
> I can however at the dead of the night run stress tests.
> 
>  
> 
> I want to simulate x-amount of concurrent calls to both a dtmf dialplan,
> which is working, as well as MoH dialplan to see if this could be the
> cause of crashing.
> 
>  
> 
> How do I test this?
> 
> Is it a call file that can handle this without ringing my extension
> first, like internal system calling?
> 
>  
> 
> Thanks,
> 
> --Eric

Hi eric,

Take a look at sipp.

with this you can easily generate sip calls against your asterisk server
up to several hundred calls per second.

call files would not generate that much load.

best regards

stefan

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] testing simultaneous calls

[Sam Govind]

A little  look at the dialplan which rings your extension, or get dtmf, and
plays DTMF will help better understand. btw you can set the
context/extension/priority in a call file to skip some priorities of a
particular extension set.

On Fri, Sep 16, 2011 at 12:18 AM, ERIC HERRON  wrote:

> ** **
>
> Asterisk 1.4.26 keeps randomly crashing then restarting itself on my live
> production.
>
> ** **
>
> I cannot run valgrind and I do not have the right flags set in menuselect.
> 
>
> ** **
>
> I can however at the dead of the night run stress tests.
>
> ** **
>
> I want to simulate x-amount of concurrent calls to both a dtmf dialplan,
> which is working, as well as MoH dialplan to see if this could be the cause
> of crashing.
>
> ** **
>
> How do I test this?
>
> Is it a call file that can handle this without ringing my extension first,
> like internal system calling?
>
> ** **
>
> Thanks,
>
> --Eric
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] testing simultaneous calls

[ERIC HERRON]

 

Asterisk 1.4.26 keeps randomly crashing then restarting itself on my live
production.

 

I cannot run valgrind and I do not have the right flags set in menuselect.

 

I can however at the dead of the night run stress tests.

 

I want to simulate x-amount of concurrent calls to both a dtmf dialplan,
which is working, as well as MoH dialplan to see if this could be the cause
of crashing.

 

How do I test this?

Is it a call file that can handle this without ringing my extension first,
like internal system calling?

 

Thanks,

--Eric

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing Asterisk with media - sipp

[Daniel - Asterisk]

Thank you Alex,

It's running without errors now and I can see the media flowing with
'rtp set debug on' but I can't still hear anything on the Asterisk's
peers, any advice?

Elder

2011/7/4, Alex Balashov :
> 488 means no mutually acceptable codecs were negotiated between the
> endpoints.
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
> On Jul 4, 2011, at 3:29 PM, Daniel - Asterisk  wrote:
>
>> I'm trying to get working SIPp with media but something is wrong (it's
>> working well without media), please help:
>>
>> This is the command I send at SIPp server:
>>   ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err
>>
>> This is the result I see:
>>   Last Error: Aborting call on unexpected message for Call-Id
>> '19-12768@12...
>>
>> What I see at sipp's logs:
>>
>> 2011-06-28  14:32:57:6241309289577.624809: Aborting call on
>> unexpected message for Call-Id '1-12768@127.0.0.1': while expecting '100'
>> (index 1), received 'SIP/2.0 488 Not acceptable here
>>
>> Via: SIP/2.0/UDP
>> 127.0.0.1:5061;branch=z9hG4bK-12768-1-0;received=192.168.1.253
>> From: sipp ;tag=12768SIPpTag091
>> To: sut ;tag=as3614adc3
>> Call-ID: 1-12768@127.0.0.1
>> CSeq: 1 INVITE
>> Server: Asterisk PBX 1.8.4.1
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
>> PUBLISH
>> Supported: replaces, timer
>> Content-Length: 0
>>
>> This is my asterisk 1.8's configuration:
>>
>> sip.conf
>> [sipp]
>> type=friend
>> context=sipp
>> host=dynamic
>> port=6000
>> user=sipp
>> canreinvite=no
>> disallow=all
>> allow=ulaw
>>
>> extensions.conf:
>> [sipp]
>> exten => 2005,1,Answer
>> same=>n,Dial(SIP/intern,30)
>> same=>n,Hangup()
>>
>> exten => 2006,1,Answer()
>> same=> n,WaitMusicOnHold(4)
>> same=> n,Hangup()
>>
>>
>> I'm using sipp.3.1.src.tar.gz and I have installed it this way:
>> ..sip.svn# make pcapplay
>>
>> Thanks in advance.
>>
>> Elder
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

-- 
Enviado desde mi dispositivo móvil

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing Asterisk with media - sipp

[Alex Balashov]

488 means no mutually acceptable codecs were negotiated between the endpoints.

--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

On Jul 4, 2011, at 3:29 PM, Daniel - Asterisk  wrote:

> I'm trying to get working SIPp with media but something is wrong (it's 
> working well without media), please help:
> 
> This is the command I send at SIPp server: 
>   ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err
> 
> This is the result I see:
>   Last Error: Aborting call on unexpected message for Call-Id 
> '19-12768@12...
> 
> What I see at sipp's logs:
> 
> 2011-06-28  14:32:57:6241309289577.624809: Aborting call on 
> unexpected message for Call-Id '1-12768@127.0.0.1': while expecting '100' 
> (index 1), received 'SIP/2.0 488 Not acceptable here
> 
> Via: SIP/2.0/UDP 
> 127.0.0.1:5061;branch=z9hG4bK-12768-1-0;received=192.168.1.253
> From: sipp ;tag=12768SIPpTag091
> To: sut ;tag=as3614adc3
> Call-ID: 1-12768@127.0.0.1
> CSeq: 1 INVITE
> Server: Asterisk PBX 1.8.4.1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
> PUBLISH
> Supported: replaces, timer
> Content-Length: 0
> 
> This is my asterisk 1.8's configuration:
> 
> sip.conf
> [sipp]
> type=friend
> context=sipp
> host=dynamic
> port=6000
> user=sipp
> canreinvite=no
> disallow=all
> allow=ulaw
> 
> extensions.conf:
> [sipp]
> exten => 2005,1,Answer
> same=>n,Dial(SIP/intern,30)
> same=>n,Hangup()
> 
> exten => 2006,1,Answer()
> same=> n,WaitMusicOnHold(4)
> same=> n,Hangup()
> 
> 
> I'm using sipp.3.1.src.tar.gz and I have installed it this way:
> ..sip.svn# make pcapplay
> 
> Thanks in advance.
> 
> Elder
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] Testing Asterisk with media - sipp

[Daniel - Asterisk]

I'm trying to get working SIPp with media but something is wrong (it's
working well without media), please help:

This is the command I send at SIPp server:
  ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err

This is the result I see:
  Last Error: Aborting call on unexpected message for Call-Id
'19-12768@12...

What I see at sipp's logs:

2011-06-28  14:32:57:6241309289577.624809: Aborting call on
unexpected message for Call-Id '1-12768@127.0.0.1': while expecting '100'
(index 1), received 'SIP/2.0 488 Not acceptable here

Via: SIP/2.0/UDP 127.0.0.1:5061
;branch=z9hG4bK-12768-1-0;received=192.168.1.253
From: sipp ;tag=12768SIPpTag091
To: sut ;tag=as3614adc3
Call-ID: 1-12768@127.0.0.1
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0

This is my asterisk 1.8's configuration:

*sip.conf*
[sipp]
type=friend
context=sipp
host=dynamic
port=6000
user=sipp
canreinvite=no
disallow=all
allow=ulaw
*
*
*extensions.conf:*
[sipp]
exten => 2005,1,Answer
same=>n,Dial(SIP/intern,30)
same=>n,Hangup()

exten => 2006,1,Answer()
same=> n,WaitMusicOnHold(4)
same=> n,Hangup()


I'm using sipp.3.1.src.tar.gz and I have installed it this way:
..sip.svn# make pcapplay

Thanks in advance.

Elder
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing from where number is...

[Benny Amorsen]

Piotr Górski  writes:

> So how to bill customers? Number portability makes it pretty impossible...

In the US, you pay the same to call a cell phone as you pay to call any
other phone. The callee pays for the airtime. This is a sensible
arrangement, as it allows for number portability and price competition.

Alas, Europe chose to pass the costs onto the caller, without even
making it reasonably possible for the caller to know whether he is
calling a cell phone or not! The Danish number plan in particular is
completely insane.


/Benny


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing from where number is...

[A J Stiles]

On Thursday 03 Mar 2011, Piotr Górski wrote:
> As free I mean no subscription. I can write AGI that will query
> numberingplans.com - that's not a problem... but I can query site only 20
> times a day without a subscription... So it's not free.

Well, free is as free does  :)

For the time being, keep making your 20 free queries per IP address per day, 
and build up a local MySQL database.  Populate it also from any other data 
sources you have available  (maybe you have letters with addresses and phone 
numbers? .)  Then have your AGI script always look in the local database 
first.  If what you need is not in there, and you still have some free 
queries remaining today  (even this information can be held within the 
database),  query numberingplans.com and save the result in your database.  
If you have run out of free queries, then you'll have to return something 
less precise  (just a country, perhaps; this information at least should be 
in your phone directory).

I can tell you now for free that 44 is the code for the UK; and UK numbers 
beginning with (0)7 are mobiles, (0)20 is London and (0)28 is Northern 
Ireland  :)

-- 
AJS

Answers come *after* questions.

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing from where number is...

[Andrew Joakimsen]

2011/3/3 Piotr Górski :
> Something free?

If your provider provides a proper rate table you will pretty much
know which is mobile and which is fixed line and assuming their
rates are accurate I assume your company wouldn't care if you allowed
the mobiles billed at fixed line rates.

-- 
Med Vennlig Hilsen,

A. Helge Joakimsen

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing from where number is...

[Paul Belanger]

On 11-03-03 03:20 PM, bayardo.sanc...@gmail.com wrote:
> I listened to your email using DriveCarefully and will respond as soon as I 
> can.
>  Download DriveCarefully for free at www.drivecarefully.com
>
Please disable your auto responder.

-- 
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing from where number is...

[Piotr Górski]

So how to bill customers? Number portability makes it pretty impossible...

W dniu 3 marca 2011 22:09 użytkownik jon pounder  napisał:

>  On 03/03/2011 03:53 PM, Danny Nicholas wrote:
>
>  Not having an in-depth knowledge of how EU numbering works, I would still
> suggest that you could get pretty far with the numberingplans AGI if you
> made a database that blocked out the number once it came up as a cell.  In
> the U.S. cell phones have their own "local" code, IE 205616 is going to
> be a cell so you can eliminate 205-616- thru 205-616-.
>
>
>
> with number portability in north america, even knowing the npa npx doesn't
> really help you, you can port a landline to a cell or vice versa, you can
> tell what carrier/media the number was originally allocated from but if it
> was ever ported that information is meaningless.
>
> Granted not a lot of people port numbers since you still have to argue to
> get it done around here (in Canada) but its been the law for several years
> if its still a local exchange the porting costs $30 and they have to do it
> if you ask.
>
>
>
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [
> mailto:asterisk-users-boun...@lists.digium.com]
> *On Behalf Of *Piotr Górski
> *Sent:* Thursday, March 03, 2011 2:34 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Testing from where number is...
>
>
>
> As free I mean no subscription. I can write AGI that will query
> numberingplans.com - that's not a problem... but I can query site only 20
> times a day without a subscription... So it's not free.
>
>
>
>
> Z poważaniem,
> Piotr Górski,
> CONCEPT MUSIC ART SP. Z O.O.
> ul.Dauna 70
> 30-629 Kraków
> http://www.cma.pl
> pi...@prnet.pl
> GSM: +48 609 127 370
> Faks: +48 12 444 1051
>
>  2011/3/3 Danny Nicholas 
>
> To "beat the dead horse", if you want it "free", write an AGI...
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Piotr Górski
> *Sent:* Thursday, March 03, 2011 1:58 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Testing from where number is...
>
>
>
> Something free?
>
>
>
> --
>
> Piotrek Gorski
>
> 2011/3/3 Faisal Hanif 
>
> www.numberingplans.com
>
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Piotr Górski
> Sent: Thursday, March 03, 2011 12:02 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Testing from where number is...
>
> Hi!
>
> My customer want's to allow calls to landlines in EU and US and disallow
> calls to cells in EU. Rest of countries are blocked.
>
> Country blocking is easy... Is there a service that allows checking phone
> number? Maybe some specific Enum? I ask for number and server responds with
> info, for example: "Cell Phone, Belgium" or "Land Line, Germany".
>
> --
> Piotr Gorski
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
> to
> Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introduc

Re: [asterisk-users] Testing from where number is...

[jon pounder]

On 03/03/2011 03:53 PM, Danny Nicholas wrote:


Not having an in-depth knowledge of how EU numbering works, I would 
still suggest that you could get pretty far with the numberingplans 
AGI if you made a database that blocked out the number once it came up 
as a cell.  In the U.S. cell phones have their own "local" code, IE 
205616 is going to be a cell so you can eliminate 205-616- 
thru 205-616-.





with number portability in north america, even knowing the npa npx 
doesn't really help you, you can port a landline to a cell or vice 
versa, you can tell what carrier/media the number was originally 
allocated from but if it was ever ported that information is meaningless.


Granted not a lot of people port numbers since you still have to argue 
to get it done around here (in Canada) but its been the law for several 
years if its still a local exchange the porting costs $30 and they have 
to do it if you ask.







*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Piotr 
Górski

*Sent:* Thursday, March 03, 2011 2:34 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Testing from where number is...

As free I mean no subscription. I can write AGI that will query 
numberingplans.com <http://numberingplans.com> - that's not a 
problem... but I can query site only 20 times a day without a 
subscription... So it's not free.



Z poważaniem,
Piotr Górski,
CONCEPT MUSIC ART SP. Z O.O.
ul.Dauna 70
30-629 Kraków
http://www.cma.pl
pi...@prnet.pl <mailto:pi...@prnet.pl>
GSM: +48 609 127 370
Faks: +48 12 444 1051

2011/3/3 Danny Nicholas mailto:da...@debsinc.com>>

To "beat the dead horse", if you want it "free", write an AGI...



*From:*asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com> 
[mailto:asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com>] *On Behalf Of *Piotr 
Górski

*Sent:* Thursday, March 03, 2011 1:58 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Testing from where number is...

Something free?

--

Piotrek Gorski

2011/3/3 Faisal Hanif mailto:fai...@vopium.com>>

www.numberingplans.com <http://www.numberingplans.com>




-Original Message-
From: asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com>
[mailto:asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com>] On Behalf Of Piotr 
Górski

Sent: Thursday, March 03, 2011 12:02 PM
To: asterisk-users@lists.digium.com 
<mailto:asterisk-users@lists.digium.com>

Subject: [asterisk-users] Testing from where number is...

Hi!

My customer want's to allow calls to landlines in EU and US and disallow
calls to cells in EU. Rest of countries are blocked.

Country blocking is easy... Is there a service that allows checking phone
number? Maybe some specific Enum? I ask for number and server responds 
with

info, for example: "Cell Phone, Belgium" or "Land Line, Germany".

--
Piotr Gorski

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- 
New to

Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

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


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

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


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

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


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

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


--
_
-- Bandwidth and Colocation P

Re: [asterisk-users] Testing from where number is...

[Steven Howes]

On 3 Mar 2011, at 20:53, Danny Nicholas wrote:
> Not having an in-depth knowledge of how EU numbering works

Sadly there is no 'EU numbering'. Europe isn't a country, thus doesn't share 
any dial plan. There appears to be some tendency towards having a '7' at the 
front of a mobile number, but it's far from universal. Each telco/regulating 
body will probably supply a list of codes. OFCOM in the UK do. Failing that, 
putting the country code (e.g. +44) into google gives a fairly rough outline 
for each country.

S--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing from where number is...

[Danny Nicholas]

Not having an in-depth knowledge of how EU numbering works, I would still
suggest that you could get pretty far with the numberingplans AGI if you
made a database that blocked out the number once it came up as a cell.  In
the U.S. cell phones have their own "local" code, IE 205616 is going to
be a cell so you can eliminate 205-616- thru 205-616-.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Piotr Górski
Sent: Thursday, March 03, 2011 2:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Testing from where number is...

 

As free I mean no subscription. I can write AGI that will query
numberingplans.com - that's not a problem... but I can query site only 20
times a day without a subscription... So it's not free.

 


Z poważaniem,
Piotr Górski,
CONCEPT MUSIC ART SP. Z O.O.
ul.Dauna 70
30-629 Kraków
http://www.cma.pl
pi...@prnet.pl
GSM: +48 609 127 370
Faks: +48 12 444 1051



2011/3/3 Danny Nicholas 

To "beat the dead horse", if you want it "free", write an AGI.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Piotr Górski
Sent: Thursday, March 03, 2011 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Testing from where number is...

 

Something free?

 

-- 

Piotrek Gorski

2011/3/3 Faisal Hanif 

www.numberingplans.com




-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Piotr Górski
Sent: Thursday, March 03, 2011 12:02 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Testing from where number is...

Hi!

My customer want's to allow calls to landlines in EU and US and disallow
calls to cells in EU. Rest of countries are blocked.

Country blocking is easy... Is there a service that allows checking phone
number? Maybe some specific Enum? I ask for number and server responds with
info, for example: "Cell Phone, Belgium" or "Land Line, Germany".

--
Piotr Gorski

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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

 


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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

 

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing from where number is...

[Piotr Górski]

As free I mean no subscription. I can write AGI that will query
numberingplans.com - that's not a problem... but I can query site only 20
times a day without a subscription... So it's not free.


Z poważaniem,
Piotr Górski,
CONCEPT MUSIC ART SP. Z O.O.
ul.Dauna 70
30-629 Kraków
http://www.cma.pl
pi...@prnet.pl
GSM: +48 609 127 370
Faks: +48 12 444 1051


2011/3/3 Danny Nicholas 

>  To "beat the dead horse", if you want it "free", write an AGI...
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Piotr Górski
> *Sent:* Thursday, March 03, 2011 1:58 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Testing from where number is...
>
>
>
> Something free?
>
>
>
> --
>
> Piotrek Gorski
>
> 2011/3/3 Faisal Hanif 
>
> www.numberingplans.com
>
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Piotr Górski
> Sent: Thursday, March 03, 2011 12:02 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Testing from where number is...
>
> Hi!
>
> My customer want's to allow calls to landlines in EU and US and disallow
> calls to cells in EU. Rest of countries are blocked.
>
> Country blocking is easy... Is there a service that allows checking phone
> number? Maybe some specific Enum? I ask for number and server responds with
> info, for example: "Cell Phone, Belgium" or "Land Line, Germany".
>
> --
> Piotr Gorski
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
> to
> Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing from where number is...

[bayardo . sanchez]

I listened to your email using DriveCarefully and will respond as soon as I can.
 Download DriveCarefully for free at www.drivecarefully.com
--
Sent from my BlackBerry®
Senior Support Engineer
US Numbers: 561-886-0664
Nicaragua Mobile: +505.8488.6876
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing from where number is...

[Danny Nicholas]

To “beat the dead horse”, if you want it “free”, write an AGI…

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Piotr Górski
Sent: Thursday, March 03, 2011 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Testing from where number is...

 

Something free?

 

-- 

Piotrek Gorski

2011/3/3 Faisal Hanif 

www.numberingplans.com




-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Piotr Górski
Sent: Thursday, March 03, 2011 12:02 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Testing from where number is...

Hi!

My customer want's to allow calls to landlines in EU and US and disallow
calls to cells in EU. Rest of countries are blocked.

Country blocking is easy... Is there a service that allows checking phone
number? Maybe some specific Enum? I ask for number and server responds with
info, for example: "Cell Phone, Belgium" or "Land Line, Germany".

--
Piotr Gorski

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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

 

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing from where number is...

[Piotr Górski]

Something free?

-- 
Piotrek Gorski

2011/3/3 Faisal Hanif 

> www.numberingplans.com
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Piotr Górski
> Sent: Thursday, March 03, 2011 12:02 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Testing from where number is...
>
> Hi!
>
> My customer want's to allow calls to landlines in EU and US and disallow
> calls to cells in EU. Rest of countries are blocked.
>
> Country blocking is easy... Is there a service that allows checking phone
> number? Maybe some specific Enum? I ask for number and server responds with
> info, for example: "Cell Phone, Belgium" or "Land Line, Germany".
>
> --
> Piotr Gorski
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
> to
> Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing from where number is...

[Faisal Hanif]

www.numberingplans.com



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Piotr Górski
Sent: Thursday, March 03, 2011 12:02 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Testing from where number is...

Hi!

My customer want's to allow calls to landlines in EU and US and disallow
calls to cells in EU. Rest of countries are blocked.

Country blocking is easy... Is there a service that allows checking phone
number? Maybe some specific Enum? I ask for number and server responds with
info, for example: "Cell Phone, Belgium" or "Land Line, Germany".

--
Piotr Gorski

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] Testing from where number is...

[Piotr Górski]

Hi!

My customer want's to allow calls to landlines in EU and US and disallow 
calls to cells in EU. Rest of countries are blocked.


Country blocking is easy... Is there a service that allows checking 
phone number? Maybe some specific Enum? I ask for number and server 
responds with info, for example: "Cell Phone, Belgium" or "Land Line, 
Germany".


--
Piotr Gorski

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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

[asterisk-users] testing my asterisk 1.6.2.8-rc1 with gtalk (and JACK) - please help

[Julien Claassen]

Hello everyone!
   I'm just trying to set up my new asterisk (version 1.6.2.8-rc1). I'd be
very grateful, if someone could help me here. I'd be very glad, if one of you
could test googletalk with me. Last time I tried (in 1.6.0.x times) it
wouldn't work in the end.
   But here are my gtalk and jabber.conf files. Could you please take a look
and tell me, if the settings same sane?

*** gtalk.conf ***

[general]
context=google-in   ;;Context to dump call into
allowguest=yes  ;;Allow calls from people not in
externip=MYIP ;; my router is nasty, so I had to note it down, everytime
   ;; it changed
bindaddr=192.168.220.105 ;;my local IP
;;list of peers
;
[guest] ;;special account for options on guest account
disallow=all 
allow=ulaw
allow=alaw
allow=speex
context=google-in
;
username=user1  ;;username of the peer your
username=user2
username=myself
username=user3
;;calling or accepting calls from
connection=gtalk_account;;client or component in jabber.conf
;;for the call to leave on.
;
*** end of gtalk.conf ***

*** jabber.conf ***

[general]
debug=yes   ;;Turn on debugging by default.
autoprune=no;;Auto remove users from buddy list.
autoregister=yes;;Auto register users from buddy list.

[gtalk_account]
type=client
serverhost=talk.google.com
username=mys...@googlemail.com/Talk
secret=my_password
port=5222
usetls=yes
usesasl=yes
buddy=user1
buddy=user2
buddy=myself
statusmessage="Asterisk for Gtalk..."
timezone=100
*** end of jabber.conf ***
   Thanks for reading through this! I very much appreciate this!
   Kindly yours
 Julien


Music was my first love and it will be my last (John Miles)

 FIND MY WEB-PROJECT AT: 
http://ltsb.sourceforge.net
the Linux TextBased Studio guide
=== AND MY PERSONAL PAGES AT: ===
http://www.juliencoder.de

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing a sip call through Asterisk?

[Nathan Clemons]

Turning on qualify=yes, or qualify=60, seems to break the BroadVoice
connection (it goes from UNKNOWN to UNREACHABLE and calls fail).

I'm wondering if they don't support OPTIONS probing or something.

-- Nathan Clemons


On Fri, Apr 16, 2010 at 3:22 PM, Jeff LaCoursiere  wrote:

>
>
> On Fri, 16 Apr 2010, Nathan Clemons wrote:
>
> > I'm looking to find a test tool that will register with our Asterisk
> > (Trixbox) server here at work and place an outgoing call via our main SIP
> > trunk (BroadVoice) to confirm that things are working. I've looked around
> > but I can't seem to find any tools that will do what I'm looking for.
> >
> > I can't just monitor the status of the trunk inside Asterisk, as this is
> the
> > normal status:
> >
>
> [snip]
>
> just add "qualify=yes" to your context and it will monitor the RT latency.
>
> j
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing a sip call through Asterisk?

[Jeff LaCoursiere]

On Fri, 16 Apr 2010, Nathan Clemons wrote:

> I'm looking to find a test tool that will register with our Asterisk
> (Trixbox) server here at work and place an outgoing call via our main SIP
> trunk (BroadVoice) to confirm that things are working. I've looked around
> but I can't seem to find any tools that will do what I'm looking for.
>
> I can't just monitor the status of the trunk inside Asterisk, as this is the
> normal status:
>

[snip]

just add "qualify=yes" to your context and it will monitor the RT latency.

j

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing a sip call through Asterisk?

[Sean Brady]

On 04/16/2010 03:39 PM, Nathan Clemons wrote:
I'm looking to find a test tool that will register with our Asterisk 
(Trixbox) server here at work and place an outgoing call via our main 
SIP trunk (BroadVoice) to confirm that things are working. I've looked 
around but I can't seem to find any tools that will do what I'm 
looking for.


I can't just monitor the status of the trunk inside Asterisk, as this 
is the normal status:


asterisk*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status
BroadVoice/425256  147.135.32.221   N  5060 
Unmonitored

...
37 sip peers [Monitored: 5 online, 31 offline Unmonitored: 1 online, 0 
offline]

asterisk*CLI>

Alternatively, any suggestions as to how I can change the trunk 
configuration so that it is monitored would be appreciated. The peer 
config is set as:


allow=ulaw
disallow=all
canreinvite=no
context=from-trunk
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com 
fromuser=425256
host=sip.broadvoice.com 
insecure=very
nat=yes
secret=XX
type=peer
username=425256


Any assistance would be appreciated. I'd rather know when things fail 
via an automated system rather than learning it's down from the users.


-- Nathan Clemons


I believe that adding qualify= to your 
trunk configuration is what you are looking for for the monitoring 
state.  This will send SIP OPTIONS packets to the trunk periodically.  
See "qualify" in the sip.conf samples or documentation.


From there you can use a monitoring solution to monitor the state of 
the trunk.  Alternatively you can use a OSS tool called SIPp to test SIP 
devices.  See *http://sipp*.sourceforge.net for more information.  This 
is an indispensable tool for SIP and Asterisk troubleshooting.


I hope this helps.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] Testing a sip call through Asterisk?

[Nathan Clemons]

I'm looking to find a test tool that will register with our Asterisk
(Trixbox) server here at work and place an outgoing call via our main SIP
trunk (BroadVoice) to confirm that things are working. I've looked around
but I can't seem to find any tools that will do what I'm looking for.

I can't just monitor the status of the trunk inside Asterisk, as this is the
normal status:

asterisk*CLI> sip show peers
Name/username  HostDyn Nat ACL Port
Status
BroadVoice/425256  147.135.32.221   N  5060
Unmonitored
...
37 sip peers [Monitored: 5 online, 31 offline Unmonitored: 1 online, 0
offline]
asterisk*CLI>

Alternatively, any suggestions as to how I can change the trunk
configuration so that it is monitored would be appreciated. The peer config
is set as:

allow=ulaw
disallow=all
canreinvite=no
context=from-trunk
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com
fromuser=425256
host=sip.broadvoice.com
insecure=very
nat=yes
secret=XX
type=peer
username=425256


Any assistance would be appreciated. I'd rather know when things fail via an
automated system rather than learning it's down from the users.

-- Nathan Clemons
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Testing the Timing Device

[Tzafrir Cohen]

On Thu, Oct 15, 2009 at 08:42:22PM +0100, Dan Journo wrote:
> Hi Danny,
> 
>  
> 
> I've tried that but I get the following errors:-
> 
>  
> 
>  [r...@templateasteriskserver ~]# /etc/init.d/dahdi start
> 
> Loading DAHDI hardware modules:
> 
> FATAL: Module dahdi not found.

The dahdi kernel modules are not available. At least not for your
running kernel.

This typically means you have either not installed them ('make install'
in the source directory of dahdi-linux if you install from source), or
installed them and then switched to a different kernel version for some
reason.

What is the output of:

  uname -a
  find /lib/modules -name dahdi.ko

How have you installed dahdi?

What version of asterisk is it?

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

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

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Testing the Timing Device

[Danny Nicholas]

Try this thread

http://forums.digium.com/viewtopic.php?p=132042&sid=297f2470a0a3d87e91efc1a5
9defcab9

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, October 15, 2009 3:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Testing the Timing Device

 

Ok, its a little better now.

But I still get a fatal message:-

 

[r...@templateasteriskserver dahdi]# /etc/init.d/dahdi start

Loading DAHDI hardware modules:

FATAL: Module dahdi not found.

  dummy:   [  OK  ]

 

Any ideas?

 

Thanks

Dan

 

  _  

Thank you for contacting Kesher Communications Ltd.

IT Maintenance Clients can now receive a faster response by using our Live
Chat and Support Service: Click
<http://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7F508
C6B61A91700343&lang=en&surpre=PreSurvey>  Here

 

This email and any files transmitted with it are confidential and intended
solely for the recipient(s). If you are not the named addressee you should
not disseminate, copy or alter this email. Under no circumstances may this
email be distributed without written permission from the sender. Warning:
Although the Company has taken reasonable precautions to ensure no viruses
are present in this email, the company cannot accept responsibility for any
loss or damage arising from the use of this email or attachments. All prices
exclude VAT unless otherwise stated. No responsibility is taken for any
recommendations made by a sender or by Kesher Communications Ltd.
Recipient(s) takes responsibility for any actions taken as a result of
advice and recommendations given by Kesher Communications Ltd.

 

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

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Testing the Timing Device

[Dan Journo]

Ok, its a little better now.

But I still get a fatal message:-

 

[r...@templateasteriskserver dahdi]# /etc/init.d/dahdi start

Loading DAHDI hardware modules:

FATAL: Module dahdi not found.

  dummy:   [  OK  ]

 

Any ideas?

 

Thanks

Dan

 



Thank you for contacting Kesher Communications Ltd.

IT Maintenance Clients can now receive a faster response by using our
Live Chat and Support Service: Click Here
<http://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7
F508C6B61A91700343&lang=en&surpre=PreSurvey> 

 

This email and any files transmitted with it are confidential and
intended solely for the recipient(s). If you are not the named addressee
you should not disseminate, copy or alter this email. Under no
circumstances may this email be distributed without written permission
from the sender. Warning: Although the Company has taken reasonable
precautions to ensure no viruses are present in this email, the company
cannot accept responsibility for any loss or damage arising from the use
of this email or attachments. All prices exclude VAT unless otherwise
stated. No responsibility is taken for any recommendations made by a
sender or by Kesher Communications Ltd. Recipient(s) takes
responsibility for any actions taken as a result of advice and
recommendations given by Kesher Communications Ltd.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: 15 October 2009 20:40
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Testing the Timing Device

 

What does /etc/dahdi/modules look like?  I suspect that it has each of
the wc* entries in it.  If so, remove those lines and put in "dummy"
(just once).

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, October 15, 2009 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Testing the Timing Device

 

Hi Danny,

 

I've tried that but I get the following errors:-

 

 [r...@templateasteriskserver ~]# /etc/init.d/dahdi start

Loading DAHDI hardware modules:

FATAL: Module dahdi not found.

  wct4xxp:  FATAL: Module wct4xxp not found.

   [FAILED]

  wcte12xp:  FATAL: Module wcte12xp not found.

   [FAILED]

  wct1xxp:  FATAL: Module wct1xxp not found.

   [FAILED]

  wcte11xp:  FATAL: Module wcte11xp not found.

   [FAILED]

  wctdm24xxp:  FATAL: Module wctdm24xxp not found.

   [FAILED]

  wcfxo:  FATAL: Module wcfxo not found.

   [FAILED]

  wctdm:  FATAL: Module wctdm not found.

   [FAILED]

  wcb4xxp:  FATAL: Module wcb4xxp not found.

   [FAILED]

  wctc4xxp:  FATAL: Module wctc4xxp not found.

   [FAILED]

  xpp_usb:  FATAL: Module xpp_usb not found.

   [FAILED]

 

 

[r...@templateasteriskserver ~]# /etc/init.d/zaptel start

Loading zaptel framework:  FATAL: Module zaptel not found.

   [FAILED]

Waiting for zap to come online...

[r...@templateasteriskserver ~]#

 

Any ideas?

 

Thanks

Dan Journo

 



Thank you for contacting Kesher Communications Ltd.

IT Maintenance Clients can now receive a faster response by using our
Live Chat and Support Service: Click Here
<http://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7
F508C6B61A91700343&lang=en&surpre=PreSurvey> 

 

This email and any files transmitted with it are confidential and
intended solely for the recipient(s). If you are not the named addressee
you should not disseminate, copy or alter this email. Under no
circumstances may this email be distributed without written permission
from the sender. Warning: Although the Company has taken reasonable
precautions to ensure no viruses are present in this email, the company
cannot accept responsibility for any loss or damage arising from the use
of this email or attachments. All prices exclude VAT unless otherwise
stated. No responsibility is taken for any recommendations made by a
sender or by Kesher Communications Ltd. Recipient(s) takes
responsibility for any actions taken as a result of advice and
recommendations given by Kesher Communications L

Re: [asterisk-users] Testing the Timing Device

[Danny Nicholas]

What does /etc/dahdi/modules look like?  I suspect that it has each of the
wc* entries in it.  If so, remove those lines and put in "dummy" (just
once).

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, October 15, 2009 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Testing the Timing Device

 

Hi Danny,

 

I've tried that but I get the following errors:-

 

 [r...@templateasteriskserver ~]# /etc/init.d/dahdi start

Loading DAHDI hardware modules:

FATAL: Module dahdi not found.

  wct4xxp:  FATAL: Module wct4xxp not found.

   [FAILED]

  wcte12xp:  FATAL: Module wcte12xp not found.

   [FAILED]

  wct1xxp:  FATAL: Module wct1xxp not found.

   [FAILED]

  wcte11xp:  FATAL: Module wcte11xp not found.

   [FAILED]

  wctdm24xxp:  FATAL: Module wctdm24xxp not found.

   [FAILED]

  wcfxo:  FATAL: Module wcfxo not found.

   [FAILED]

  wctdm:  FATAL: Module wctdm not found.

   [FAILED]

  wcb4xxp:  FATAL: Module wcb4xxp not found.

   [FAILED]

  wctc4xxp:  FATAL: Module wctc4xxp not found.

   [FAILED]

  xpp_usb:  FATAL: Module xpp_usb not found.

   [FAILED]

 

 

[r...@templateasteriskserver ~]# /etc/init.d/zaptel start

Loading zaptel framework:  FATAL: Module zaptel not found.

   [FAILED]

Waiting for zap to come online...

[r...@templateasteriskserver ~]#

 

Any ideas?

 

Thanks

Dan Journo

 

  _  

Thank you for contacting Kesher Communications Ltd.

IT Maintenance Clients can now receive a faster response by using our Live
Chat and Support Service: Click
<http://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7F508
C6B61A91700343&lang=en&surpre=PreSurvey>  Here

 

This email and any files transmitted with it are confidential and intended
solely for the recipient(s). If you are not the named addressee you should
not disseminate, copy or alter this email. Under no circumstances may this
email be distributed without written permission from the sender. Warning:
Although the Company has taken reasonable precautions to ensure no viruses
are present in this email, the company cannot accept responsibility for any
loss or damage arising from the use of this email or attachments. All prices
exclude VAT unless otherwise stated. No responsibility is taken for any
recommendations made by a sender or by Kesher Communications Ltd.
Recipient(s) takes responsibility for any actions taken as a result of
advice and recommendations given by Kesher Communications Ltd.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: 15 October 2009 19:58
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Testing the Timing Device

 

You need to do /etc/init.d/dahdi start or /etc/init.d/zaptel start to load
the devices or dummy devices

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, October 15, 2009 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Testing the Timing Device

 

Hello,

 

Does anyone know how to test the timing device?

I've tried the following but with no luck. 

Zaptel is installed.

I'm trying to use ztdummy as a timer.

 

[r...@templateasteriskserver ~]# dahdi_test

Unable to open dahdi interface: No such file or directory

[r...@templateasteriskserver ~]# zttool

Unable to open /dev/zap/ctl: No such file or directory

 

Thanks

Dan Journo

 

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

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Testing the Timing Device

[Dan Journo]

Hi Danny,

 

I've tried that but I get the following errors:-

 

 [r...@templateasteriskserver ~]# /etc/init.d/dahdi start

Loading DAHDI hardware modules:

FATAL: Module dahdi not found.

  wct4xxp:  FATAL: Module wct4xxp not found.

   [FAILED]

  wcte12xp:  FATAL: Module wcte12xp not found.

   [FAILED]

  wct1xxp:  FATAL: Module wct1xxp not found.

   [FAILED]

  wcte11xp:  FATAL: Module wcte11xp not found.

   [FAILED]

  wctdm24xxp:  FATAL: Module wctdm24xxp not found.

   [FAILED]

  wcfxo:  FATAL: Module wcfxo not found.

   [FAILED]

  wctdm:  FATAL: Module wctdm not found.

   [FAILED]

  wcb4xxp:  FATAL: Module wcb4xxp not found.

   [FAILED]

  wctc4xxp:  FATAL: Module wctc4xxp not found.

   [FAILED]

  xpp_usb:  FATAL: Module xpp_usb not found.

   [FAILED]

 

 

[r...@templateasteriskserver ~]# /etc/init.d/zaptel start

Loading zaptel framework:  FATAL: Module zaptel not found.

   [FAILED]

Waiting for zap to come online...

[r...@templateasteriskserver ~]#

 

Any ideas?

 

Thanks

Dan Journo

 



Thank you for contacting Kesher Communications Ltd.

IT Maintenance Clients can now receive a faster response by using our
Live Chat and Support Service: Click Here
<http://eu.ntrsupport.com/inquiero/web/digisign/digisign.asp?login=I23E7
F508C6B61A91700343&lang=en&surpre=PreSurvey> 

 

This email and any files transmitted with it are confidential and
intended solely for the recipient(s). If you are not the named addressee
you should not disseminate, copy or alter this email. Under no
circumstances may this email be distributed without written permission
from the sender. Warning: Although the Company has taken reasonable
precautions to ensure no viruses are present in this email, the company
cannot accept responsibility for any loss or damage arising from the use
of this email or attachments. All prices exclude VAT unless otherwise
stated. No responsibility is taken for any recommendations made by a
sender or by Kesher Communications Ltd. Recipient(s) takes
responsibility for any actions taken as a result of advice and
recommendations given by Kesher Communications Ltd.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: 15 October 2009 19:58
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Testing the Timing Device

 

You need to do /etc/init.d/dahdi start or /etc/init.d/zaptel start to
load the devices or dummy devices

 

 



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, October 15, 2009 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Testing the Timing Device

 

Hello,

 

Does anyone know how to test the timing device?

I've tried the following but with no luck. 

Zaptel is installed.

I'm trying to use ztdummy as a timer.

 

[r...@templateasteriskserver ~]# dahdi_test

Unable to open dahdi interface: No such file or directory

[r...@templateasteriskserver ~]# zttool

Unable to open /dev/zap/ctl: No such file or directory

 

Thanks

Dan Journo

 

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

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Testing the Timing Device

[Danny Nicholas]

You need to do /etc/init.d/dahdi start or /etc/init.d/zaptel start to load
the devices or dummy devices

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, October 15, 2009 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Testing the Timing Device

 

Hello,

 

Does anyone know how to test the timing device?

I've tried the following but with no luck. 

Zaptel is installed.

I'm trying to use ztdummy as a timer.

 

[r...@templateasteriskserver ~]# dahdi_test

Unable to open dahdi interface: No such file or directory

[r...@templateasteriskserver ~]# zttool

Unable to open /dev/zap/ctl: No such file or directory

 

Thanks

Dan Journo

 

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

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Testing the Timing Device

[Dan Journo]

Hello,

 

Does anyone know how to test the timing device?

I've tried the following but with no luck. 

Zaptel is installed.

I'm trying to use ztdummy as a timer.

 

[r...@templateasteriskserver ~]# dahdi_test

Unable to open dahdi interface: No such file or directory

[r...@templateasteriskserver ~]# zttool

Unable to open /dev/zap/ctl: No such file or directory

 

Thanks

Dan Journo

 

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

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] testing music

[Steve Howes]

On 11 Aug 2009, at 12:03, Tzafrir Cohen wrote:
> So I'm looking for a music file that:
>
> 1. Sounds well (enough) even at the standard PSTN quality (8kHz, mono,
> 16 bits per sample).
>
> 2. Is long enough. E.g. ~10 minutes.
>
> 3. No usage limitation. Freely usable. So I can point it out to  
> someone
> and ask him to try it.
>
>
> I'm too lazy to start looking for such a file. And I bet somebody in
> this list can point me to an existing one.

Standard hold music? One of the tunes is a bit shoddy for stuff like  
this (slow kinda wobbly sound), but the rest seem ok..

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

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] testing music

[Tzafrir Cohen]

While I read on some other mailing list that the human ear is a poor
testing device, it is still a widely available testing device and I
often don't have anything better.

In order to help that device better detect sound quality issues, I tend
to prefer to use lengthy music files. Once I'm familiar enough with the
music I can sense "something is wrong" with relatively little effort and
attention.

So I'm looking for a music file that:

1. Sounds well (enough) even at the standard PSTN quality (8kHz, mono,
16 bits per sample).

2. Is long enough. E.g. ~10 minutes.

3. No usage limitation. Freely usable. So I can point it out to someone
and ask him to try it.


I'm too lazy to start looking for such a file. And I bet somebody in
this list can point me to an existing one.

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

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

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Testing the manager.conf: sending and receiving commands

[Jared Smith]

On Wed, 2009-07-01 at 10:25 -0700, bilal ghayyad wrote:
> Can I telnet to the asterisk machine at the port 5038 and send and receive 
> commands to test if the manager is working fine?

Absolutely!

>  How?

1) Make sure manager is enabled in manager.conf (enabled=yes in
[general] section)

2) Create a manager user, and give that user permissions (see the sample
section in manager.conf named [mark])

3) Type "manager reload" from the Asterisk CLI

4) Telnet to port 5038, as shown below:

[jsm...@mybox ~]$ telnet localhost 5038
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/1.1
Action: Login
Username: jsmith
Secret: doughnuts
Events: on
ActionID: 12345

Response: Success
ActionID: 12345
Message: Authentication accepted

Action: ExtensionState
Exten: 555
Context: lab
ActionID: 987654321

Response: Success
ActionID: 987654321
Message: Extension Status
Exten: 555
Context: lab
Hint: SIP/linksys
Status: 0

-- 
Jared Smith
Training Manager
Digium, Inc.


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

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

Re: [asterisk-users] Testing the manager.conf: sending and receiving commands

[Gordon Henderson]

On Wed, 1 Jul 2009, bilal ghayyad wrote:

> Hi All;
>
> How can I test manager.conf?
>
> Can I telnet to the asterisk machine at the port 5038 and send and 
> receive commands to test if the manager is working fine? How?

Yes!

RTFM would be a fine place to start - or at least the wiki:

   http://www.voip-info.org/wiki/view/Asterisk+manager+API

I suggest preparing a file in one window with commands in it, then copy & 
paste these into the telnet window while you're experimenting.

Gordon

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

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

[asterisk-users] Testing the manager.conf: sending and receiving commands

[bilal ghayyad]

Hi All;

How can I test manager.conf?

Can I telnet to the asterisk machine at the port 5038 and send and receive 
commands to test if the manager is working fine? How?

Regards
Bilal


  

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

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

[asterisk-users] testing the list

[tloginbr-asteriskusers]

I'm having problems sending e-mails to the list. Please ignore this
message, I'm just testing. sorry for the inconvenience.

Thiago




  Abra sua conta no Yahoo! Mail, o único sem limite de espaço para 
armazenamento!
http://br.mail.yahoo.com/

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

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

Re: [asterisk-users] testing please ignore

[Joseph]

On 04/08/08 07:23, John covici wrote:
>If I see this, then messages are getting through.
>

You are lucky :-/
I sent two messages to the list and they never arrived.

-- 
#Joseph

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

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

[asterisk-users] testing please ignore

[John covici]

If I see this, then messages are getting through.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 [EMAIL PROTECTED]

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

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

Re: [asterisk-users] Testing Framework

[Kyle Sexton]

Matt Riddell <[EMAIL PROTECTED]> writes:

> Hi,
>
> I propose we start with a list of things that we think should be tested
> in Asterisk, and means to test them.
>
> Any takers?  Add to the list?  If there is something you believe is
> mission critical to your business, write up a test case for it, and
> we'll all try to code something that can run automatically to test it.
>

I think a good way to get new items on the testing list would be for
Digium/community to take a look at the bug tracker for released versions and ask
'Was there a way we could have caught this bug?'.  Then add that test
case to the list.  In my mind I would rather have too many tests than
not enough! :)

-- 
Kyle Sexton

___

Sign up now for AstriCon 2007!  September 25-28th.  http://www.astricon.net/ 

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

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

Re: [asterisk-users] Testing Framework

[Hariharan Veerappan]

I think the testing frame work includes both the components
and system testing.
I wish to add some more test even though all giants may aware,
since i wish to do some contribution to asterisk what ever i can.

i am plannig for the framework and addon as given below, expecting
techies advise in this,
Testing frame work may contain testing internally and externally,
i mean to say internally, some client may stick into the problem of
voice quality, when more phones on the PBX, that time, we may not
leave the system from the network, their call may get interrupted,
that time internal testing daemon will work for the performance analysis
till the lelvel of without affecting present calls.

externally means PBX system on production for performance analysis,
as per the test case started in the mail.

internal and external testing may be configurable at the run time, through
some verbose like variable configurable at run time, addon  should have the
capability
of releasing the testing call bandwidth, whenever PBX gets new call, this
might be simple
example.

procesding with Testing frame work requirements,
call on volume,
1. SIP - SIP, multiple SIP clients support, through which we can either
direct the call for testing to another client registered in testing frame
work,
or return back to the different client registered in the same testing  frame
work,
when the call incoming and outgoing call are handled in single framework
point,
wave analysis also cane be done with the script and performance can be
easily
evaluated.

On production performance testing, by connecting multiple testing framework
point to the PBX,
having the sending files  in  all the frame work, analysis and performance
evaluation can be
done very easily,

i also think that once it is done for SIP with compatiblity of like
channel driver, we can adapt IAX2, anything we want.

i think this type of testing would make the system stable and provide
good support on system on running also.

Hariharan.V.
R&D Engineer,
NEEVEE Technologies,
On 9/3/07, dave cantera <[EMAIL PROTECTED]> wrote:
>
> matt,
> are you looking for unit testing of the * components or systems testing,
> testing the finished product?  or both?
> I think you are onto something here...  I hope it takes root.  I would
> say put it in the addons.  it would be Great if digium takes it up. it
> is a smart move for them to foster, cajole, nudge, and support it.
> call volume I would leave to others as different processors, O/S,
> builds, kernel versions, and configurations will have too many variables.
>
> I was playing with the idea of monitoring multiple * systems.  perhaps
> we can start out with testing the components and then migrate the
> project (future) to one pbx monitor the other.  we will need scripts to
> initiate some action, config to make some measurements, the scripts to
> gather the results into a nice neat little summary report.  you will
> want to take the human aspect out of the picture as much as possible.
> for example:
>
> on pbx A
>
> * create a recording in multiple formats .gsm, .wav, etc.
> * initiate a script to generate 5,10, or 25 calls to pbx B and
>   play the file
>
> on pbx B
>
> * pbx B gets the calls, records them,
> * copy the recordings from pbx A to pbx B (or have that already
>   done)
> * have a wave analyzer compare the recordings to the original
>   files (you know I won't be writing that program! :)
> * report on anomalies
>
> *call
> *   *Technology
> *   *recording
> delta
> *
> 1
> Zap Provider 1
> 2%
> 2
> VoIP Provider 2
> 5%
> 3
> VoIP Provider 2
> 15%
> ...
> VoIP Provider 3
> ...
>
>
> let me know what you think!
> daveC
>
>
>
> Matt Riddell wrote:
> > Hash: SHA1
> >
> > Hi,
> >
> > So, now that we've all complained about the state of testing of Open
> > Source versions of Asterisk, lets do something about it.
> >
> > I propose we start with a list of things that we think should be tested
> > in Asterisk, and means to test them.
> >
> > Maybe we could run certain tests based on the changes between minor
> > versions?
> >
> > Anyway lets start.
> >
> > Call Volumes
> >
> > 1) Call volume up to x channels from SIP to SIP (i.e. sipp)
> > 2) Call volume up to x channels from IAX2 to SIP
> > 3) Call volume up to x channels from IAX2 to IAX2
> >
> > Application testing
> >
> > 4) Connect x calls between techs to Meetme (leave running for 1 hour)
> > 5) Connect x concurrent calls to VoiceMail
> >
> > Call Centre Testing
> >
> > 6) Send x calls to a queue with no agents in it, leave them holding for
> > x minutes
> > 7) Run x calls against AMD connected to recorded known good files
> >
> > Recording
> >
> > 8) Run x calls recording simultaneously from an automatically generated
> > call, play ulaw/alaw - compare outputs.
> >
> > You get the idea.
> >
> > If people can add to this list, I can start mak

Re: [asterisk-users] Testing Framework

[dave cantera]

matt,
are you looking for unit testing of the * components or systems testing, 
testing the finished product?  or both?
I think you are onto something here...  I hope it takes root.  I would 
say put it in the addons.  it would be Great if digium takes it up. it 
is a smart move for them to foster, cajole, nudge, and support it. 
call volume I would leave to others as different processors, O/S, 
builds, kernel versions, and configurations will have too many variables.

I was playing with the idea of monitoring multiple * systems.  perhaps 
we can start out with testing the components and then migrate the 
project (future) to one pbx monitor the other.  we will need scripts to 
initiate some action, config to make some measurements, the scripts to 
gather the results into a nice neat little summary report.  you will 
want to take the human aspect out of the picture as much as possible.  
for example:

on pbx A

* create a recording in multiple formats .gsm, .wav, etc.
* initiate a script to generate 5,10, or 25 calls to pbx B and
  play the file

on pbx B

* pbx B gets the calls, records them,
* copy the recordings from pbx A to pbx B (or have that already
  done)
* have a wave analyzer compare the recordings to the original
  files (you know I won't be writing that program! :)
* report on anomalies

*call
*   *Technology
*   *recording
delta
*
1
Zap Provider 1
2%
2
VoIP Provider 2
5%
3
VoIP Provider 2
15%
...
VoIP Provider 3
...


let me know what you think!
daveC



Matt Riddell wrote:
> Hash: SHA1
>
> Hi,
>
> So, now that we've all complained about the state of testing of Open
> Source versions of Asterisk, lets do something about it.
>
> I propose we start with a list of things that we think should be tested
> in Asterisk, and means to test them.
>
> Maybe we could run certain tests based on the changes between minor
> versions?
>
> Anyway lets start.
>
> Call Volumes
>
> 1) Call volume up to x channels from SIP to SIP (i.e. sipp)
> 2) Call volume up to x channels from IAX2 to SIP
> 3) Call volume up to x channels from IAX2 to IAX2
>
> Application testing
>
> 4) Connect x calls between techs to Meetme (leave running for 1 hour)
> 5) Connect x concurrent calls to VoiceMail
>
> Call Centre Testing
>
> 6) Send x calls to a queue with no agents in it, leave them holding for
> x minutes
> 7) Run x calls against AMD connected to recorded known good files
>
> Recording
>
> 8) Run x calls recording simultaneously from an automatically generated
> call, play ulaw/alaw - compare outputs.
>
> You get the idea.
>
> If people can add to this list, I can start making a few scripts and
> programs that will test them (as I'm sure others can).
>
> If we end up with a complete list, I'm sure some of our individual QA
> departments can take the responsibility for certain items.
>
> The call volume ones are obviously going to either need a live person to
> dial in at volume and check everything is ok, or a recording which can
> later be checked.
>
> I'm of the opinion that the majority of tests should test individual
> components, but that we should also form some "Application Type"
> frameworks so that we can test integration between Asterisk apps.
>
> Any takers?  Add to the list?  If there is something you believe is
> mission critical to your business, write up a test case for it, and
> we'll all try to code something that can run automatically to test it.
>
> If we try and keep to ANSI C for the testing apps, Digium should be able
> to run them on their multi platform machines as well.
>
> Should these tests be added to Asterisk-Addons or maintained outside of
> the tree?
>
> Anyway, what do you think? Feasible? I already have a few tests here and
> I'm sure others have a few too.  Lets put them all together and get a
> framework going.
>
> - --
> Kind Regards,
>
> Matt Riddell
> Director
> ___
>
> http://www.venturevoip.com (Great new VoIP end to end solution)
> http://www.venturevoip.com/news.php (Daily Asterisk News - html)
> http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.7 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>
> iD8DBQFG1yKhDQNt8rg0Kp4RAv5UAJ48tW28T5lWCQIPTwVimyvlhEPJowCgpnE6
> OF3L2M/6Hc+YBNL1NFx6dzA=
> =OXNn
> -END PGP SIGNATURE-
>
> ___
> --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
>
>
>
>   

-- 
My wife's sister is in California.  
I should buy her a Videophone2008!

Truly, The Next Best Thing to Being There!
--

WorldWideVideoPhones.com
856.380.0894




__

Re: [asterisk-users] Testing Framework

[Atis]

On 8/30/07, Russell Bryant <[EMAIL PROTECTED]> wrote:
> Matt Riddell wrote:
> > Should these tests be added to Asterisk-Addons or maintained outside of
> > the tree?
>
> If people start writing test utilities, I would be happy to host them in a
> subversion repository.  Depending on the size of this stuff, it could probably
> go into the main Asterisk repository.  We'll see where things go ...

Our company have a scriptable testing utility (it's quite far from
being simple to use), but we could share it as basis for testing
framework or at least - as example for inspiration. I just talked to
management,  and this week it will be discussed. I'll write you in
case of success.

Regards,
Atis


-- 
Atis Lezdins,
IT Responsible of BEST Riga,
[EMAIL PROTECTED]
ICQ: 142239285
Skype: atis.lezdins
Cell Phone: +371 28806004 [Tele2, Latvia]
Work phone: +1 800 7502835 [Toll free, USA]
?BEST? -> www.BEST.eu.org

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

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

Re: [asterisk-users] Testing Framework

[Russell Bryant]

Matt Riddell wrote:
> Should these tests be added to Asterisk-Addons or maintained outside of
> the tree?

If people start writing test utilities, I would be happy to host them in a
subversion repository.  Depending on the size of this stuff, it could probably
go into the main Asterisk repository.  We'll see where things go ...

-- 
Russell Bryant
Software Engineer
Digium, Inc.

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

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

[asterisk-users] Testing Framework

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi,

So, now that we've all complained about the state of testing of Open
Source versions of Asterisk, lets do something about it.

I propose we start with a list of things that we think should be tested
in Asterisk, and means to test them.

Maybe we could run certain tests based on the changes between minor
versions?

Anyway lets start.

Call Volumes

1) Call volume up to x channels from SIP to SIP (i.e. sipp)
2) Call volume up to x channels from IAX2 to SIP
3) Call volume up to x channels from IAX2 to IAX2

Application testing

4) Connect x calls between techs to Meetme (leave running for 1 hour)
5) Connect x concurrent calls to VoiceMail

Call Centre Testing

6) Send x calls to a queue with no agents in it, leave them holding for
x minutes
7) Run x calls against AMD connected to recorded known good files

Recording

8) Run x calls recording simultaneously from an automatically generated
call, play ulaw/alaw - compare outputs.

You get the idea.

If people can add to this list, I can start making a few scripts and
programs that will test them (as I'm sure others can).

If we end up with a complete list, I'm sure some of our individual QA
departments can take the responsibility for certain items.

The call volume ones are obviously going to either need a live person to
dial in at volume and check everything is ok, or a recording which can
later be checked.

I'm of the opinion that the majority of tests should test individual
components, but that we should also form some "Application Type"
frameworks so that we can test integration between Asterisk apps.

Any takers?  Add to the list?  If there is something you believe is
mission critical to your business, write up a test case for it, and
we'll all try to code something that can run automatically to test it.

If we try and keep to ANSI C for the testing apps, Digium should be able
to run them on their multi platform machines as well.

Should these tests be added to Asterisk-Addons or maintained outside of
the tree?

Anyway, what do you think? Feasible? I already have a few tests here and
I'm sure others have a few too.  Lets put them all together and get a
framework going.

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFG1yKhDQNt8rg0Kp4RAv5UAJ48tW28T5lWCQIPTwVimyvlhEPJowCgpnE6
OF3L2M/6Hc+YBNL1NFx6dzA=
=OXNn
-END PGP SIGNATURE-

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

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

[asterisk-users] testing

[John D. Scott]

Please disregard.




___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.

[Gavin Henry]

On 09/05/07, Stelios Koroneos <[EMAIL PROTECTED]> wrote:

Hello !

For isdn BRI, the zaphfc in NT mode works fine. Some equipment might require
voltage on the line (although they don't use it to powerup and it just draws
a few mil amps)
As for PRI never tested, i would be interested to know how your test goes


I'll report back on making the first * server as the peer/provider
etc. The card is arrving tomorrow.

Thanks.




Stelios S. Koroneos

Digital OPSiS - Embedded Intelligence
http://www.digital-opsis.com


> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Gavin Henry
> Sent: Wednesday, May 09, 2007 2:09 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
> Non-Commercial Discussion
> Subject: Re: RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.
>
>
> On 09/05/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> > Hi Gavin,
> >
> > A second Asterisk server replacing the provider (best way), or
> doing a loop
> > between two different ISDN ports on a same card (worst way)
> must help you.
>
> Thanks for that. Will get a spare * box.
>
> >
> > Best Regards,
> > Francois BERGERET,
> > France.
> >
> >
> > -Message d'origine-
> > De : [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] De la part de
> Gavin Henry
> > Envoyé : mercredi 9 mai 2007 09:40
> > À : asterisk-users@lists.digium.com
> > Objet : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.
> >
> >
> > Hi All,
> >
> > Can anyone recommend any test kit that you can hook up your
> Pri/Bri cards to
> > without having actual ISDN in your office. For example testing
> an * system
> > before it goes to a clients office.
> >
> > Thanks,
> >
> > Gavin.
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


___
--Bandwidth and Colocation provided by Easynews.com --

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


___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.

[Stelios Koroneos]

Hello !

For isdn BRI, the zaphfc in NT mode works fine. Some equipment might require
voltage on the line (although they don't use it to powerup and it just draws
a few mil amps)
As for PRI never tested, i would be interested to know how your test goes


Stelios S. Koroneos

Digital OPSiS - Embedded Intelligence
http://www.digital-opsis.com


> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Gavin Henry
> Sent: Wednesday, May 09, 2007 2:09 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
> Non-Commercial Discussion
> Subject: Re: RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.
>
>
> On 09/05/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> > Hi Gavin,
> >
> > A second Asterisk server replacing the provider (best way), or
> doing a loop
> > between two different ISDN ports on a same card (worst way)
> must help you.
>
> Thanks for that. Will get a spare * box.
>
> >
> > Best Regards,
> > Francois BERGERET,
> > France.
> >
> >
> > -Message d'origine-
> > De : [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] De la part de
> Gavin Henry
> > Envoyé : mercredi 9 mai 2007 09:40
> > À : asterisk-users@lists.digium.com
> > Objet : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.
> >
> >
> > Hi All,
> >
> > Can anyone recommend any test kit that you can hook up your
> Pri/Bri cards to
> > without having actual ISDN in your office. For example testing
> an * system
> > before it goes to a clients office.
> >
> > Thanks,
> >
> > Gavin.
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.

[Gavin Henry]

On 09/05/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:

Hi Gavin,

A second Asterisk server replacing the provider (best way), or doing a loop
between two different ISDN ports on a same card (worst way) must help you.


Thanks for that. Will get a spare * box.



Best Regards,
Francois BERGERET,
France.


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Gavin Henry
Envoyé : mercredi 9 mai 2007 09:40
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.


Hi All,

Can anyone recommend any test kit that you can hook up your Pri/Bri cards to
without having actual ISDN in your office. For example testing an * system
before it goes to a clients office.

Thanks,

Gavin.
___
--Bandwidth and Colocation provided by Easynews.com --

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

___
--Bandwidth and Colocation provided by Easynews.com --

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


___
--Bandwidth and Colocation provided by Easynews.com --

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

RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.

[f6hqz-m]

Hi Gavin,

A second Asterisk server replacing the provider (best way), or doing a loop
between two different ISDN ports on a same card (worst way) must help you.

Best Regards,
Francois BERGERET,
France.


-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Gavin Henry
Envoyé : mercredi 9 mai 2007 09:40
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.


Hi All,

Can anyone recommend any test kit that you can hook up your Pri/Bri cards to
without having actual ISDN in your office. For example testing an * system
before it goes to a clients office.

Thanks,

Gavin.
___
--Bandwidth and Colocation provided by Easynews.com --

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

___
--Bandwidth and Colocation provided by Easynews.com --

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

[asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.

[Gavin Henry]

Hi All,

Can anyone recommend any test kit that you can hook up your Pri/Bri
cards to without having actual ISDN in your office. For example
testing an * system before it goes to a clients office.

Thanks,

Gavin.
___
--Bandwidth and Colocation provided by Easynews.com --

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

[asterisk-users] Testing Asterisk and Zaptel

[Martin Smith]

Hello Asterisk-Users,

My organization is putting together a VoIP setup with Asterisk for a
call center, and we currently have two identical machines for
redundancy. How do you test your redundant machines? How do you test for
load problems? Do you have a strategy or regular plan?

I'm most concerned about load testing Zaptel, as Zaptel + Load is the
quickest way to "out" any problems for us when they exist. We have 3 x
PRI circuits, and we'd specifically love some suggestions for ways to
load test our Zaptel cards without needing to interrupt service  on the
PRIs (I've heard of a crossover-ish cable that may do that, but I
haven't been able to find any more about it?).

If it helps, our PRIs look like: span=1,1,0,esf,b8zs.

We'd love to load test those (Zaptel) and perhaps other parts of the
setup, and we'd like to do this for our redundant machine and our
primary, active one. We can't really give up the PRIs without some
downtime, so we're specifically interested in solutions that allow a
primary machine to remain in operation while testing a secondary, and
without using up the PRI circuits for testing (but we want to test our
cards for load).

Thanks!

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221
___
--Bandwidth and Colocation provided by Easynews.com --

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

[asterisk-users] Testing asterisk with sipp

[John Albano]

Hi all,

I'm trying to use SIPP (http://sipp.sourceforge.net/) to stress-test our 
asterisk installation. We have a very simple dialplan that uses FastAgi. 
I'm finding that all calls to "GET VARIABLE" from the FastAgi are 
returning null when the dialplan is invoked from sipp -- and they work 
fine when invoked from a softphone on the same machine, for example.


Does anyone have any insight as to what might be going on here?

Thanks
John

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [asterisk-users] Testing IVR / Callcenter applications

[miguel gmail]

Hi,


We are developing an application to be deployed on E1 lines (inbound and
outbound calls)
What is the best way to fully test the application if we do not have E1
lines in the development environment?
Is there some kind of software tester to test IVR/Callcenter
applications virtually??


I just begun to think how to do the same thing... but considering a
Cisco infrastructure (CallManager, IPIVR, voice gateway/router,
proggers...)

Is there any way i can trigger a bunch of calls to the cisco
callmanager (and then to the IVR). Ideally, i was thinking about
something scripteable, so i can extract processing times and so.

Sorry if i sound like a newbie... it is because i am a newbie (first
time on voip, and just discovered open source voip packages).

Thanks very much in advance!

--
Saludos,
miguel

Los agujeros negros son lugares donde dios dividió por cero.

Black holes are places where god divided by zero.
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [asterisk-users] Testing IVR / Callcenter applications

[Rehan Allah Wala]

you can use a SIP based phone service to try it out




> 
> Hello
> We are developing an application to be deployed on E1 lines (inbound and
> outbound calls)
> What is the best way to fully test the application if we do not have E1
> lines in the development environment?
> Is there some kind of software tester to test IVR/Callcenter 
> applications virtually??
> 
> thanks and best regards 


Super Technologies Inc., Pensacola, Florida
http://www.SuperTec.com - Technologies from tomorrow, Today!

MSN: [EMAIL PROTECTED]
Skype: Rehan33

"First they ignore you, then they laugh at you, then they fight you, then you 
win." By Mahatma Gandhi.

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [asterisk-users] Testing IVR / Callcenter applications

[Matt Florell]

Hello,

We usually use a crossover T1/E1 cable and a multi-port T1/E1 card and
call the server from itself or another Asterisk server. We have used
this method to do stress testing in VICIDIAL, which has a builtin set
of tools for stress testing outbound dialing.

MATT---

On 1/31/07, fadi mujahid <[EMAIL PROTECTED]> wrote:

Thanks for ur suggestion.
But the problem is that won't test the queuing of the outbound and inbound
calls of the callcenter

thanks again


On 1/31/07, Alejandro Lengua <[EMAIL PROTECTED]> wrote:
> Why don´t you put the IVR in an extension...
> and call it also from an extension of the same PBX.
>
> On 1/31/07, fadi mujahid <[EMAIL PROTECTED]> wrote:
> > Hello
> > We are developing an application to be deployed on E1 lines (inbound and
> > outbound calls)
> > What is the best way to fully test the application if we do not have E1
> > lines in the development environment?
> > Is there some kind of software tester to test IVR/Callcenter
> > applications virtually??
> >
> > thanks and best regards
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> >
http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
http://lists.digium.com/mailman/listinfo/asterisk-users
>


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

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



___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [asterisk-users] Testing IVR / Callcenter applications

[Joe Dennick]

You can use a "cross-over" cable between Asterisk boxes to imitate the 
functionality of a T/E-1.


Bill Gibbs wrote:


What do you mean?  Setup another box, make a bunch of calls (as if you 
were clients) into the production box, use back to back E1 cards.


 


Bill

 




*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *fadi 
mujahid

*Sent:* Wednesday, January 31, 2007 10:34 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Testing IVR / Callcenter applications

 


Thanks for ur suggestion.
But the problem is that won't test the queuing of the outbound and 
inbound calls of the callcenter


thanks again

On 1/31/07, *Alejandro Lengua* <[EMAIL PROTECTED] 
<mailto:[EMAIL PROTECTED]>> wrote:


Why don´t you put the IVR in an extension...
and call it also from an extension of the same PBX.

On 1/31/07, fadi mujahid <[EMAIL PROTECTED] 
<mailto:[EMAIL PROTECTED]>> wrote:

> Hello
> We are developing an application to be deployed on E1 lines (inbound and
> outbound calls)
> What is the best way to fully test the application if we do not have E1
> lines in the development environment?
> Is there some kind of software tester to test IVR/Callcenter
> applications virtually??
>
> thanks and best regards
> ___
> --Bandwidth and Colocation provided by Easynews.com 
<http://Easynews.com> --

>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
___
--Bandwidth and Colocation provided by Easynews.com 
<http://Easynews.com> --


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

 




___
--Bandwidth and Colocation provided by Easynews.com --

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

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

Re: [asterisk-users] Testing IVR / Callcenter applications

[Time Bandit]

We are developing an application to be deployed on E1 lines (inbound and
outbound calls)
What is the best way to fully test the application if we do not have E1
lines in the development environment?
Is there some kind of software tester to test IVR/Callcenter
applications virtually??

Just use an IAX or SIP thrunk to/from another Asterisk.

there is no real difference from Asterisk's stand point if the call
comes from IAX, SIP or ZAP

hth
___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [asterisk-users] Testing IVR / Callcenter applications

[Bill Gibbs]

What do you mean?  Setup another box, make a bunch of calls (as if you were 
clients) into the production box, use back to back E1 cards.

 

Bill

 



From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of fadi mujahid
Sent: Wednesday, January 31, 2007 10:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Testing IVR / Callcenter applications

 

Thanks for ur suggestion. 
But the problem is that won't test the queuing of the outbound and inbound 
calls of the callcenter

thanks again

On 1/31/07, Alejandro Lengua <[EMAIL PROTECTED]> wrote:

Why don´t you put the IVR in an extension...
and call it also from an extension of the same PBX.

On 1/31/07, fadi mujahid <[EMAIL PROTECTED]> wrote:
> Hello
> We are developing an application to be deployed on E1 lines (inbound and
> outbound calls)
> What is the best way to fully test the application if we do not have E1
> lines in the development environment? 
> Is there some kind of software tester to test IVR/Callcenter
> applications virtually??
>
> thanks and best regards
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
___
--Bandwidth and Colocation provided by Easynews.com --

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

 

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [asterisk-users] Testing IVR / Callcenter applications

[fadi mujahid]

Thanks for ur suggestion.
But the problem is that won't test the queuing of the outbound and inbound
calls of the callcenter

thanks again

On 1/31/07, Alejandro Lengua <[EMAIL PROTECTED]> wrote:


Why don´t you put the IVR in an extension...
and call it also from an extension of the same PBX.

On 1/31/07, fadi mujahid <[EMAIL PROTECTED]> wrote:
> Hello
> We are developing an application to be deployed on E1 lines (inbound and
> outbound calls)
> What is the best way to fully test the application if we do not have E1
> lines in the development environment?
> Is there some kind of software tester to test IVR/Callcenter
> applications virtually??
>
> thanks and best regards
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
___
--Bandwidth and Colocation provided by Easynews.com --

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

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [asterisk-users] Testing IVR / Callcenter applications

[Alejandro Lengua]

Why don´t you put the IVR in an extension...
and call it also from an extension of the same PBX.

On 1/31/07, fadi mujahid <[EMAIL PROTECTED]> wrote:

Hello
We are developing an application to be deployed on E1 lines (inbound and
outbound calls)
What is the best way to fully test the application if we do not have E1
lines in the development environment?
Is there some kind of software tester to test IVR/Callcenter
applications virtually??

thanks and best regards
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

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




___
--Bandwidth and Colocation provided by Easynews.com --

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

[asterisk-users] Testing IVR / Callcenter applications

[fadi mujahid]

Hello
We are developing an application to be deployed on E1 lines (inbound and
outbound calls)
What is the best way to fully test the application if we do not have E1
lines in the development environment?
Is there some kind of software tester to test IVR/Callcenter
applications virtually??

thanks and best regards
___
--Bandwidth and Colocation provided by Easynews.com --

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

[asterisk-users] testing

[Forum]

 

 

 






___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [asterisk-users] Testing 911?

[Brian Vincent \(C\)]

The place answering the calls is generally known as the PSAP ("public
safety answering point").  As others noted, test calls are fine as long
as you call the non-emergency number first to let them know you're about
to do it.  I'll admit I don't always call in advance though.  Anyway,
calling the fire department or police department may not get you in
contact with the right person.  Asking for the PSAP number should get
you to the right person.

---
Brian Vincent
Copper Mountain Telecom
[EMAIL PROTECTED]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Monday, July 17, 2006 5:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Testing 911?

voiplist wrote:
> It seems that 911 is important enough that when setting up an Asterisk
> box, it should be tested.
> 
> How do you go about testing 911 dialing without getting fined for
> calling for a non-emergency reason?
> 
> Is there some circumstances where you can ask permission from the city
> ahead of time?

As others have posted, test calls are allowed but the 911 center would 
prefer they be completed during non-peak times. The only way to know 
what their non-peak periods are is to give them a call on the 
non-emergency number and ask.

___
--Bandwidth and Colocation provided by Easynews.com --

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

Confidentiality Warning: This message and any attachments are intended only for 
the use of the intended recipient(s), 
are confidential, and may be privileged. If you are not the intended recipient, 
you are hereby notified that any review, 
retransmission, conversion to hard copy, copying, circulation or other use of 
this message and any attachments is strictly 
prohibited. If you are not the intended recipient, please notify the sender 
immediately by return e-mail, and delete this 
message and any attachments from your system. Thank you.
__


___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [asterisk-users] Testing 911?

[C F]

I do it all the time, after I finish installing a PBX (asterisk or
other PBX) I dial 911 and say: Hi this is a test call, I'm a PBX tech,
just finished an installation and just wanted to make sure that 911
works. Then I ask the operator on the other end of the line to confirm
the e911 info he has with me, to make sure that it matches the Address
and phone number that I am realy calling from.

On 7/17/06, voiplist <[EMAIL PROTECTED]> wrote:

It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.

How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?

Is there some circumstances where you can ask permission from the city
ahead of time?

I realize this may be a real stupid question but I have not seen this
discussed and I am curious.
___
--Bandwidth and Colocation provided by Easynews.com --

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


___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [asterisk-users] Testing 911?

[Rich Adamson]

voiplist wrote:

It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.

How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?

Is there some circumstances where you can ask permission from the city
ahead of time?


As others have posted, test calls are allowed but the 911 center would 
prefer they be completed during non-peak times. The only way to know 
what their non-peak periods are is to give them a call on the 
non-emergency number and ask.


___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [asterisk-users] Testing 911?

[Watkins, Bradley]

This is the tact that I take, and it's never been a problem for us.

Regards,
- Brad 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Monday, July 17, 2006 2:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Testing 911?

I call and immediately identify this as a test call.

I state the following. My Nane, and the fact that I am the PBX tech,
(engineer confuses them). I ask them to confirm my address and call back
number I provide to them.  If all is OK I thank them and hang up. I do
not think it is a false call if you identify it as such and give the
information. 

I once was almost charged with a false 911 call, I had added a daemon to
call my pager with a server number followed by 911 when a particular
server went down. It was a typo and not only did the server number NOT
appear but it was dialing 911 instead of my pager. I get a call from the
building security that my office door was open and that there were
firemen and police inside. I rushed over, thinking the worst and while I
was there trying to figure out why they were there they get another call
from dispatch stating that the person was calling again. After asking
the dispatcher the phone number (ANI) that was calling, I disco'ed the
modem. Thank god I was using POTS at the time.  I acted stupid and told
them it must have been a virus or something. I until this day had kept
quiet, I hope the statue of limitations has passed

Alex


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of voiplist
Sent: Monday, July 17, 2006 2:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Testing 911?

It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.

How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?

Is there some circumstances where you can ask permission from the city
ahead of time?

I realize this may be a real stupid question but I have not seen this
discussed and I am curious.
___
--Bandwidth and Colocation provided by Easynews.com --

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

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

The contents of this e-mail are intended for the named addressee only. It 
contains information that may be confidential. Unless you are the named 
addressee or an authorized designee, you may not copy or use it, or disclose it 
to anyone else. If you received it in error please notify us immediately and 
then destroy it. 
___
--Bandwidth and Colocation provided by Easynews.com --

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

RE: [asterisk-users] Testing 911?

[Alexander Lopez]

I call and immediately identify this as a test call.

I state the following. My Nane, and the fact that I am the PBX tech,
(engineer confuses them). I ask them to confirm my address and call back
number I provide to them.  If all is OK I thank them and hang up. I do
not think it is a false call if you identify it as such and give the
information. 

I once was almost charged with a false 911 call, I had added a daemon to
call my pager with a server number followed by 911 when a particular
server went down. It was a typo and not only did the server number NOT
appear but it was dialing 911 instead of my pager. I get a call from the
building security that my office door was open and that there were
firemen and police inside. I rushed over, thinking the worst and while I
was there trying to figure out why they were there they get another call
from dispatch stating that the person was calling again. After asking
the dispatcher the phone number (ANI) that was calling, I disco'ed the
modem. Thank god I was using POTS at the time.  I acted stupid and told
them it must have been a virus or something. I until this day had kept
quiet, I hope the statue of limitations has passed

Alex


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of voiplist
Sent: Monday, July 17, 2006 2:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Testing 911?

It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.

How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?

Is there some circumstances where you can ask permission from the city
ahead of time?

I realize this may be a real stupid question but I have not seen this
discussed and I am curious.
___
--Bandwidth and Colocation provided by Easynews.com --

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

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

Re: [asterisk-users] Testing 911?

[Martin Joseph]

On Jul 16, 2006, at 11:05 PM, voiplist wrote:


It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.

How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?

Is there some circumstances where you can ask permission from the city
ahead of time?

I realize this may be a real stupid question but I have not seen this
discussed and I am curious.
Actually, not stupid at all.  I know in one case I have configured 911 
to dial on a 7 digit number for the local police, and I spoke to the 
police to let them know this is my setup.


In other words,  when I dial 911 from my house (shoreline, Washington 
state USA) I don't want it to dial 911 from my office (FXO is in 
Seattle) as that would be calling the wrong police to the wrong 
address.  I requested from the Shoreline police that they make a record 
of the fact that calls to them from my seattle number are actually 
coming from my shoreline address.


Hopefully I never need to test this under fire...


___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [asterisk-users] Testing 911?

[Brian Swan]

I don't think it's a stupid question at all.  Testing 911 routing is  
very important, and it would suck to find out it didn't work when you  
needed it to.  When I tested 911 at my wife's small business (we're  
on ZAP channels), I first called the non-emergency number for our  
local police department.  Depending on the size of the city your in,  
they may tell you to call a different policy department where the 911  
center for your area is located.  I then called their non-emergency  
number and explained to them that we were installing a new phone  
system and needed to test 911 functionality.  They said "No problem,  
let me transfer you to the radio room", I assume when they transfered  
me I was then talking to one of the 911 supervisors or something.  I  
explained to them that I needed to make two test calls (one to 9,911  
and one to 911 as I have our system setup) in order to test 911  
functionality, and informed them that I would be calling back  
immediately after I hung up with them.   They said "Sure, no problem."


When you do the actual do the deed, identify who you are (full name),  
where you are calling from (business name, etc), and that this is a  
test call on a new phone system.  They will usually read back to you  
the address they have on file for your phone number, and possibly  
some other information.  If you are using a T1, PRI they will also  
verify some E-911 information you are sending (ANI?  Help me out here  
someone...)  Also, I think it's important that you close by telling  
them that you're done testing, or that you have one (or two, or X)  
more test calls to make.


I tried to test out as much as I could in advance, so that I was  
fairly certain I wouldn't have to call them more then twice -- even  
though they know it's a test call, they may still be a little "short"  
with you on call #2 since I'm sure they have plenty of real  
emergencies to deal with. :)  Along those same lines, use some  
judgement as to when you perform your testing.  For instance, testing  
during severe weather, or during a hurricane probably wouldn't be a  
good idea.  Along those same lines (and some what less obvious ;),  
you may NOT want to test on a Friday or Saturday night if it could be  
avoided.   I actually used 411 while I was doing the initial setting  
up and testing to make sure I got everything right, then when I was  
99.99% sure it would work, I switched the 4 to a 9 and tested it for  
real.


Hope that helps!
Swannie



On Jul 17, 2006, at 1:05 AM, voiplist wrote:


It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.

How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?

Is there some circumstances where you can ask permission from the city
ahead of time?

I realize this may be a real stupid question but I have not seen this
discussed and I am curious.
___
--Bandwidth and Colocation provided by Easynews.com --

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


___
--Bandwidth and Colocation provided by Easynews.com --

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

[asterisk-users] Testing 911?

[voiplist]

It seems that 911 is important enough that when setting up an Asterisk
box, it should be tested.

How do you go about testing 911 dialing without getting fined for
calling for a non-emergency reason?

Is there some circumstances where you can ask permission from the city
ahead of time?

I realize this may be a real stupid question but I have not seen this
discussed and I am curious.
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] Testing a FastAGI script

[didier]

Bonjour,

En attendant la naturalisation Espagnole, es tu rentré dans les arcanes 
des DEADAGI ? (qui dit Fastagi dit un peu.)

J'avoue ne pas etre sur des limites ou non de ces bestioles.
-Juste une AGI qui ne s'interromp pas sur hangup ?
-une AGI qui permet d'accèder aux $var d'un channel meme en cas de hangup?
-Les 2 ? (probable)
- ou encore un peu plus mais là, euh, peu documenté et même les sources.

on l'utilise sur l'extension h, ok et même ailleurs si l'on veut, 
mais bon, hum, des bribes à droite à gauche (interagit avec le cdr pour 
ne pas effacer)


Bon, on ne sait jamais, un francophone à peut être des vues plus claires 
que les miennes (parce que les explications anglophones, parfois 
'intraduisibles' )


Bonne journée jusqu'à 20 h

Didier









Olivier a écrit :


Hi,

(I tried to post this message a week ago but I don't think it could 
reach the list. Please forgive me if you already received it).


I would like to develop my first FastAGI script.
I would like to test it independently from Asterisk for the sake of 
simplicity.


Which linux (or cygwin) tool is the best for that ?
Using this tool, I will open a FastAGI connection, throw data in and 
read data from.


With AGI script, echo or cat commands are enough.
But what are the simplest ones for FastAGI ?

Regards



___
--Bandwidth and Colocation provided by Easynews.com --

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



___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] Testing a FastAGI script

[Olivier]

Hi,(I tried to post this message a week ago but I don't think it could reach the list. Please forgive me if you already received it).I would like to develop my first FastAGI script.I would like to test 
it independently from Asterisk for the sake of simplicity.Which linux 
(or cygwin) tool is the best for that ?Using this tool, I will open a 
FastAGI connection, throw data in and read data from. With AGI script, 
echo or cat commands are enough.But what are the simplest ones for FastAGI 
?Regards
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] testing list mail - please ignore

[Tim Litwiller]

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] Testing asterisk faxing functionality

[patryk]

"You could always use System() to copy a call spool file to launch the
outbound fax call.  I don't really think a 3rd party app is necessary."

Could You explain this  please?  Or  maybe some links to 
documentation and examples  ?


Thanks Patryk.
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] Testing asterisk faxing functionality

[Corey S. McFadden]

You could always use System() to copy a call spool file to launch the 
outbound fax call.  I don't really think a 3rd party app is necessary.

-Corey



On Mon, 27 Mar 2006, Gary Richardson wrote:

> I was playing with the fax stuff over IP on Friday. Unless you're
> receiving faxes from a PSTN circuit, it doesn't work so well.
> 
> Also, I don't think you can chain txfax and rxfax like that. When you
> hit the s,2 part, it's going to play the fax out to the handset you
> dialed from. You'll need something like hylafax to send the fax.
> 
> And you probably want to call Dial(Local/[EMAIL PROTECTED]) to call a
> local extension..
> 
> On 3/27/06, patryk <[EMAIL PROTECTED]> wrote:
> > I have asterisk with rxfax txfax modules.I want
> > to test fax sendig and reciving in one asterisk
> > instance, in extensions.conf I have :
> >
> > exten => 1234567,1,rxfax(/home/patryk/fax-new.tif|debug)
> >
> > exten => s,1,Dial(1234567)
> > exten => s,2,txfax(/home/patryk/fax.tif|caller|debug)
> >
> > but I doesn't seem to work.But when I'm calling on this number I can
> > hear fax tones.
> > I can't find sip client with fax fuctionality for linux I think it would
> > help with testing.
> >
> > ___
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> ___
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> *
> This message has been scanned for viruses and
> dangerous content, and is believed to be clean.
> 
> 


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

___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] Testing asterisk faxing functionality

[Gary Richardson]

I was playing with the fax stuff over IP on Friday. Unless you're
receiving faxes from a PSTN circuit, it doesn't work so well.

Also, I don't think you can chain txfax and rxfax like that. When you
hit the s,2 part, it's going to play the fax out to the handset you
dialed from. You'll need something like hylafax to send the fax.

And you probably want to call Dial(Local/[EMAIL PROTECTED]) to call a
local extension..

On 3/27/06, patryk <[EMAIL PROTECTED]> wrote:
> I have asterisk with rxfax txfax modules.I want
> to test fax sendig and reciving in one asterisk
> instance, in extensions.conf I have :
>
> exten => 1234567,1,rxfax(/home/patryk/fax-new.tif|debug)
>
> exten => s,1,Dial(1234567)
> exten => s,2,txfax(/home/patryk/fax.tif|caller|debug)
>
> but I doesn't seem to work.But when I'm calling on this number I can
> hear fax tones.
> I can't find sip client with fax fuctionality for linux I think it would
> help with testing.
>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] Testing asterisk faxing functionality

[patryk]

I have asterisk with rxfax txfax modules.I want
to test fax sendig and reciving in one asterisk
instance, in extensions.conf I have :

exten => 1234567,1,rxfax(/home/patryk/fax-new.tif|debug)

exten => s,1,Dial(1234567)
exten => s,2,txfax(/home/patryk/fax.tif|caller|debug)

but I doesn't seem to work.But when I'm calling on this number I can
hear fax tones.
I can't find sip client with fax fuctionality for linux I think it would
help with testing.

___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] Testing IAX links

[Michael Welter]

I need to test QoS on an IAX link between a server in Colorado and a 
server in Europe.  I know I could install a Milliwatt extension on the 
European server and just listen, but is there a more scientific method 
to collect QoS metrics?


Thanks

P.S.  I'm getting a lot of "Page Not Found" on lists.digium.com.  Are 
the older posts being purged?


--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] Testing List (JUST A TEST)

[Facundo Ameal]

we hear you loud and clear

2006/1/23, [EMAIL PROTECTED] <[EMAIL PROTECTED]>:
> Sorry, I haven't received a message in a few hours, just testing to see if
> it is alive.
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


--
Facundo Ameal.
famealgmailcom
Linux User #395088

Open your mind, use open source.
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] Testing List (JUST A TEST)

[burke]

Sorry, I haven't received a message in a few hours, just testing to see if
it is alive.
___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] Testing 10.0.0.203 with 10.0.0.0

[Tomislav Parcina]

FC4, Asterisk 1.0.9 and SjPhone softphone. On CLI I get this message 
every 20 sec.
# Testing 10.0.0.203 with 10.0.0.0

10.0.0.203 is the IP of softphone and 10.0.0.0 is the network defind in 
sip.conf. Asterisk server is on 10.0.0.26 address.

Why do I get this message?

sip.conf

[general]
externip = 123.123.143.254  
fromdomain=lama.hr
localnet=10.0.0.0/255.255.255.0

port=5060   
bindaddr=0.0.0.0
context=sip 
srvlookup=yes   
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=ulaw  
allow=alaw
musicclass=default
useragent=PBX

[201]   
type=friend 
username=201
secret=myswc
host=dynamic
defaultip=10.0.0.83
mailbox=201 
canreinvite=yes 

[211]   
type=friend 
username=211
secret=mysec
host=dynamic
defaultip=10.0.0.203
mailbox=211 
canreinvite=yes 



-- 

Tomislav Parcina
[EMAIL PROTECTED]

___
--Bandwidth and Colocation provided by Easynews.com --

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

[Asterisk-Users] Testing Sangoma A2022-SO card with Asterisk 1.2

[Mike Dent]

Hi,
I'm testing a Sangoma analogue card with two modules cards installed
(2 x FXO and 2 x FXS total)
I'm actually doing the testing so far in a new install of [EMAIL PROTECTED] 2.1.
The kernel is seeing the card but I'm struggling with my understanding of the
zaptel.conf file and possibly the wanpipe1.conf file.

[EMAIL PROTECTED] wanpipe]# cat /proc/zaptel/2
Span 2: WRTDM/0 "wrtdm Board 1"

   2 WRTDM/0/0
   3 WRTDM/0/1
   4 WRTDM/0/2
   5 WRTDM/0/3
   6
   7
   8
   9
  10
  11
  12
  13
  14
  15
  16
  17
[EMAIL PROTECTED] wanpipe]#

Does anybody have an example of the relevant entries for these two files please?

Many thanks,
Mike

(keywords: analog, A202, A200)
___
--Bandwidth and Colocation provided by Easynews.com --

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

Re: [Asterisk-Users] testing

[Matt Riddell]

ram wrote:
> Hi
>  
> why my posting are not accepting in this list

Don't know.

-- 
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] testing

[ram]

Hi
 
why my posting are not accepting in this list
 
ram
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] Testing with X101P

[Jennifer Hales]

Hello Carlos,

Try putting in a 
exten => s,5,WaitExten,5
after your background,welcome.
It will give you 5 seconds to input your extension.

Regards
Jenn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos Medina
Sent: Monday, November 07, 2005 11:56 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Testing with X101P

Hi there, im testing my asterisk box using a Modem
Intel 56K which on the documentation says it must have
the same behavior as an X101P. So im trying to
configure just a simple line with 6 extensions.
Asterisk loads fine and when im testing an incoming
call the welcome message answers but when im trying to
dial to any extension, anything happens is like
asterisk dont recognice any digit after the welcome.
Im using Asterisk version 1.0.9 and im attaching my
extensions.conf which is very simple.

Any clue will be very helpful.

Thanks a lot for your help.

Carlos Andres Medina



__ 
Yahoo! FareChase: Search multiple travel sites in one click.
http://farechase.yahoo.com

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Testing with X101P

[Carlos Medina]

Hi there, im testing my asterisk box using a Modem
Intel 56K which on the documentation says it must have
the same behavior as an X101P. So im trying to
configure just a simple line with 6 extensions.
Asterisk loads fine and when im testing an incoming
call the welcome message answers but when im trying to
dial to any extension, anything happens is like
asterisk dont recognice any digit after the welcome.
Im using Asterisk version 1.0.9 and im attaching my
extensions.conf which is very simple.

Any clue will be very helpful.

Thanks a lot for your help.

Carlos Andres Medina



__ 
Yahoo! FareChase: Search multiple travel sites in one click.
http://farechase.yahoo.com

extensions.conf
Description: 3949034846-extensions.conf
___
--Bandwidth and Colocation sponsored by Easynews.com --

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