Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[George Joseph]

On Wed, Jun 6, 2018 at 1:51 AM Olivier  wrote:

>
>
> 2018-06-05 20:29 GMT+02:00 George Joseph :
>
>>
>>
>> On Tue, Jun 5, 2018 at 10:59 AM Olivier  wrote:
>>
>>>
>>>
>>> 2018-06-05 15:27 GMT+02:00 George Joseph :
>>> Thank  you very much, George for replying.
>>>


 On Tue, Jun 5, 2018 at 3:35 AM Olivier  wrote:

> Hi,
>
> After a long discussion with a friend, I would like to ask here:
>
> 1.According SIP RFCs, is possible/recommended to have different values
> in From and P-Asserted-Id fields ?
> For instance, From field showing 123456789 and P-Asserted-Id showing
> 987654321 (beside privacy considerations) ?
>

 Possible? yes absolutely.

>>>
>>> How would you then configure both headers, respectively ?
>>>
>>> From memory, in previous testings, whenever  CALLERID was set to
>>> WHATEVER, P-Asserted-Id was also set to WHATEVER and vice versa, so that I
>>> inferred from this that P-Asserted-Id was meant for Privacy
>>> considerations and nothing else (see [1])
>>>
>>
>> PAI should be used to indicate the calling party's identification
>> regardless of privacy concerns.  In the dialplan you can use the CALLERPRES
>> function to control privacy on a call by call basis.
>>
>>
>>
> I'm sorry but I still have a doubt ...
> Let me re-phrase my question:
>
> My setup is:
> Asterisk <--- PJSIP ---> Bob
>
> For a reason, I want Bob's phone to receive a call with the following
> headers:
>
> From: "Foo" ;tag=as75ee8c7c
> P-Asserted-Id: "Foo" >;whatever
>
> My dialplan is:
> same = n,Set(CALLERID(num)=999)
> XXX
> same = n,Dial(PJSIP/123456@bob)
>

> What shall I replace XXX with to allow me to set 8 in the user part of
> P-Asserted-Id URI (see example above) ?
>

You don't need anything for XXX.  Just the "Set(CALLERID(num)=999)" should
do it.


> CALLERPRES would change From or P-Asserted-Id but not having different
> user parts in URI, would it ?
>

CALLERPRES affects From and Contact but doesn't affect PAI at all.  It does
add a Privacy header though.
BTW, you should use CALLERID(pres) instead of CALLERPRES.


>
> To my knowledge, a possible way to implement what I'm after is to "turn
> off" P-Asserted-Id feature, add a custom P-Asserted-Id header with
> PJSIP_HEADER.
> Am I missing something ?
>

Either that or I am. :)   The example you have above should work.



>
>
>
>>
>>
>>>
>>>
>>> [1]
>>> https://www.voip-info.org/p-asserted-identity-and-remote-party-id-header/
>>>
>>>
 Recommended? who knows?  Implementations are all over the place.  I've
 always thought of the From header as identifying the user agent making the
 request which kinda agrees with RFC3261.   The PAI header should contain
 the identity of the original caller.


>
> 2. When Bob forwards to Cory a call coming from Alice, would expect
> Diversion/History-Info header to include Alice's number ?
>

 No.  The diversion header shows who the diverter is.
 https://tools.ietf.org/html/rfc5806

>>>
>
> Thank for  this reference: I think I confused diverter/caller/callee roles
> when I first read this document.
>
> So, if Bob forwards to Cory a call from Alice, in which headers would you
> expect Alice and Bob numbers to respectively appear ?
>
>
Well, if you just have 3 user agents without asterisk in the middle
Alice sends INVITE to Bob.
Bob returns a 302 to Alice with Cory as the Contact and Bob as the Diversion
Alice sends an ACK to Bob.
Alice sends a new INVITE to Cory.

If Asterisk is in the middle then...
Alice sends INVITE to Bob via Asterisk
Asterisk sends INVITE to Bob with Alice in From/PAI
Bob returns a 302 to Asterisk with Cory in Contact and Bob in Diversion
Asterisk returns a 181 "Call is being forwarded" to Alice.
Asterisk goes back to the dialplan to find Cory.
Asterisk sends an INVITE to Cory with Alice in From/PAI and Bob in
Diversion.
When Cory answers, Asterisk sends back a 200 OK to Alice with Cory in PAI
and Bob in Diversion


>
>
>
>>
>> Best regards
>> --
>> _
>> -- 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
>
>

 --
 George Joseph
 Digium, Inc. | Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com & 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:

Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[Olivier]

2018-06-05 20:29 GMT+02:00 George Joseph :

>
>
> On Tue, Jun 5, 2018 at 10:59 AM Olivier  wrote:
>
>>
>>
>> 2018-06-05 15:27 GMT+02:00 George Joseph :
>> Thank  you very much, George for replying.
>>
>>>
>>>
>>> On Tue, Jun 5, 2018 at 3:35 AM Olivier  wrote:
>>>
 Hi,

 After a long discussion with a friend, I would like to ask here:

 1.According SIP RFCs, is possible/recommended to have different values
 in From and P-Asserted-Id fields ?
 For instance, From field showing 123456789 and P-Asserted-Id showing
 987654321 (beside privacy considerations) ?

>>>
>>> Possible? yes absolutely.
>>>
>>
>> How would you then configure both headers, respectively ?
>>
>> From memory, in previous testings, whenever  CALLERID was set to
>> WHATEVER, P-Asserted-Id was also set to WHATEVER and vice versa, so that I
>> inferred from this that P-Asserted-Id was meant for Privacy
>> considerations and nothing else (see [1])
>>
>
> PAI should be used to indicate the calling party's identification
> regardless of privacy concerns.  In the dialplan you can use the CALLERPRES
> function to control privacy on a call by call basis.
>
>
>
I'm sorry but I still have a doubt ...
Let me re-phrase my question:

My setup is:
Asterisk <--- PJSIP ---> Bob

For a reason, I want Bob's phone to receive a call with the following
headers:

From: "Foo" ;tag=as75ee8c7c
P-Asserted-Id: "Foo" >;whatever

My dialplan is:
same = n,Set(CALLERID(num)=999)
XXX
same = n,Dial(PJSIP/123456@bob)

What shall I replace XXX with to allow me to set 8 in the user part of
P-Asserted-Id URI (see example above) ?
CALLERPRES would change From or P-Asserted-Id but not having different user
parts in URI, would it ?

To my knowledge, a possible way to implement what I'm after is to "turn
off" P-Asserted-Id feature, add a custom P-Asserted-Id header with
PJSIP_HEADER.
Am I missing something ?



>
>
>>
>>
>> [1] https://www.voip-info.org/p-asserted-identity-and-remote-par
>> ty-id-header/
>>
>>
>>> Recommended? who knows?  Implementations are all over the place.  I've
>>> always thought of the From header as identifying the user agent making the
>>> request which kinda agrees with RFC3261.   The PAI header should contain
>>> the identity of the original caller.
>>>
>>>

 2. When Bob forwards to Cory a call coming from Alice, would expect
 Diversion/History-Info header to include Alice's number ?

>>>
>>> No.  The diversion header shows who the diverter is.
>>> https://tools.ietf.org/html/rfc5806
>>>
>>

Thank for  this reference: I think I confused diverter/caller/callee roles
when I first read this document.

So, if Bob forwards to Cory a call from Alice, in which headers would you
expect Alice and Bob numbers to respectively appear ?




>
> Best regards
> --
> _
> -- 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


>>>
>>> --
>>> George Joseph
>>> Digium, Inc. | Software Developer
>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>>> Check us out at: www.digium.com & 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
>>>
>>
>> --
>> _
>> -- 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
>
>
>
> --
> George Joseph
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & 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 

Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[George Joseph]

On Tue, Jun 5, 2018 at 10:59 AM Olivier  wrote:

>
>
> 2018-06-05 15:27 GMT+02:00 George Joseph :
> Thank  you very much, George for replying.
>
>>
>>
>> On Tue, Jun 5, 2018 at 3:35 AM Olivier  wrote:
>>
>>> Hi,
>>>
>>> After a long discussion with a friend, I would like to ask here:
>>>
>>> 1.According SIP RFCs, is possible/recommended to have different values
>>> in From and P-Asserted-Id fields ?
>>> For instance, From field showing 123456789 and P-Asserted-Id showing
>>> 987654321 (beside privacy considerations) ?
>>>
>>
>> Possible? yes absolutely.
>>
>
> How would you then configure both headers, respectively ?
>
> From memory, in previous testings, whenever  CALLERID was set to WHATEVER,
> P-Asserted-Id was also set to WHATEVER and vice versa, so that I inferred
> from this that P-Asserted-Id was meant for Privacy considerations and
> nothing else (see [1])
>

PAI should be used to indicate the calling party's identification
regardless of privacy concerns.  In the dialplan you can use the CALLERPRES
function to control privacy on a call by call basis.




>
>
> [1]
> https://www.voip-info.org/p-asserted-identity-and-remote-party-id-header/
>
>
>> Recommended? who knows?  Implementations are all over the place.  I've
>> always thought of the From header as identifying the user agent making the
>> request which kinda agrees with RFC3261.   The PAI header should contain
>> the identity of the original caller.
>>
>>
>>>
>>> 2. When Bob forwards to Cory a call coming from Alice, would expect
>>> Diversion/History-Info header to include Alice's number ?
>>>
>>
>> No.  The diversion header shows who the diverter is.
>> https://tools.ietf.org/html/rfc5806
>>
>>
>>
>>
>>>
>>> Best regards
>>> --
>>> _
>>> -- 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
>>
>>
>>
>> --
>> George Joseph
>> Digium, Inc. | Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & 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
>>
>
> --
> _
> -- 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



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & 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

Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[Olivier]

2018-06-05 15:27 GMT+02:00 George Joseph :
Thank  you very much, George for replying.

>
>
> On Tue, Jun 5, 2018 at 3:35 AM Olivier  wrote:
>
>> Hi,
>>
>> After a long discussion with a friend, I would like to ask here:
>>
>> 1.According SIP RFCs, is possible/recommended to have different values in
>> From and P-Asserted-Id fields ?
>> For instance, From field showing 123456789 and P-Asserted-Id showing
>> 987654321 (beside privacy considerations) ?
>>
>
> Possible? yes absolutely.
>

How would you then configure both headers, respectively ?

>From memory, in previous testings, whenever  CALLERID was set to WHATEVER,
P-Asserted-Id was also set to WHATEVER and vice versa, so that I inferred
from this that P-Asserted-Id was meant for Privacy considerations and
nothing else (see [1])


[1]
https://www.voip-info.org/p-asserted-identity-and-remote-party-id-header/


> Recommended? who knows?  Implementations are all over the place.  I've
> always thought of the From header as identifying the user agent making the
> request which kinda agrees with RFC3261.   The PAI header should contain
> the identity of the original caller.
>
>
>>
>> 2. When Bob forwards to Cory a call coming from Alice, would expect
>> Diversion/History-Info header to include Alice's number ?
>>
>
> No.  The diversion header shows who the diverter is.
> https://tools.ietf.org/html/rfc5806
>
>
>
>
>>
>> Best regards
>> --
>> _
>> -- 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
>
>
>
> --
> George Joseph
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & 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
>
-- 
_
-- 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] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[George Joseph]

On Tue, Jun 5, 2018 at 3:35 AM Olivier  wrote:

> Hi,
>
> After a long discussion with a friend, I would like to ask here:
>
> 1.According SIP RFCs, is possible/recommended to have different values in
> From and P-Asserted-Id fields ?
> For instance, From field showing 123456789 and P-Asserted-Id showing
> 987654321 (beside privacy considerations) ?
>

Possible? yes absolutely.  Recommended? who knows?  Implementations are all
over the place.  I've always thought of the From header as identifying the
user agent making the request which kinda agrees with RFC3261.   The PAI
header should contain the identity of the original caller.


>
> 2. When Bob forwards to Cory a call coming from Alice, would expect
> Diversion/History-Info header to include Alice's number ?
>

No.  The diversion header shows who the diverter is.
https://tools.ietf.org/html/rfc5806




>
> Best regards
> --
> _
> -- 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



-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & 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

Re: [asterisk-users] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[Daniel Tryba]

On Tue, Jun 05, 2018 at 11:34:51AM +0200, Olivier wrote:
> 1.According SIP RFCs, is possible/recommended to have different values in
> From and P-Asserted-Id fields ?
> For instance, From field showing 123456789 and P-Asserted-Id showing
> 987654321 (beside privacy considerations) ?

Yes, most obviuos need for PAI is a call where anonimity is desired by
caller. Set the From to anonymous@anonymous.invalid and PAI to a real
user if the destination is trusted, any proxy that handles this message
that doesn't trust a destination will strip PAI thus ensuring privacy.
 
> 2. When Bob forwards to Cory a call coming from Alice, would expect
> Diversion/History-Info header to include Alice's number ?

No, diversion/history should contain Bob.

-- 
_
-- 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] Questions about SIP From, P-Asserted-Id fields and Diversion headers ?

[Olivier]

Hi,

After a long discussion with a friend, I would like to ask here:

1.According SIP RFCs, is possible/recommended to have different values in
>From and P-Asserted-Id fields ?
For instance, From field showing 123456789 and P-Asserted-Id showing
987654321 (beside privacy considerations) ?

2. When Bob forwards to Cory a call coming from Alice, would expect
Diversion/History-Info header to include Alice's number ?

Best regards
-- 
_
-- 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] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

[JM or AJS]

On 29/03/17 16:18, Olivier wrote:

Hello,

After reading [1] (in french), I would be very happy if I could get 
answers to:


1. Does this 13.7+20161113-3 package version has any relation with 
asterisk's version it complements ? Current asterisk version in repo 
is 13.14.0. Does this 13.7 complies with it ?
Debian's versioning scheme is all their own.  And I would not expect it 
to work with anything but a Debian-packaged Asterisk.


Stretch is currently the "testing" distribution.  This means that new 
versions of packages could appear at any time; but if a newly-introduced 
package breaks any other packages, they will be removed from "testing"  
(and replaced as soon as possible with newer, compatible versions)  
rather than allow packages to exist in the repository that cannot be 
co-installed.


If you really want to use a newer Asterisk version, the Debian source 
will contain a file called "rules", which is really a Makefile "in 
disguise".  This should give you a good clue as to how to hand-build an 
equivalent based on more up-to-date Source Code (if the compile-time 
options have not changed too much, then you might even get away with 
using it directly, but consider this a bodge).
2. From package description, is this package enough or not to allow 
transcoding with G711 ?

For instance, in the following situation:
SIP Phone < Opus > Asterisk < G111 ---> ITSP
All codecs can input and output raw, uncompressed PCM; so as long as you 
build all the relevant modules, your Asterisk will be able to transcode 
between any two codecs it supports.


(Is "G111" a typo for "G711" ?)
3. Can you share here any personal field experience with this codec, 
for home worker use case ?

Is there a better user experience with Opus than with G729 or G711 ?
Opus is, to the best of my knowledge, fully Open Source.  G729 was 
encumbered by patents in some jurisdictions, though it's now patent-free.


G.711 A-Law is what the PSTN uses natively, and that is unlikely to 
change anytime soon; though some VoIP providers are bringing Opus online 
already.  If you have many phones connected to your Asterisk, then you 
may run into CPU limitations transcoding incoming and outgoing calls 
between G711 and Opus.  But that depends on your Asterisk server.  If 
you are recording calls, Asterisk will already have to convert both the 
incoming and outgoing legs to raw PCM anyway.  In any case, if your 
provider supports Opus, you can offload the donkey work to them .

4. Does it work on ARM boxes (Raspberry, ...) ?
The only thing that would prevent any software from working on ARM / 
Raspberry Pi would be if it
contained any architecture-specific binary code without Source Code 
(which you could just about get away with, if you released it under LGPL 
plus exceptions or an Apache licence).  And I suspect if any such code 
existed, it would be rewritten in fairly short order anyway.


Also, it's Debian; and they really, really don't like binary blobs, only 
grudgingly banishing them to a special "non-free" section which is not 
even enabled by default.  And that package was in the main repository, 
suggesting full Source Code availability.  In any case, I see builds for 
armhf  (R.Pi 1 and 2)  and arm64  (R.Pi 3);  so even if there is some 
sneaky binary-only component, you will be able to get it to work.


[1] https://packages.debian.org/stretch/asterisk-opus

Regards



--

JM or AJS

-- 
_
-- 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] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

[Tzafrir Cohen]

On Wed, Mar 29, 2017 at 06:04:49PM +0200, Olivier wrote:

> Is there any relation between this external patch and the binary mentioned
> in [2]
> [2] http://blogs.digium.com/2016/09/30/opus-in-asterisk/
> 
> The later one mentions a binary-only distribution to comply with legal
> constraints.

No, it is not. This package is in Debian's main archive, which tells you
it is not based on any binary blob.

Opus is widely implemented in software, including free software
(Firefox, Chromium, Linphone, Jitsi and a host of others). See also
https://en.wikipedia.org/wiki/Opus_(audio_format)#Software

My understanding is that to Digium's best legal advice, there are still
patent issues with the Opus codec. Even though many others disagree (as
evident from above) and I also happen to disagree. But I certainly am
not the one who runs Digium. And the powers that be there probably
decided that whatever patent issues there are, have merit and need to be
mitigated.

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

-- 
_
-- 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] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

[Olivier]

2017-03-29 17:28 GMT+02:00 Tzafrir Cohen :

> On Wed, Mar 29, 2017 at 05:18:18PM +0200, Olivier wrote:
> > Hello,
> >
> > After reading [1] (in french), I would be very happy if I could get
> answers
> > to:
> >
> > 1. Does this 13.7+20161113-3 package version has any relation with
> > asterisk's version it complements ? Current asterisk version in repo is
> > 13.14.0. Does this 13.7 complies with it ?
>
> The opus codec was used as an external patch. It looked ugly and thus a
> separate package was preffered.
>

Thank you very much for this informative answer.

Is there any relation between this external patch and the binary mentioned
in [2]
[2] http://blogs.digium.com/2016/09/30/opus-in-asterisk/

The later one mentions a binary-only distribution to comply with legal
constraints.
-- 
_
-- 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] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

[Tzafrir Cohen]

On Wed, Mar 29, 2017 at 05:18:18PM +0200, Olivier wrote:
> Hello,
> 
> After reading [1] (in french), I would be very happy if I could get answers
> to:
> 
> 1. Does this 13.7+20161113-3 package version has any relation with
> asterisk's version it complements ? Current asterisk version in repo is
> 13.14.0. Does this 13.7 complies with it ?

The opus codec was used as an external patch. It looked ugly and thus a
separate package was preffered.

Its version number is not directly related to Asterisk. It has
originally been split from the Debian packaging of Asterisk, and
starting from the same version number allowed easier upgrading. There is
no version number for the upstream code (the patch).

> 
> 2. From package description, is this package enough or not to allow
> transcoding with G711 ?
> For instance, in the following situation:
> SIP Phone < Opus > Asterisk < G111 ---> ITSP

Technically Asterisk codecs translate to/from (typically) linear and
Asterisk combines codecs to do whatever transcoding needed. So the codec
does not transcode directly to G.711. But Asterisk can transcode between
opus and G.711.

> 
> 3. Can you share here any personal field experience with this codec, for
> home worker use case ?
> Is there a better user experience with Opus than with G729 or G711 ?
> 
> 4. Does it work on ARM boxes (Raspberry, ...) ?

Should work just the same.

> 
> 
> [1] https://packages.debian.org/stretch/asterisk-opus


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

-- 
_
-- 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] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

[Olivier]

Hello,

After reading [1] (in french), I would be very happy if I could get answers
to:

1. Does this 13.7+20161113-3 package version has any relation with
asterisk's version it complements ? Current asterisk version in repo is
13.14.0. Does this 13.7 complies with it ?

2. From package description, is this package enough or not to allow
transcoding with G711 ?
For instance, in the following situation:
SIP Phone < Opus > Asterisk < G111 ---> ITSP

3. Can you share here any personal field experience with this codec, for
home worker use case ?
Is there a better user experience with Opus than with G729 or G711 ?

4. Does it work on ARM boxes (Raspberry, ...) ?


[1] https://packages.debian.org/stretch/asterisk-opus

Regards
-- 
_
-- 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] Questions regarding Dial's D option

[Olivier]

Hello,

I'm currently playing with Application Dial D option.
This option is documented with:

D([called][:calling[:progress]]): Send the specified DTMF strings
*after*
the called party has answered, but before the call gets bridged.  The
 DTMF string is sent to the called party, and the  DTMF
string is sent to the calling party.  Both arguments can be used
alone.  If
 is specified, its DTMF is sent to the called party
immediately
after receiving a 'PROGRESS' message.
See 'SendDTMF' for valid digits.


My questions are:

1. Shall I expect those DTMF to be logged (in dtmf logs) with lines such as
the one bellow ? In my testing, it doesn't seem to be the case.

[2017-03-28 12:25:03] DTMF[9943][C-0041]: channel.c:4103 __ast_read:
DTMF begin '#' received on PJSIP/Foobar-004e



2. When my Dial call contains D(#::) option, I'm reading this in Asterisk
console:
-- Sending DTMF '#' to the called party.

When my Dial call contains D(#::progress) option, I'm reading this in
Asterisk console:
-- Sending DTMF 'progress' to the called party as result of receiving a
PROGRESS message.

What is the proper way to send to called party, a DTMF sequence when
Progress tone is received ?
Would you rate "Sending DTMF 'progress' to the called party as result of
receiving a PROGRESS message" as misleading or not ?



3. The service I'm testing the above things with, works this way:
- you dial the service number,
- caller hear an announcement like "next time, to skip legal announcement,
type #",
- if caller effectively dials #, the rest of the announcement is skipped,
- call is answered
- caller is then welcomed with another message and things go on.

My understanding is:
- whatever is done before call is answered is never billed
- as such, providers are likely to forbid anything (either voice or DTMF)
to be sent from caller to callee before call's answering.

Do you agree ?


Best regards
-- 
_
-- 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] Questions... connecting Asterisk to the World

[A J Stiles]

On Saturday 14 May 2016, Stefan Becker wrote:
> Greetings,
> 
> asterisk list and community,
> 
> I have a problem in how our telefon switch (Siemens HiCOM)
> "talks" with my new configured Asterisk server (V.11.18.0)
> 
> without my Asterisks server in the middle
> 
>  <--> Siemens HiCOM <-ISDN-> NTBA <-...-> PBX Telekom
> 
> A phone connected to the switch requests an "Outgoing" line
> by dialing "0". The party is connected via ISDN to
> the carrier (deutsche Telekom) where the party preceeds
> to dial numbers... and the call is connected
> 
> What I can see while I am dialing is that with every
> digit I press it is being displayed on my phone.
> Further more, these digits are being processed by the
> carrier. The call goes through, rings, immediately on
> completion on the number or is rejected if busy.
> 
> 
> WITH my Asterisks server in the middle of the exchange...
> 
> A phone connected to the switch requests an "Outgoing" line
> by dialing "0". -->  Asterisks recieves incoming call on "s".
> The dialed digits are collected. The dial plan is
> executed accordingly but the "caller" recieves no
> more information about the dialed number. The number is
> not placed in the "dialed" numbers simple functions like
> "redial" do not work anymore.
> 
> Does anybody know what I am doing wrong here. Is there a
> way to teach asterisk to behave exactly as if it were the
> PBX (deutsche Telekom).
> So, as to say, act in a way that NO ONE will rightly know
> the differance between having asterisk taking over the
> function of the ISDN PBX.
> 
> What do I need?  A better dial plan to somehow better simulate
> the way the switch normaly behaves?
> Is hardware the problem?
> 
> 
> My ISDN card in the server is:
> "QuadBRI ISDN Digium Wildcard b410P"
> 
> Most everything else functionly works. incoming and outgoing calls
> from and to ISDN, VoIP and other equipment work fine.
> 
> Just that the phones and switch don't recieve the "collected"
> number sequence the was dialed.
> 
> Any help or ideas anyone might have would be greatly appreciated.


Your problem is that you are still thinking in terms of old-fashioned, clicky-
clicky mechanical telephone exchanges.  Instead of "dialling 0 to request an 
outside line", you need to let Asterisk accept all the digits and then 
determine for itself whether the call is going to be an inside or outside one.

- If the user dials 3 digits  (or however long your internal numbers are),  
treat it as an internal number.
- If the user dials 6 digits  (or however long numbers are on your local 
exchange),  treat it as an external, local number.
- If the user dials 11 digits starting with 0  (or however long a number is in 
your country, including the STD code),  treat it as an external, STD number.
- If the number dials 9 or more digits starting with 00, treat it as an 
external, IDD number.


It will make your dialplan a little more complicated; but if it is too simple, 
you won't be taking full advantage of the power of Asterisk.

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

-- 
_
-- 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] Questions... connecting Asterisk to the World

[Steve Edwards]

On Sat, 14 May 2016, Stefan Becker wrote:


On Sat, 14 May 2016, Steve Edwards wrote:

I think you need to make the outbound dial a single 'transaction' 
either by using an extension pattern that includes the 0 like 
'055' to dial 555-555- or eliminate the 0 (and the idiom of 
'requesting an outgoing line') and detect an internal vs external call 
via extension pattern matching.


this is the dialplan that I use:

[ReceiveCallOut]
exten = s,1,Read(LOKAL1,5)
 same =   n,Dial(SIP/${LOKAL}@tt)
 same =   n,Hangup()


I would:

) Drop the 'dial 0' anachronism.

) Not use read().

) Use extension pattern matching.

For example, in the US, I would have something like (off the top of my head):

; external, local
exten = _nxx,1, verbose(1,[${EXTEN}@${CONTEXT})
same = n,   goto(dial-local,${EXTEN},1)
same = n,   hangup()

; external, domestic
exten = _nxxnxx,1,  verbose(1,[${EXTEN}@${CONTEXT})
same = n,   goto(${CONTEXT},1${EXTEN},1)
same = n,   hangup()

; external, domestic
exten = _1nxxnxx,1, verbose(1,[${EXTEN}@${CONTEXT})
same = n,   goto(dial-domestic,${EXTEN},1)
same = n,   hangup()

; international
exten = _011x.,1,   verbose(1,[${EXTEN}@${CONTEXT})
same = n,   goto(dial-international,${EXTEN},1)
same = n,   hangup()

; internal
exten = _[2-9]xxx,1,verbose(1,[${EXTEN}@${CONTEXT})
same = n,   goto(dial-internal,${EXTEN},1)
same = n,   hangup()


When the user dials "0", the HiCOM ISDN switch immediately
goes "online" to the outgoing ISDN Copper Cable - connected
to ... A)  B)

A) connected to the NTBA in the wall jack to the NTBA phone company...
  the dialing preceeds to continue "offline" no dailtones are heard.
  The call is completed and connects


This sounds weird and very foreign (strange and unfamiliar, not as being a 
characteristic of a different country) to me. So, as a caller, I would 
hear the '0' DTMF but no other tones? No feedback as I press keys?



B) connected to the Asterisk ISDN Card

  Asterisk server reacts by executing the above dial plan...


The dialplan does not reflect your intentions.


  CLI > "answered call from "" to "s"

  The user has an open "answered" line and the dialing are collected by
  listening to the DTMF tones.  The generated dial tones can be heard
  on the phone line.

Somehow the signaling on the line of the outgoing call is differant
when the cable is handeled by the PBX or by asterisk.

But why ?


Dialplan and channel configuration.

Can't asterisk be configured to handle a call exactly as the otherwise 
connected phone company's PBX would?


My guess is yes.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

--
_
-- 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] Questions... connecting Asterisk to the World

[Stefan Becker]

On Sat, 14 May 2016, Steve Edwards wrote:

> I think you need to make the outbound dial a single 'transaction' either by
> using an extension pattern that includes the 0 like '055' to dial
> 555-555- or eliminate the 0 (and the idiom of 'requesting an outgoing
> line') and detect an internal vs external call via extension pattern matching.

this is the dialplan that I use:

[ReceiveCallOut]
 exten = s,1,Read(LOKAL1,5)
  same =   n,Dial(SIP/${LOKAL}@tt)
  same =   n,Hangup()

When the user dials "0", the HiCOM ISDN switch immediately
goes "online" to the outgoing ISDN Copper Cable - connected 
to ... A)  B)

A) connected to the NTBA in the wall jack to the NTBA phone company...
   the dialing preceeds to continue "offline" no dailtones are heard.
   The call is completed and connects
   
B) connected to the Asterisk ISDN Card

   Asterisk server reacts by executing the above dial plan...

   CLI > "answered call from "" to "s"

   The user has an open "answered" line and the dialing are collected by
   listening to the DTMF tones.  The generated dial tones can be heard
   on the phone line.


Somehow the signaling on the line of the outgoing call is differant
when the cable is handeled by the PBX or by asterisk.

But why ?

Can't asterisk be configured to handle a call exactly as the 
otherwise connected phone company's PBX would?


thanks for listening,

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] Questions... connecting Asterisk to the World

[Steve Edwards]

On Sat, 14 May 2016, Stefan Becker wrote:

A phone connected to the switch requests an "Outgoing" line by dialing 
"0". --> Asterisks recieves incoming call on "s". The dialed digits are 
collected. The dial plan is executed accordingly but the "caller" 
recieves no more information about the dialed number. The number is not 
placed in the "dialed" numbers simple functions like "redial" do not 
work anymore.


This is not my area of expertise, but I'll throw my $0.02 in...

When you 'request an outgoing line' by dialing 0, that call leg is 
processed by Asterisk, thus, that is what the phone 'sees' as the dialed 
number and that's what the phone will send when 'redial' is pressed.


I think you need to make the outbound dial a single 'transaction' either 
by using an extension pattern that includes the 0 like '055' to 
dial 555-555- or eliminate the 0 (and the idiom of 'requesting an 
outgoing line') and detect an internal vs external call via extension 
pattern matching.


Does your internal extension numbering plan conflict with external 
national numbering plan?


Dialing a prefix digit seems so 1970s to me.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

--
_
-- 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] Questions... connecting Asterisk to the World

[Stefan Becker]

Greetings,

asterisk list and community,

I have a problem in how our telefon switch (Siemens HiCOM)
"talks" with my new configured Asterisk server (V.11.18.0)

without my Asterisks server in the middle

 <--> Siemens HiCOM <-ISDN-> NTBA <-...-> PBX Telekom

A phone connected to the switch requests an "Outgoing" line
by dialing "0". The party is connected via ISDN to
the carrier (deutsche Telekom) where the party preceeds
to dial numbers... and the call is connected

What I can see while I am dialing is that with every 
digit I press it is being displayed on my phone.  
Further more, these digits are being processed by the
carrier. The call goes through, rings, immediately on
completion on the number or is rejected if busy.

 
WITH my Asterisks server in the middle of the exchange...

A phone connected to the switch requests an "Outgoing" line
by dialing "0". -->  Asterisks recieves incoming call on "s".
The dialed digits are collected. The dial plan is 
executed accordingly but the "caller" recieves no 
more information about the dialed number. The number is
not placed in the "dialed" numbers simple functions like
"redial" do not work anymore.

Does anybody know what I am doing wrong here. Is there a
way to teach asterisk to behave exactly as if it were the
PBX (deutsche Telekom).  
So, as to say, act in a way that NO ONE will rightly know
the differance between having asterisk taking over the
function of the ISDN PBX.

What do I need?  A better dial plan to somehow better simulate
the way the switch normaly behaves?
Is hardware the problem?


My ISDN card in the server is:
"QuadBRI ISDN Digium Wildcard b410P"

Most everything else functionly works. incoming and outgoing calls
from and to ISDN, VoIP and other equipment work fine.

Just that the phones and switch don't recieve the "collected" 
number sequence the was dialed.

Any help or ideas anyone might have would be greatly appreciated.


thanks,


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] Questions regarding ICE and STUN with Asterisk

[Kirill Marchuk]

Well, after a more specific research I came to 2 conclusions:

1) no need to specify "stunaddr" option in Asterisk configs in this 
case, as we know that host definitely has a public IP


2) as of the inclusion of all local IP-addresses as candidates, this is 
(apparently) done in "rtp_add_candidates_to_ice" function of 
res_rtp_asterisk.c, where it has a code:

/* Add all the local interface IP addresses */
..

And as fas as I can tell from basic ICE overview [1], this should NOT 
prevent proper session functioning, as long as candidate pairs 
(local/remote) are checked for connectivity first.


Still, I would think it to be useful, to have an option to EXCLUDE local 
IP-addresses from using as local candidates.


What does the community think on this ?

Thanks

Kirill Marchuk

[1] https://trac.pjsip.org/repos/wiki/Using_Standalone_ICE

18.03.2016 14:37, Kirill Marchuk пишет:

Hi everyone

 I would like to get some help and clarification from the experienced 
ones, upon the following:


- we're using Asterisk 13.7.0, that is deployed on a host, that has a 
public IP *and* a couple of gray IPs (192.168.x.x & 10.10.x.x)


- we're using WebRTC web-page (jsSIP) as a client

Which is the proper setup of ICE/STUN related config (on the Asterisk 
and on the client) for the most reliable work in most cases ?


For example, now we're trying to use our own STUN server (from 
Debian's "stund" package), whose documentation says "You have to have 
2 different public IPs on the same server in order to run STUN server"


Is it really so? and what are the implications of using it with only 
one IP (which is possible, at least it runs seemingly well without that)


On the client side, we've configured jsSIP.UA to use our own STUN 
server via "pcConfig" object


On Asterisk, we have icesupport=yes both in sip.conf and rtp.conf. 
We've also enabled stunaddr=stun.l.google.com:19302 in rtp.conf.


Is it proper solution for this case ?

When I inspect SIP packets, I see that there are ICE candidates in 
both offers and answers. BUT: SDP bodies in the packets from server to 
client contain "gray" IPs of the Asterisk host:


a=ice-ufrag:636c49c84158d2b45840291c6724c0f9
a=ice-pwd:6b012c01092ec01275964eaa55a8784b
a=candidate:H904cc6da 1 UDP 2130706431 144.76.x.y 51604 typ host
a=candidate:Ha0a0202 1 UDP 2130706431 10.10.2.2 51604 typ host
a=candidate:S904cc6da 1 UDP 1694498815 144.76.x.y 51604 typ srflx 
raddr 144.76.x.y rport 51604

a=candidate:H904cc6da 2 UDP 2130706430 144.76.x.y 51605 typ host
a=candidate:Ha0a0202 2 UDP 2130706430 10.10.2.2 51605 typ host
a=candidate:S904cc6da 2 UDP 1694498814 144.76.x.y 51605 typ srflx 
raddr 144.76.x.y rport 51605


I am afraid it might be a potential problem, when a client will have 
his private IP in similar subnets. Or am I wrong here ?


So far we are not experiencing any issues, but this seems to be 
alarming..


Can this behaviour (namely, which IP addresses does Asterisk include 
into SDPs body) be configured somehow ?


Many thanks for any help with this question..

Kirill Marchuk



--
_
-- 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] Questions regarding ICE and STUN with Asterisk

[Kirill Marchuk]

Hi everyone

 I would like to get some help and clarification from the experienced 
ones, upon the following:


- we're using Asterisk 13.7.0, that is deployed on a host, that has a 
public IP *and* a couple of gray IPs (192.168.x.x & 10.10.x.x)


- we're using WebRTC web-page (jsSIP) as a client

Which is the proper setup of ICE/STUN related config (on the Asterisk 
and on the client) for the most reliable work in most cases ?


For example, now we're trying to use our own STUN server (from Debian's 
"stund" package), whose documentation says "You have to have 2 different 
public IPs on the same server in order to run STUN server"


Is it really so? and what are the implications of using it with only one 
IP (which is possible, at least it runs seemingly well without that)


On the client side, we've configured jsSIP.UA to use our own STUN server 
via "pcConfig" object


On Asterisk, we have icesupport=yes both in sip.conf and rtp.conf. We've 
also enabled stunaddr=stun.l.google.com:19302 in rtp.conf.


Is it proper solution for this case ?

When I inspect SIP packets, I see that there are ICE candidates in both 
offers and answers. BUT: SDP bodies in the packets from server to client 
contain "gray" IPs of the Asterisk host:


a=ice-ufrag:636c49c84158d2b45840291c6724c0f9
a=ice-pwd:6b012c01092ec01275964eaa55a8784b
a=candidate:H904cc6da 1 UDP 2130706431 144.76.x.y 51604 typ host
a=candidate:Ha0a0202 1 UDP 2130706431 10.10.2.2 51604 typ host
a=candidate:S904cc6da 1 UDP 1694498815 144.76.x.y 51604 typ srflx raddr 
144.76.x.y rport 51604

a=candidate:H904cc6da 2 UDP 2130706430 144.76.x.y 51605 typ host
a=candidate:Ha0a0202 2 UDP 2130706430 10.10.2.2 51605 typ host
a=candidate:S904cc6da 2 UDP 1694498814 144.76.x.y 51605 typ srflx raddr 
144.76.x.y rport 51605


I am afraid it might be a potential problem, when a client will have his 
private IP in similar subnets. Or am I wrong here ?


So far we are not experiencing any issues, but this seems to be alarming..

Can this behaviour (namely, which IP addresses does Asterisk include 
into SDPs body) be configured somehow ?


Many thanks for any help with this question..

Kirill Marchuk

--
_
-- 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] Questions about API Asterisk Java

[pierre.guyard]

Hello everyone,

I have some questions about one of the Asterisk API called Asterisk Java:


* How did it work?

* How can we use it in order to connect one external program to 
Asterisk?

* Can we use it with Asterisk 12?

Thank you in advance,
Pierre

--

Maybe, I have already asked you this lately but I don't know if the 
asterisk-users list and asterisk-users bounces liste are the same, that's why I 
write the message twice.
If they are the same, sorry for the spam!



_

Ce message et ses pieces jointes peuvent contenir des informations 
confidentielles ou privilegiees et ne doivent donc
pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce 
message par erreur, veuillez le signaler
a l'expediteur et le detruire ainsi que les pieces jointes. Les messages 
electroniques etant susceptibles d'alteration,
Orange decline toute responsabilite si ce message a ete altere, deforme ou 
falsifie. Merci.

This message and its attachments may contain confidential or privileged 
information that may be protected by law;
they should not be distributed, used or copied without authorisation.
If you have received this email in error, please notify the sender and delete 
this message and its attachments.
As emails may be altered, Orange is not liable for messages that have been 
modified, changed or falsified.
Thank you.

-- 
_
-- 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] Questions on musiconhold.conf custom mode

[Olivier]

2014-10-27 11:50 GMT+01:00 Thorsten Göllner t...@ovm-group.com:

 Am 27.10.2014 08:54, schrieb Olivier:
 2014-10-25 19:33 GMT+02:00 Thorsten Göllner t...@ovm-group.com:
 Am 25.10.2014 00:09, schrieb Olivier:
 Hello,

 I need to play some musiconhold content starting at a random duration
 from the start.

 Thanks to mode=custom option and either madplay or mpg123 programs, I
 could successfully get what I was after on a Debian Wheezy system.

 Now I realized sox version on my target system (Debian Squeeze) cannot
 convert to MP3 format.
 So I'm looking after workarounds.

 0. I've read many  mpg123 or madplay examples. All of them are
 clutered with option converting MP3 input file into an appropriate
 format that Asterisk requires for music on hold.
 What is the name of this appropriate format ? sln ? wav ?

 1. Is there a player like mpg123, that can repeat content in
 appropriate format (see above)  to stdout but can read from anything
 different from MP3 ?

 2. Is there an option on Squeeze to convert audio files to MP3
 (reverse coversion works OK).

 3. Which options could I have for such custom MOH, if I was building
 on system without g729 transaltion capabilites ans with g729-only SIP
 trunks or phones ?

 Is the gsm-format an option for you? So you may convert your moh-File to
 gsm:
 sox YouWavFile.wav -r 8000 -c1 MohFile.gsm
 Hi Thorsten,

 Yes gsm-format is an option for me but how can you play such gsm file as MOH 
 ?

 If I'm not mistaken, both madplay or mpg123 would only play MP3 files
 (I've not tested with other formats, yet).
 I could successfully play a RAW file with cat but cat has no repeat
 option, so I still have to find something else anyway.

 When your musiconhold.conf looks like that ...

  cut -
 [general]

 [default]
 mode=files
 directory=moh

 [your_moh_class]
 mode=files
 directory=/your/path/to/your/moh/files
  cut -

Yes this is true but when you need your MOH to start randomly (ie not
start from the very start but from anywhere within your MOH file), you
need to switch to custom mode and customize application parameter.

In this specific case, I didn't find many options avoiding MP3 files.

 ... then you can put any supported file format into the specified
 directory. GSM is only one option. Asterisk will take the best (meaning
 cheapest) file format availble in this directory.




 If you really need mp3 you have to compile sox with mp3-support by
 yourself OR maybe this is a solution on Debian:
 http://www.howtoinstall.co/en/debian/wheezy/main/libsox-fmt-mp3/
 Yes, you're correct.
 I'll suggest my customer a Wheezy upgrade.

 --
 _
 -- 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] Questions on musiconhold.conf custom mode

[Olivier]

2014-10-25 19:33 GMT+02:00 Thorsten Göllner t...@ovm-group.com:

 Am 25.10.2014 00:09, schrieb Olivier:
 Hello,

 I need to play some musiconhold content starting at a random duration
 from the start.

 Thanks to mode=custom option and either madplay or mpg123 programs, I
 could successfully get what I was after on a Debian Wheezy system.

 Now I realized sox version on my target system (Debian Squeeze) cannot
 convert to MP3 format.
 So I'm looking after workarounds.

 0. I've read many  mpg123 or madplay examples. All of them are
 clutered with option converting MP3 input file into an appropriate
 format that Asterisk requires for music on hold.
 What is the name of this appropriate format ? sln ? wav ?

 1. Is there a player like mpg123, that can repeat content in
 appropriate format (see above)  to stdout but can read from anything
 different from MP3 ?

 2. Is there an option on Squeeze to convert audio files to MP3
 (reverse coversion works OK).

 3. Which options could I have for such custom MOH, if I was building
 on system without g729 transaltion capabilites ans with g729-only SIP
 trunks or phones ?


 Is the gsm-format an option for you? So you may convert your moh-File to
 gsm:
 sox YouWavFile.wav -r 8000 -c1 MohFile.gsm

Hi Thorsten,

Yes gsm-format is an option for me but how can you play such gsm file as MOH ?

If I'm not mistaken, both madplay or mpg123 would only play MP3 files
(I've not tested with other formats, yet).
I could successfully play a RAW file with cat but cat has no repeat
option, so I still have to find something else anyway.



 If you really need mp3 you have to compile sox with mp3-support by
 yourself OR maybe this is a solution on Debian:
 http://www.howtoinstall.co/en/debian/wheezy/main/libsox-fmt-mp3/

Yes, you're correct.
I'll suggest my customer a Wheezy upgrade.




 -Thorsten-

 --
 _
 -- 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] Questions on musiconhold.conf custom mode

[Thorsten Göllner]

Am 27.10.2014 08:54, schrieb Olivier:
 2014-10-25 19:33 GMT+02:00 Thorsten Göllner t...@ovm-group.com:
 Am 25.10.2014 00:09, schrieb Olivier:
 Hello,

 I need to play some musiconhold content starting at a random duration
 from the start.

 Thanks to mode=custom option and either madplay or mpg123 programs, I
 could successfully get what I was after on a Debian Wheezy system.

 Now I realized sox version on my target system (Debian Squeeze) cannot
 convert to MP3 format.
 So I'm looking after workarounds.

 0. I've read many  mpg123 or madplay examples. All of them are
 clutered with option converting MP3 input file into an appropriate
 format that Asterisk requires for music on hold.
 What is the name of this appropriate format ? sln ? wav ?

 1. Is there a player like mpg123, that can repeat content in
 appropriate format (see above)  to stdout but can read from anything
 different from MP3 ?

 2. Is there an option on Squeeze to convert audio files to MP3
 (reverse coversion works OK).

 3. Which options could I have for such custom MOH, if I was building
 on system without g729 transaltion capabilites ans with g729-only SIP
 trunks or phones ?

 Is the gsm-format an option for you? So you may convert your moh-File to
 gsm:
 sox YouWavFile.wav -r 8000 -c1 MohFile.gsm
 Hi Thorsten,

 Yes gsm-format is an option for me but how can you play such gsm file as MOH ?

 If I'm not mistaken, both madplay or mpg123 would only play MP3 files
 (I've not tested with other formats, yet).
 I could successfully play a RAW file with cat but cat has no repeat
 option, so I still have to find something else anyway.

When your musiconhold.conf looks like that ...

 cut -
[general]

[default]
mode=files
directory=moh

[your_moh_class]
mode=files
directory=/your/path/to/your/moh/files
 cut -

... then you can put any supported file format into the specified
directory. GSM is only one option. Asterisk will take the best (meaning
cheapest) file format availble in this directory.




 If you really need mp3 you have to compile sox with mp3-support by
 yourself OR maybe this is a solution on Debian:
 http://www.howtoinstall.co/en/debian/wheezy/main/libsox-fmt-mp3/
 Yes, you're correct.
 I'll suggest my customer a Wheezy upgrade.

-- 
_
-- 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] Questions on musiconhold.conf custom mode

[Thorsten Göllner]

Am 25.10.2014 00:09, schrieb Olivier:
 Hello,

 I need to play some musiconhold content starting at a random duration
 from the start.

 Thanks to mode=custom option and either madplay or mpg123 programs, I
 could successfully get what I was after on a Debian Wheezy system.

 Now I realized sox version on my target system (Debian Squeeze) cannot
 convert to MP3 format.
 So I'm looking after workarounds.

 0. I've read many  mpg123 or madplay examples. All of them are
 clutered with option converting MP3 input file into an appropriate
 format that Asterisk requires for music on hold.
 What is the name of this appropriate format ? sln ? wav ?

 1. Is there a player like mpg123, that can repeat content in
 appropriate format (see above)  to stdout but can read from anything
 different from MP3 ?

 2. Is there an option on Squeeze to convert audio files to MP3
 (reverse coversion works OK).

 3. Which options could I have for such custom MOH, if I was building
 on system without g729 transaltion capabilites ans with g729-only SIP
 trunks or phones ?


Is the gsm-format an option for you? So you may convert your moh-File to
gsm:
sox YouWavFile.wav -r 8000 -c1 MohFile.gsm

If you really need mp3 you have to compile sox with mp3-support by
yourself OR maybe this is a solution on Debian:
http://www.howtoinstall.co/en/debian/wheezy/main/libsox-fmt-mp3/


-Thorsten-

-- 
_
-- 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] Questions on musiconhold.conf custom mode

[Olivier]

Hello,

I need to play some musiconhold content starting at a random duration
from the start.

Thanks to mode=custom option and either madplay or mpg123 programs, I
could successfully get what I was after on a Debian Wheezy system.

Now I realized sox version on my target system (Debian Squeeze) cannot
convert to MP3 format.
So I'm looking after workarounds.

0. I've read many  mpg123 or madplay examples. All of them are
clutered with option converting MP3 input file into an appropriate
format that Asterisk requires for music on hold.
What is the name of this appropriate format ? sln ? wav ?

1. Is there a player like mpg123, that can repeat content in
appropriate format (see above)  to stdout but can read from anything
different from MP3 ?

2. Is there an option on Squeeze to convert audio files to MP3
(reverse coversion works OK).

3. Which options could I have for such custom MOH, if I was building
on system without g729 transaltion capabilites ans with g729-only SIP
trunks or phones ?

Regards

-- 
_
-- 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] Questions about chan_dahdi, PRI, MWI (and Q.SIG)

[Jens Bürger]

Hello everyone,

My setup:
Debian squeeze
Asterisk 1.8, DAHDI, libpri, compiled from source
TE110P, attached to a Deutsche Telekom Octopus E Modell 300/800

I'm trying to get MWI for Voicemail working. In the same server I have 
also got an Eicon DIVA PRI card for testing purposes (it is integrated 
via CAPI and the chan-capi channel driver into my Asterisk). MWI works 
just fine there.


I read through chan_dahdi.conf and have some questions:

1. The documentation of
mwi_mailboxes
says: You can give a comma separated list of up to 8 mailboxes per span.

Is this constraint really existing? How am I supposed to use the MWI 
feature in even a semi-professional environment? My PBX is used in a 
non-commercial project, but I have connected about 50 phones to my PBX, 
though, and it is interconnected with two further PBXs which would also 
need voicemail with MWI...


2. How can I set the MWI origin number?

3. Are there any debug possibilities for MWI?

Thanks in advance,

Jens

--
_
-- 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] Questions about chan_dahdi, PRI, MWI (and Q.SIG)

[Richard Mudgett]

On Fri, Jun 28, 2013 at 3:59 AM, Jens Bürger jbuer...@arcor.de wrote:

 Hello everyone,

 My setup:
 Debian squeeze
 Asterisk 1.8, DAHDI, libpri, compiled from source
 TE110P, attached to a Deutsche Telekom Octopus E Modell 300/800

 I'm trying to get MWI for Voicemail working. In the same server I have
 also got an Eicon DIVA PRI card for testing purposes (it is integrated via
 CAPI and the chan-capi channel driver into my Asterisk). MWI works just
 fine there.

 I read through chan_dahdi.conf and have some questions:

 1. The documentation of
 mwi_mailboxes
 says: You can give a comma separated list of up to 8 mailboxes per span.

 Is this constraint really existing? How am I supposed to use the MWI
 feature in even a semi-professional environment? My PBX is used in a
 non-commercial project, but I have connected about 50 phones to my PBX,
 though, and it is interconnected with two further PBXs which would also
 need voicemail with MWI...


This option is intended for BRI spans communicating to a group of ISDN
phones connected to that span and the Asterisk server is handling voicemail
itself.


 2. How can I set the MWI origin number?

 3. Are there any debug possibilities for MWI?


Other than the limited support for sending MWI messages to ISDN phones on
BRI lines there is no other ISDN MWI support.

Richard
--
_
-- 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] Questions about sRTP

[Mike Diehl]

Hi all,

I'm getting ready to setup SIP/TLS and SRTP.  But I have a few questions.
The first one is that I was reading an article at:

https://supportforums.cisco.com/docs/DOC-15381

That indicated that Asterisk doesn't support TLS as an OPTIONAL transport.
It's either all or nothing.  Specifically, this is what it said:

==
*Note: There is no optional SRTP mode in Asterisk, i.e. if encryption is
active on peer, it will not accept non-ciphered audio and viceversa. On the
IP phones, however, it is possible to have unsecure calls if the other peer
does not support SRTP, i.e. incoming calls may work, but not outgoing
calls. This is an Asterisk limitation (Snom supports also the
“optional”mode on SRTP sending two m=audio attributes, but Asterisk does
not know how to handle those descriptors).*
==

This is from a quite dated article (2011), so I'm hoping that I newer
versions of Asterisk will fall back on plaintext if TLS isn't available for
some reason.

Secondly, is there any way to detect if a call is secure from inside the
dialplan or AGI script?

I think that's all for now.

Thanks in advance,

Mike Diehl.
--
_
-- 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] Questions about sRTP

[Joshua Colp]

Mike Diehl wrote:

Hi all,

I'm getting ready to setup SIP/TLS and SRTP.  But I have a few
questions.  The first one is that I was reading an article at:

https://supportforums.cisco.com/docs/DOC-15381

That indicated that Asterisk doesn't support TLS as an OPTIONAL
transport.  It's either all or nothing.  Specifically, this is what it said:


Your statement is incorrect. Asterisk supports TLS as an optional 
signaling transport (although if you do SDES SRTP without it then 
someone can snoop on your keys and ultimately decrypt your media).


What it does not support is optional *SRTP*. If a device requests SRTP 
and it's not possible, the call will fail.


--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com   www.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] Questions about sRTP

[Mike Diehl]

On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp jc...@digium.com wrote:

 Mike Diehl wrote:

 Hi all,

 I'm getting ready to setup SIP/TLS and SRTP.  But I have a few
 questions.  The first one is that I was reading an article at:

 https://supportforums.cisco.com/docs/DOC-15381

 That indicated that Asterisk doesn't support TLS as an OPTIONAL
 transport.  It's either all or nothing.  Specifically, this is what it
 said:


 Your statement is incorrect. Asterisk supports TLS as an optional
 signaling transport (although if you do SDES SRTP without it then someone
 can snoop on your keys and ultimately decrypt your media).

 What it does not support is optional *SRTP*. If a device requests SRTP and
 it's not possible, the call will fail.


So then, is it safe to say that Asterisk will ALLOW a secure phone call,
but the client hast to REQUEST it?

I understand that requesting SRTP without SIP/TLS is evil; I just
misunderstood what I was reading.

I'm also thinking that the AGI script I use to route calls can check if
either leg of a call comes from or goes to port 5061 and play a sound file
to indicate that the cal is 'secure.'  Does this seem reasonable?

Thanks,

Mike.
--
_
-- 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] Questions about sRTP

[Matthew Jordan]

On Thu, Jun 20, 2013 at 5:10 PM, Mike Diehl mdiehlena...@gmail.com wrote:



 On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp jc...@digium.com wrote:

 Mike Diehl wrote:

 Hi all,

 I'm getting ready to setup SIP/TLS and SRTP.  But I have a few
 questions.  The first one is that I was reading an article at:

 https://supportforums.cisco.com/docs/DOC-15381

 That indicated that Asterisk doesn't support TLS as an OPTIONAL
 transport.  It's either all or nothing.  Specifically, this is what it
 said:


 Your statement is incorrect. Asterisk supports TLS as an optional
 signaling transport (although if you do SDES SRTP without it then someone
 can snoop on your keys and ultimately decrypt your media).

 What it does not support is optional *SRTP*. If a device requests SRTP
 and it's not possible, the call will fail.


 So then, is it safe to say that Asterisk will ALLOW a secure phone call,
 but the client hast to REQUEST it?

 I understand that requesting SRTP without SIP/TLS is evil; I just
 misunderstood what I was reading.

 I'm also thinking that the AGI script I use to route calls can check if
 either leg of a call comes from or goes to port 5061 and play a sound file
 to indicate that the cal is 'secure.'  Does this seem reasonable?


You can query a channel using the CHANNEL function (
https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL) to see if the
channel currently supports secure communication, and you can request that
the outbound channel be made secure using the same function.

An example of doing this is on the wiki:

https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Specifics

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

[asterisk-users] Questions about extension.conf

[Shitian Long]

Hello 

I have been reading the sample extension.conf

;###


[outbound-freenum2]
; This is the handler which performs the dialing logic. It is called
; from the [outbound-freenum] context
;
exten = _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same = n,Set(SUFFIX=${CUT(EXTEN,*,2-)}); make 
sure the suffix is all digits as well
same = n,GotoIf($[${FILTER(0-9,${SUFFIX})} != ${SUFFIX}]?fn-CONGESTION,1)
; 
filter out bad characters per the README-SERIOUSLY.best-practices.txt document
same = n,Set(TIMEOUT(absolute)=10800)
same = n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ; 
perform our lookup with freenum.org
same = n,GotoIf($[${isnresult} != ]?from)
same = n,Set(DIALSTATUS=CONGESTION)
same = n,Goto(fn-CONGESTION,1)
same = n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same = n,GotoIf($[${GLOBAL(FREENUMDOMAIN)} = ]?dial)   ; check 
if we set the FREENUMDOMAIN global variable in [global]
same = n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ;if 
we did set it, then we'll use it for our outbound dialing domain
same = n(dial),Dial(SIP/${isnresult},40)
same = n,Goto(fn-${DIALSTATUS},1)

exten = fn-BUSY,1,Busy()

exten = _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same = n,Congestion()

;##


According to 
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf;

Syntax for defining a context: keywords exten, include, ignorepat and switch. 
same is not mentioned in this wiki. 

There is a part of dial plan from sample extension.conf above. My Question is  
how same = key word works . 

Thanks


--
_
-- 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] Questions about extension.conf

[Danny Nicholas]

As I understand it, same = is a way to shorthand your list of the other
keywords. In the example you posted, you save 4 keystrokes for each line you
enter; not a lot of savings for this short example, but put it in a 1000+
line dialplan and it's quite a time-saver.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shitian Long
Sent: Thursday, November 29, 2012 10:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Questions about extension.conf

 

Hello 

 

I have been reading the sample extension.conf

 

;###

 

 

[outbound-freenum2]

; This is the handler which performs the dialing logic. It is called

; from the [outbound-freenum] context

;

exten = _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})

same = n,Set(SUFFIX=${CUT(EXTEN,*,2-)});
make sure the suffix is all digits as well

same = n,GotoIf($[${FILTER(0-9,${SUFFIX})} !=
${SUFFIX}]?fn-CONGESTION,1)

;
filter out bad characters per the README-SERIOUSLY.best-practices.txt
document

same = n,Set(TIMEOUT(absolute)=10800)

same = n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ;
perform our lookup with freenum.org

same = n,GotoIf($[${isnresult} != ]?from)

same = n,Set(DIALSTATUS=CONGESTION)

same = n,Goto(fn-CONGESTION,1)

same = n(from),Set(__SIPFROMUSER=${CALLERID(num)})

same = n,GotoIf($[${GLOBAL(FREENUMDOMAIN)} = ]?dial)   ;
check if we set the FREENUMDOMAIN global variable in [global]

same = n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ;
if we did set it, then we'll use it for our outbound dialing domain

same = n(dial),Dial(SIP/${isnresult},40)

same = n,Goto(fn-${DIALSTATUS},1)

 

exten = fn-BUSY,1,Busy()

 

exten = _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})

same = n,Congestion()

 

;##

 

 

According to
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf;

 

Syntax for defining a context: keywords exten, include, ignorepat and
switch. same is not mentioned in this wiki. 

 

There is a part of dial plan from sample extension.conf above. My Question
is  how same = key word works . 

 

Thanks

 

 

--
_
-- 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] Questions about extension.conf

[Mikhail Lischuk]

 

Shitian Long wrote 29.11.2012 18:40: 

 There is a part of dial
plan from sample extension.conf above. My Question is how same = key
word works . 
 
 Thanks

same is used for complex templates, if you
don't want to copy previous line or afraid you can make a typo. 

exten
= _1XXNXXX,1,Answer 

same = n,HangUp 

is the substitution for:


exten = _1XXNXXX,1,Answer 

exten = _1XXNXXX,n,HangUp 

Also, it
makes grepping the particular exten in a file a lot easier, and if you
want to change some template for exten which has 50 lines, you don't
have to edit all 50 of them. 

-- 
With Best Regards
Mikhail Lischuk

 --
_
-- 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] Questions about extension.conf

[Ron Wheeler]

On 29/11/2012 11:47 AM, Salman Zafar wrote:

It is self explanatory, for example:

exten =  _X.,1, Noop(Let say we have allowed all numbers i.e. _X 
means and . specifies any range)
same = n,NoOp(Here we have skipped mentioning dial-pattern again and 
thats it)



Hope I have answered your question.

Not for me.
What part of those lines and comments discusses same?

What is the syntax for a same line? what does it mean to use same 
rather than exten?






On Thu, Nov 29, 2012 at 8:40 AM, Shitian Long longst...@gmail.com 
mailto:longst...@gmail.com wrote:


Hello

I have been reading the sample extension.conf

;###


[outbound-freenum2]
; This is the handler which performs the dialing logic. It is called
; from the [outbound-freenum] context
;
exten = _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same = n,Set(SUFFIX=${CUT(EXTEN,*,2-)})  ;
make sure the suffix is all digits as well
same = n,GotoIf($[${FILTER(0-9,${SUFFIX})} !=
${SUFFIX}]?fn-CONGESTION,1)
  ; filter out bad characters per the
README-SERIOUSLY.best-practices.txt document
same = n,Set(TIMEOUT(absolute)=10800)
same = n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org
http://freenum.org)}) ; perform our lookup with freenum.org
http://freenum.org
same = n,GotoIf($[${isnresult} != ]?from)
same = n,Set(DIALSTATUS=CONGESTION)
same = n,Goto(fn-CONGESTION,1)
same = n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same = n,GotoIf($[${GLOBAL(FREENUMDOMAIN)} = ]?dial)
  ; check if we set the FREENUMDOMAIN global variable in [global]

same = n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) ;
   if we did set it, then we'll use it for our outbound dialing domain
same = n(dial),Dial(SIP/${isnresult},40)
same = n,Goto(fn-${DIALSTATUS},1)

exten = fn-BUSY,1,Busy()

exten = _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same = n,Congestion()

;##


According to
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf;

Syntax for defining a context: keywords *exten*, *include*,
*ignorepat* and *switch*. same is not mentioned in this wiki.

There is a part of dial plan from sample extension.conf above. My
Question is  how same = key word works .

Thanks



--
_
-- 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




--
Regards

**
Muhammad Salman
***



--
_
-- 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



--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

--
_
-- 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] Questions about extension.conf

[Ron Wheeler]

That is a good answer.
Thanks.
Any reason why it is not documented?

Ron

On 29/11/2012 11:52 AM, Mikhail Lischuk wrote:


Shitian Long wrote 29.11.2012 18:40:

There is a part of dial plan from sample extension.conf above. My 
Question is  how same = key word works .

Thanks
  


same is used for complex templates, if you don't want to copy 
previous line or afraid you can make a typo.


exten = _1XXNXXX,1,Answer

same = n,HangUp

is the substitution for:

exten = _1XXNXXX,1,Answer

exten = _1XXNXXX,n,HangUp

Also,  it makes grepping the particular exten in a file a lot easier, 
and if you want to change some template for exten which has 50 lines, 
you don't have to edit all 50 of them.


--
With Best Regards
Mikhail Lischuk  mailto:mlisc...@itx.com.ua

  



--
_
-- 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



--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

--
_
-- 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] Questions about extension.conf

[Eric Wieling]

The Wiki is (always) out of date.  You might consider taking a look at 
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#DialplanBasics_id262049
 which is likely less out of data.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Thursday, November 29, 2012 12:17 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Questions about extension.conf

On 29/11/2012 11:47 AM, Salman Zafar wrote:


It is self explanatory, for example:

exten =  _X.,1, Noop(Let say we have allowed all numbers i.e. _X 
means and . specifies any range)
same = n,NoOp(Here we have skipped mentioning dial-pattern again and 
thats it)


Hope I have answered your question. 


Not for me.
What part of those lines and comments discusses same?

What is the syntax for a same line? what does it mean to use same rather 
than exten?






On Thu, Nov 29, 2012 at 8:40 AM, Shitian Long longst...@gmail.com 
wrote:


Hello  

I have been reading the sample extension.conf

;###


[outbound-freenum2]
; This is the handler which performs the dialing logic. It is 
called
; from the [outbound-freenum] context
;
exten = _X!,1,Verbose(2,Performing ISN lookup for ${EXTEN})
same = n,Set(SUFFIX=${CUT(EXTEN,*,2-)})
; make sure the suffix is all digits as well
same = n,GotoIf($[${FILTER(0-9,${SUFFIX})} != 
${SUFFIX}]?fn-CONGESTION,1)

; filter out bad characters per the README-SERIOUSLY.best-practices.txt 
document
same = n,Set(TIMEOUT(absolute)=10800)
same = 
n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) ; perform our 
lookup with freenum.org
same = n,GotoIf($[${isnresult} != ]?from)
same = n,Set(DIALSTATUS=CONGESTION)
same = n,Goto(fn-CONGESTION,1)
same = n(from),Set(__SIPFROMUSER=${CALLERID(num)})
same = n,GotoIf($[${GLOBAL(FREENUMDOMAIN)} = ]?dial)   
; check if we set the FREENUMDOMAIN global variable in [global]
same = n,Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) 
;if we did set it, then we'll use it for our outbound dialing domain
same = n(dial),Dial(SIP/${isnresult},40)
same = n,Goto(fn-${DIALSTATUS},1)

exten = fn-BUSY,1,Busy()

exten = _f[n]-.,1,NoOp(ISN: ${DIALSTATUS})
same = n,Congestion()

;##


According to 
http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf;

Syntax for defining a context: keywords exten, include, 
ignorepat and switch. same is not mentioned in this wiki. 

There is a part of dial plan from sample extension.conf above. 
My Question is  how same = key word works . 

Thanks



--

_
-- 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





-- 
Regards 


**
Muhammad Salman
***


 

--
_
-- 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



-- 
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

--
_
-- 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

Re: [asterisk-users] Questions about extension.conf

[David M. Lee]

On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote:

 That is a good answer.
 Thanks.
 Any reason why it is not documented?

It's documented on the Asterisk wiki:
  
https://wiki.asterisk.org/wiki/display/AST/Contexts,+Extensions,+and+Priorities

 Ron

-- 
David M. Lee
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com   www.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] Questions about extension.conf

[Ron Wheeler]

Excellent.
It appears that Getting Started has a lot more stuff in it than the 
documentation for 1.8.


Very helpful.

Ron

On 29/11/2012 12:31 PM, David M. Lee wrote:


On Nov 29, 2012, at 11:18 AM, Ron Wheeler wrote:


That is a good answer.
Thanks.
Any reason why it is not documented?


It's documented on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Contexts,+Extensions,+and+Priorities


Ron


--
David M. Lee
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com http://www.digium.com/   
www.asterisk.org http://www.asterisk.org/





--
Ron Wheeler
President
Artifact Software Inc
email: rwhee...@artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

--
_
-- 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] Questions on converting to ConfBridge

[kenner]

 Why are you wanting to use CLI commands instead of AMI? The available 
 AMI actions for ConfBridge can do listing/locking/muting/kicking etc as 
 you want.

Because I can't easily manually do an AMI command, but instead have to
write code to do it.  It's important to me to be able to clean up things
from the command-line if something is stuck or broken.

 As for dialplan applications to do the various things - what are you 
 trying to achieve using them?

And IVR application that people can call into and manipulate people in
conference rooms.  Note that this depends on dialplan commands *and*
having a number index for them.  It's unclear how I'd do this with
confbridge.

Here's the dialplan I'm using.

exten = 210/_[12]XX,1,NoOp ; Valid if internal.
exten = 210,s,Gosub(Authenticate,s,1()); Else authenticate.
 same = n,Mset(C=conferenceha/roomdigits/2digits/0,E=adacore/not-exist)
 same = n,Mset(STATS_INC(conf_mgr)=1,__G=conf_op) ; Count the usage.
 same = n(r),Macro(Get-Speech,${G},${EFN}adacore/conf_mgr,2,10,100,w)
 same = n,GotoIf(${S_T}?${S_T},1:r); Retry or do action.

exten = _[lLK]20Z,1,GotoIf(${MEETME_INFO(parties,20${EXTEN:-1})}?:err)
exten = _L20Z,n,Mset(V=lock,T=locked,E=isadacore/already-locked)
exten = _l20Z,s,Mset(V=unlock,T=unlocked,E=isadacore/already-unlocked)
exten = _K20Z,s,Mset(V=terminate,T=terminated) ; To terminate.
exten = 
_[lLK]20Z,n,Set(CFN=adacore/you-want-toadacore/${V}${C}digits/${EXTEN:-1})
exten = _[lLK]20Z,n,Gosub(Is-That-Correct,s,1) ; ... and see if correct.
exten = _[lLK]20Z,n,GotoIf($[${GOSUB_RETVAL}=2]?210,r) ; Retry it not.
exten = _L20Z,n,GotoIf(${MEETME_INFO(lock,20${EXTEN:-1})}?err) ; Bad status.
exten = _l20Z,s,GotoIf(${MEETME_INFO(lock,20${EXTEN:-1})}?:err) ; Likewise.
exten = _K20Z,s,NoOp   ; No test needed for termination.
exten = _[lLK]20Z,n,MeetMeAdmin(${EXTEN:-3},${EXTEN:0:1}) ; Perform op.
exten = _[lLK]20Z,n,Set(EFN=${C}digits/${EXTEN:-1}isnowadacore/${T})
exten = _[lLK]20Z,n,Goto(210,r); Ask for another operation.

exten = _[lLK]20Z,n(err),Set(EFN=im-sorry${C}digits/${EXTEN:-1}${E})
exten = _[lLK]20Z,n,Goto(210,r); See if another operation is wanted.

exten = _s20Z,1,Goto(s20${EXTEN:-1}${MEETME_INFO(parties,${EXTEN:-3})},1)
exten = _s20Z.,1,Playback(${C}digits/${EXTEN:3:1}) ; Say that conference ...
exten = _s20Z0,n,Playback(adacore/not-exist)   ; ... doesn't exist,
exten = _s20Z1,s,Swift(has one participant); ... or has one person,
exten = _s20Z.,s,Swift(has ${EXTEN:4} participants) ; ... or more.
exten = _s20Z.,n,ExecIf(${MEETME_INFO(lock,${EXTEN:1})}?Swift(and is locked)
exten = _s20Z.,n,Set(M=$[CEIL(MEETME_INFO(activity,${EXTEN:1:3})/60)])
exten = _s20Z[1-9]!,n,Swift(and has been active for ${M} minutes)
exten = _s20Z.,s,NoOp  ; In other cases, do nothing.
exten = _s20Z.,n,Goto(210,r)   ; Go back for another operation.

exten = _j20Z,1,Set(CFN=you-wish-to-join${C}digits/${EXTEN:-1})
 same = n,Gosub(Is-That-Correct,s,1) ; See if correct.
 same = n,GotoIf($[${GOSUB_RETVAL}=2]?210,r) ; Retry it not.
 same = n,SpeechDestroy; Else free speech channel.
 same = n,Goto(${EXTEN:-3},1)  ; And go there.

exten = 
_[pP]20Z,1,GotoIf($[MEETME_INFO(parties,${EXTEN:1})=0]?s20${EXTEN:-1}0,1)
 same = n,Swift(participants in)   ; Say the header and ...
 same = n,Playback(${C}digits/${EXTEN:3:1})   ; ... conference number.
 same = n,ExecIf($[x${G:0:3}=xtmp]?System(rm -f ${GRAMS}/${G}.gram))
 same = n,Set(__G=tmp/r${RAND(1,9)})   ; Grammar filename part.
 same = n,AGI(conflist.php,${EXTEN:1},${GRAMS}/${G}.gram,${EXTEN:0:1})
 same = n,Goto(210,r)  ; And go back.

exten = _m20ZXX.,1,Mset(Q=adacore/unmute,OP=adacore/unmuted)
exten = _M20ZXX.,s,Mset(Q=adacore/mute,OP=adacore/muted)
exten = _k20ZXX.,s,Mset(Q=adacore/remove,OP=removed)
exten = _[Mmk]20ZXX.,n,Playback(adacore/you-want-to${Q}) ; Start question.
 same = n,Swift(${EXTEN:6}); Say who ...
 same = n,Gosub(Is-That-Correct,s,1)   ; ... and see if correct.
 same = n,GotoIf($[${GOSUB_RETVAL}=2]?210,r)   ; Retry it not.
 same = n,Set(U=${IF($[${EXTEN:4:1}=0]?${EXTEN:5:1}:${EXTEN:4:2})})
 same = n,MeetMeAdmin(${EXTEN:1:3},${EXTEN:0:1},${U}) ; Do operation.
 same = n,Swift(${EXTEN:6}); Say name ...
 same = n,Playback(${OP})  ; ... and what we did.
 same = n,Goto(210,r)  ; Go back for another try.

exten = What,1,Set(EFN=adacore/confop_what) ; Say what options are available.
 same = n,Goto(210,r)  ; And go back and prompt again.
exten = Done,1,Playback(vm-goodbye); Here to hangup.

Here's the grammar:

#ABNF 1.0 UTF-8;
language en-US;
mode voice;
tag-format semantics/1.0.2006;
root $conf_op;

$Operation = lock {out = L;}
| unlock {out = l;}
| (end | kill | terminate) {out = K;}
| join { out = j;}
| [(say | get)] status [of] { 

Re: [asterisk-users] Questions on converting to ConfBridge

[Leif Madsen]

On 02/10/12 06:07 PM, Richard Kenner wrote:

I'm looking at what would be involved in converting from MeetMe to
ConfBridge and there seems to be a lot of missing administrative things,
but I hope I'm just missing it.



There also doesn't
seem to be a way to lock conferences or mute or kick out users from
the dialplan.

What am I missing?


You're missing the custom DTMF based menus in confbridge.conf, which 
allows you to set menus separately for admins and users of the 
conference bridge. This menu allows you to control kicking, muting, etc 
of users within the conference bridge.


No need to manipulate from the dialplan anymore.


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

--
_
-- 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] Questions on converting to ConfBridge

[Richard Kenner]

I'm looking at what would be involved in converting from MeetMe to
ConfBridge and there seems to be a lot of missing administrative things,
but I hope I'm just missing it.  We all know about the missing realtime
linkage.  That's a major nuisance, but can be worked around.

More serious is that the CLI command to display users in a ConfBridge
don't show the caller ID information, so it becomes very hard to
have web applications that show who's in a conference.  There also doesn't
seem to be a way to lock conferences or mute or kick out users from
the dialplan.  And the CLI command needs a channel, not a user index,
making scripting via the dialplan that much harder.

What am I missing?

--
_
-- 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] Questions on converting to ConfBridge

[Joshua Colp]

Hola,

Richard Kenner wrote:


I'm looking at what would be involved in converting from MeetMe to
ConfBridge and there seems to be a lot of missing administrative things,
but I hope I'm just missing it.  We all know about the missing realtime
linkage.  That's a major nuisance, but can be worked around.


I think you are only the second or third person to really even mention 
realtime in the context of ConfBridge in the places I see.



More serious is that the CLI command to display users in a ConfBridge
don't show the caller ID information, so it becomes very hard to
have web applications that show who's in a conference.  There also doesn't
seem to be a way to lock conferences or mute or kick out users from
the dialplan.  And the CLI command needs a channel, not a user index,
making scripting via the dialplan that much harder.


Why are you wanting to use CLI commands instead of AMI? The available 
AMI actions for ConfBridge can do listing/locking/muting/kicking etc as 
you want.


As for dialplan applications to do the various things - what are you 
trying to achieve using them?


Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com   www.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

[asterisk-users] Questions about fax detection

[Olivier]

Hello,

I want to offer SIP phone user a custom fax-to-email feature.
Here is how I would describe this feature:

- for every SIP phone,a custom email address is defined
- when a SIP phone answers an incoming call (from a trunk or another SIP
endpoint), it detects the call is coming from a fax machine and then :
+ it plays a pre-recorded audio file to the receiving user (You are now
receiving a fax call, please check you email box)
+ while at the same time, the incoming channel is forwarded to an
appropriate statement within Asterisk dialplan.
- when an unanswered call is forwarded to a voicemail, the fax call is also
detected and teated appropriately.


1. It is possible to play a pre-recorded audio file to the receiving user ?
If positive, how can it be done ?

2. What is the exact purpose of sip.conf faxdetect setting in this case
given the assumption faxdetect is set to yes in general section of
sip.conf.
I would say the following applies:
faxdetect for the incoming channel has no influence at all.
If faxdetect is set to yes or unset in the outgoing channel, then Asterisk
will jump to fax extension.
If faxdetect is set to no in the outgoing channel, then Asterisk will jump
to fax extension.

Do you agree ?

3. Using CLI, is there a way to read the faxdetect parameter value of a
given SIP peer ?
To me sip show peer foo doesn't (seem to) display this.

4. When I type fax show settings in, I've got (on an Asterisk 10 box):
CLI fax show settings
FAX For Asterisk Settings:
ECM: Enabled
Status Events: Off
Minimum Bit Rate: 2400
Maximum Bit Rate: 14400
Modem Modulations Allowed: V17,V27,V29


FAX Technology Modules:

Spandsp (Spandsp FAX Driver) Settings:
CLI

This last line troubles me a little (did I forget to configure spandsp ?).
Should I care ?


Regards
--
_
-- 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] Questions on hardware or software-based echo cancellation

[Olivier]

2012/1/12, Kevin P. Fleming kpflem...@digium.com:
 On 01/12/2012 06:39 AM, Olivier wrote:
 Hi,

 I'm having some questions related to echo cancellation configuration
 on a Digium board enabled systems (B410P, TE420, TE420B, ) for
 cases when a hardware ech canceller is present or not.

 I read in TEXXX manual that when setting echocancel=yes in
 chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo
 cancellation was enabled.

 1. I'm correct thinking that it is then impossible to switch from
 hardware to software echo can without removing the VPMOCT64 module
 itself ?
 2. Does the same also apply to HA8 and its VPMOCT032 module ?

 With DAHDI 2.6 (and possibly 2.5), it is possible to override the
 configuration and apply a software echo canceller to a channel even if
 it has a hardware one. With prior versions, yes, the echo cancellation
 module would have to be physically removed (or disabled using a
 parameter to the kernel module).

Then, maybe a line mentioning that in the next User Manual edition
would be perfect.
Thanks for replying.


 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.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


--
_
-- 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] Questions on hardware or software-based echo cancellation

[Kevin P. Fleming]

On 01/13/2012 02:12 AM, Olivier wrote:

2012/1/12, Kevin P. Flemingkpflem...@digium.com:

On 01/12/2012 06:39 AM, Olivier wrote:

Hi,

I'm having some questions related to echo cancellation configuration
on a Digium board enabled systems (B410P, TE420, TE420B, ) for
cases when a hardware ech canceller is present or not.

I read in TEXXX manual that when setting echocancel=yes in
chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo
cancellation was enabled.

1. I'm correct thinking that it is then impossible to switch from
hardware to software echo can without removing the VPMOCT64 module
itself ?
2. Does the same also apply to HA8 and its VPMOCT032 module ?


With DAHDI 2.6 (and possibly 2.5), it is possible to override the
configuration and apply a software echo canceller to a channel even if
it has a hardware one. With prior versions, yes, the echo cancellation
module would have to be physically removed (or disabled using a
parameter to the kernel module).


Then, maybe a line mentioning that in the next User Manual edition
would be perfect.


Sure, but you have to understand that the user manuals for our board 
products are typically only updated when the board itself gets changed; 
we don't actually deliver DAHDI with the boards, and 
installation/configuration instructions for DAHDI are primarily included 
in the manual for user convenience. Users should be aware that they 
could easily be out of date, and not include all options that are 
currently offered (although if the user manual's instructions become 
incorrect, we'll update the manual). Regardless, I'll mention this to 
the people who manage those products. Thanks!


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.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

[asterisk-users] Questions on hardware or software-based echo cancellation

[Olivier]

Hi,

I'm having some questions related to echo cancellation configuration
on a Digium board enabled systems (B410P, TE420, TE420B, ) for
cases when a hardware ech canceller is present or not.

I read in TEXXX manual that when setting echocancel=yes in
chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo
cancellation was enabled.

1. I'm correct thinking that it is then impossible to switch from
hardware to software echo can without removing the VPMOCT64 module
itself ?
2. Does the same also apply to HA8 and its VPMOCT032 module ?
3. Are the only options for OSLEC configuration the  echocancel=128 or
echocancel=256 values in chan_dahdi.conf ?
4. How could be compared user experience with oslec/256, mg2/256,
mg2/1024 on a HA8 without hardware module ? Which would you recommend
?

Regards

--
_
-- 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] Questions on hardware or software-based echo cancellation

[Kevin P. Fleming]

On 01/12/2012 06:39 AM, Olivier wrote:

Hi,

I'm having some questions related to echo cancellation configuration
on a Digium board enabled systems (B410P, TE420, TE420B, ) for
cases when a hardware ech canceller is present or not.

I read in TEXXX manual that when setting echocancel=yes in
chan_dahdi.conf on a VPMOCT64-equiped system, 128ms hardware echo
cancellation was enabled.

1. I'm correct thinking that it is then impossible to switch from
hardware to software echo can without removing the VPMOCT64 module
itself ?
2. Does the same also apply to HA8 and its VPMOCT032 module ?


With DAHDI 2.6 (and possibly 2.5), it is possible to override the 
configuration and apply a software echo canceller to a channel even if 
it has a hardware one. With prior versions, yes, the echo cancellation 
module would have to be physically removed (or disabled using a 
parameter to the kernel module).


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.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

[asterisk-users] Questions on IAX client

[asterisk asterisk]

Hi,

I used to use Zoiper IAX to connect to my asterisk server from remote site.
On asterisk CLI, I can see my zoiper client registered and stay on line.
HOwever, I don't know why now I can't call this client. It always show up as
Unable to create channel IAX2 (Cause 20 Unknown)

I am using Asterisk 1.8.7.1

CK
--
_
-- 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] Questions on IAX client

[Steve Edwards]

On Sun, 23 Oct 2011, asterisk asterisk wrote:

I used to use Zoiper IAX to connect to my asterisk server from remote 
site. On asterisk CLI, I can see my zoiper client registered and stay on 
line. HOwever, I don't know why now I can't call this client. It always 
show up as Unable to create channel IAX2 (Cause 20 Unknown)


If you enable IAX debugging you may get some clues.

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

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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] Questions on Dahdi

[asterisk asterisk]

I have naive question. I do not have any hardware on my asterisk host. All I
have are either SIP trunk for DID or hardware ATA which bridges the asterisk
to PSTN. Do I need Dahdi install? Do i have ztdummy for timing issue? I
encounter problem in this when I try to install Dahdi latest but I found it
is not running, Instead it runs when service starts but I can't find its
status when I type in service dahdi status.

I am using Asterisk 1.8.7 on centos 5.7 32 bit.

CK
--
_
-- 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] Questions about FMFM with linked servers

[Faisal Hanif]

Did you tried to execute Set(CALLERID(num)=you-required-callerid)?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman
Sent: Friday, July 29, 2011 1:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Questions about FMFM with linked servers

 

All;

 

In a linked server environment, running Asterisk 1.6 I am noticing that when
a call is placed from server A to server B (via 4 digit extension) and
server B ext has a FMFM to call their mobile, the mobile phone shows the
default caller ID setting on server B instead of the actual caller id of the
person who initiated the call on server A.

 

This scenario, of course, works in the event a call in placed via the PSTN
into Server A (or B) and rings the FMFM extension. In this case, the mobile
phones sees the correct (initial) caller ID on the mobile.

 

Thanks!

 

--Dovey

--
_
-- 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] Questions about FMFM with linked servers

[Dovey Forman]

All;

In a linked server environment, running Asterisk 1.6 I am noticing that when
a call is placed from server A to server B (via 4 digit extension) and
server B ext has a FMFM to call their mobile, the mobile phone shows the
default caller ID setting on server B instead of the actual caller id of the
person who initiated the call on server A.

This scenario, of course, works in the event a call in placed via the PSTN
into Server A (or B) and rings the FMFM extension. In this case, the mobile
phones sees the correct (initial) caller ID on the mobile.

Thanks!

--Dovey
--
_
-- 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] Questions About Fax for Asterisk

[Steve Totaro]

On Fri, May 7, 2010 at 9:56 PM, Steve Underwood ste...@coppice.org wrote:

 On 05/08/2010 08:15 AM, Steve Totaro wrote:
 
 
  On Fri, May 7, 2010 at 2:01 PM, Martin asteriskl...@callthem.info
  mailto:asteriskl...@callthem.info wrote:
 
  On Thu, May 6, 2010 at 3:11 PM, Steve Totaro
  stot...@totarotechnologies.com
  mailto:stot...@totarotechnologies.com wrote:
   Yes, I purchased licenses for Fax for Asterisk and yes I called
  tech support
   and had the WORST experience I have ever had with any technical
  support
   call.
  
   I am running Asterisk 1.6.2.6 and:
  
   FAX For Asterisk Components:
   Applications: 1.6.2.0_1.2.0
   voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (optimized for c3_2_32)
  
   The guy was arrogant and absolutely a jerk and I don't like to
  call people
   names, but call it as I see it.  This has not been my experience
  the five or
   six times I have had to call Digium over the years, but it has
  been many
   years since my last call so I have no idea what the general
  support staff is
   like.
  
   I could not get any questions answered by the tech that took
  hours to call
   me back to tell me to read the readme.  That would be all well
  and good if I
   didn't pay money.
  
   He could not explain Digium's math as far as faxing and failed
  to offer to
   get back to me with any kind of answer.
  
   Maybe someone on the list can make sense of this Enron style of
  accounting:
  
   voipgw01*CLI fax show stats
   voipgw01*CLI
   FAX Statistics:
   ---
  
   Current Sessions : 1
   Transmit Attempts: 0
   Receive Attempts : 336
   Completed FAXes  : 320
   Failed FAXes : 57
  
   Digium G.711
   Licensed Channels: 4
   Max Concurrent   : 1
   Success  : 0
   Switched to T.38 : 0
   Canceled : 0
   No FAX   : 1
   Partial  : 0
   Negotiation Failed   : 0
   Train Failure: 3
   Protocol Error   : 0
   IO Partial   : 0
   IO Fail  : 0
   voipgw01*CLI
   Digium T.38
   Licensed Channels: 4
   Max Concurrent   : 4
   Success  : 175
   Canceled : 0
   No FAX   : 6
   Partial  : 19
   Negotiation Failed   : 0
   Train Failure: 83
   Protocol Error   : 33
   IO Partial   : 0
   IO Fail  : 0
  
   Thanks,
   Steve Totaro
 
 
  wow definitely the acccounting engine is broken ...
 
  I can only make sense of this
 
   Receive Attempts : 336
   Completed FAXes  : 320
   Failed FAXes : 57
 
  1) your receive app was called 336 times but the fax hanged up before
  negotiating
  2) you had 320 of this completed (partially or fully)
  3) but 57 out of 320 failed to transmit entirely
 
  57/320=17.8% which is too high for a commercial product IHMO
 
  Martin
 
 
  Considering that this is a direct cross connect from Leve3's cage to
  my my cage in the same DC at an Equinix facility, 100Mb DIA w/EIPT
  VoIP service, I would expect nearly 100% success.
 
  Considering the circuit was just turned up and there is no data except
  Level3's phone traffic.  They are our carrier, RespOrg, origination
  and termination, no 3rd parties, all on net.
 
  I could understand if it was a peaked out DIA circuit to some cut rate
  VoIP provider, but not under perfect circumstances.
 
  Thanks,
  Steve Totaro
 Were these all test calls made from a well defined source? It takes
 *two* correctly working FAX terminals to make a successful call. Its
 easy to get a high failure rate for silly reasons. In volume testing of
 spandsp and iaxmodem we had times where a high percentage of calls
 failed, which turned out to be just one rouge machine calling over and
 over again trying to achieve success. On the other hand, failures
 between known good FAX terminals should be far below 1%.

 Steve


These are not test calls.  These are real world calls from a real world.

Since this is Fax for Asterisk, volume is not really an issue, since I only
have four licenses on a brand new CentOS box (HP DL360 G6(or whatever is
currently shipping from HP).

Based on caller ID, it is not one caller inflating the numbers.  Generally,
a failed fax will succeed on a subsequent try.

Previously, we were terminating faxes to a quad port Digium PRI card,
everything from the OS to the hardware were from 2006.  Although, I do not
have exact numbers, they were much better from this Unsupported Digium
setup over this Supported and sold for profit solution.

Maybe there is a simple setting somewhere, but RTFM from Digium tech
support when the FM offers 

Re: [asterisk-users] Questions About Fax for Asterisk

[David Backeberg]

On Sat, May 8, 2010 at 7:21 AM, Steve Totaro
stot...@first-notification.com wrote:
 Maybe there is a simple setting somewhere, but RTFM from Digium tech
 support when the FM offers no suggestion on how to possibly tweak settings
 for better success.

Do you want to dump some samples from your dialplan?

I know I personally had better success with fax over sip when I put a:
Playback(silence/1)
infront of
ReceiveFax()

-- 
_
-- 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] Questions About Fax for Asterisk

[Steve Totaro]

On Sat, May 8, 2010 at 2:27 PM, David Backeberg dbackeb...@gmail.comwrote:

 On Sat, May 8, 2010 at 7:21 AM, Steve Totaro
 stot...@first-notification.com wrote:
  Maybe there is a simple setting somewhere, but RTFM from Digium tech
  support when the FM offers no suggestion on how to possibly tweak
 settings
  for better success.

 Do you want to dump some samples from your dialplan?

 I know I personally had better success with fax over sip when I put a:
 Playback(silence/1)
 infront of
 ReceiveFax()


I will post Monday, I have had exten=whatever,1,Answer(3) since day one, I
would guess that does just about the same thing as Playback(silence/1).

Thanks,
Steve Totaro
-- 
_
-- 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] Questions About Fax for Asterisk

[Martin]

On Thu, May 6, 2010 at 3:11 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
 Yes, I purchased licenses for Fax for Asterisk and yes I called tech support
 and had the WORST experience I have ever had with any technical support
 call.

 I am running Asterisk 1.6.2.6 and:

 FAX For Asterisk Components:
     Applications: 1.6.2.0_1.2.0
 voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (optimized for c3_2_32)

 The guy was arrogant and absolutely a jerk and I don't like to call people
 names, but call it as I see it.  This has not been my experience the five or
 six times I have had to call Digium over the years, but it has been many
 years since my last call so I have no idea what the general support staff is
 like.

 I could not get any questions answered by the tech that took hours to call
 me back to tell me to read the readme.  That would be all well and good if I
 didn't pay money.

 He could not explain Digium's math as far as faxing and failed to offer to
 get back to me with any kind of answer.

 Maybe someone on the list can make sense of this Enron style of accounting:

 voipgw01*CLI fax show stats
 voipgw01*CLI
 FAX Statistics:
 ---

 Current Sessions : 1
 Transmit Attempts    : 0
 Receive Attempts : 336
 Completed FAXes  : 320
 Failed FAXes : 57

 Digium G.711
 Licensed Channels    : 4
 Max Concurrent   : 1
 Success  : 0
 Switched to T.38 : 0
 Canceled : 0
 No FAX   : 1
 Partial  : 0
 Negotiation Failed   : 0
 Train Failure    : 3
 Protocol Error   : 0
 IO Partial   : 0
 IO Fail  : 0
 voipgw01*CLI
 Digium T.38
 Licensed Channels    : 4
 Max Concurrent   : 4
 Success  : 175
 Canceled : 0
 No FAX   : 6
 Partial  : 19
 Negotiation Failed   : 0
 Train Failure    : 83
 Protocol Error   : 33
 IO Partial   : 0
 IO Fail  : 0

 Thanks,
 Steve Totaro


wow definitely the acccounting engine is broken ...

I can only make sense of this

 Receive Attempts : 336
 Completed FAXes  : 320
 Failed FAXes : 57

1) your receive app was called 336 times but the fax hanged up before
negotiating
2) you had 320 of this completed (partially or fully)
3) but 57 out of 320 failed to transmit entirely

57/320=17.8% which is too high for a commercial product IHMO

Martin




 --
 _
 -- 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] Questions About Fax for Asterisk

[Steve Totaro]

On Fri, May 7, 2010 at 2:01 PM, Martin asteriskl...@callthem.info wrote:

 On Thu, May 6, 2010 at 3:11 PM, Steve Totaro
 stot...@totarotechnologies.com wrote:
  Yes, I purchased licenses for Fax for Asterisk and yes I called tech
 support
  and had the WORST experience I have ever had with any technical support
  call.
 
  I am running Asterisk 1.6.2.6 and:
 
  FAX For Asterisk Components:
  Applications: 1.6.2.0_1.2.0
  voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (optimized for c3_2_32)
 
  The guy was arrogant and absolutely a jerk and I don't like to call
 people
  names, but call it as I see it.  This has not been my experience the five
 or
  six times I have had to call Digium over the years, but it has been many
  years since my last call so I have no idea what the general support staff
 is
  like.
 
  I could not get any questions answered by the tech that took hours to
 call
  me back to tell me to read the readme.  That would be all well and good
 if I
  didn't pay money.
 
  He could not explain Digium's math as far as faxing and failed to offer
 to
  get back to me with any kind of answer.
 
  Maybe someone on the list can make sense of this Enron style of
 accounting:
 
  voipgw01*CLI fax show stats
  voipgw01*CLI
  FAX Statistics:
  ---
 
  Current Sessions : 1
  Transmit Attempts: 0
  Receive Attempts : 336
  Completed FAXes  : 320
  Failed FAXes : 57
 
  Digium G.711
  Licensed Channels: 4
  Max Concurrent   : 1
  Success  : 0
  Switched to T.38 : 0
  Canceled : 0
  No FAX   : 1
  Partial  : 0
  Negotiation Failed   : 0
  Train Failure: 3
  Protocol Error   : 0
  IO Partial   : 0
  IO Fail  : 0
  voipgw01*CLI
  Digium T.38
  Licensed Channels: 4
  Max Concurrent   : 4
  Success  : 175
  Canceled : 0
  No FAX   : 6
  Partial  : 19
  Negotiation Failed   : 0
  Train Failure: 83
  Protocol Error   : 33
  IO Partial   : 0
  IO Fail  : 0
 
  Thanks,
  Steve Totaro


 wow definitely the acccounting engine is broken ...

 I can only make sense of this

  Receive Attempts : 336
  Completed FAXes  : 320
  Failed FAXes : 57

 1) your receive app was called 336 times but the fax hanged up before
 negotiating
 2) you had 320 of this completed (partially or fully)
 3) but 57 out of 320 failed to transmit entirely

 57/320=17.8% which is too high for a commercial product IHMO

 Martin


Considering that this is a direct cross connect from Leve3's cage to my my
cage in the same DC at an Equinix facility, 100Mb DIA w/EIPT VoIP service, I
would expect nearly 100% success.

Considering the circuit was just turned up and there is no data except
Level3's phone traffic.  They are our carrier, RespOrg, origination and
termination, no 3rd parties, all on net.

I could understand if it was a peaked out DIA circuit to some cut rate VoIP
provider, but not under perfect circumstances.

Thanks,
Steve Totaro
-- 
_
-- 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] Questions About Fax for Asterisk

[Steve Underwood]

On 05/08/2010 08:15 AM, Steve Totaro wrote:


 On Fri, May 7, 2010 at 2:01 PM, Martin asteriskl...@callthem.info 
 mailto:asteriskl...@callthem.info wrote:

 On Thu, May 6, 2010 at 3:11 PM, Steve Totaro
 stot...@totarotechnologies.com
 mailto:stot...@totarotechnologies.com wrote:
  Yes, I purchased licenses for Fax for Asterisk and yes I called
 tech support
  and had the WORST experience I have ever had with any technical
 support
  call.
 
  I am running Asterisk 1.6.2.6 and:
 
  FAX For Asterisk Components:
  Applications: 1.6.2.0_1.2.0
  voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (optimized for c3_2_32)
 
  The guy was arrogant and absolutely a jerk and I don't like to
 call people
  names, but call it as I see it.  This has not been my experience
 the five or
  six times I have had to call Digium over the years, but it has
 been many
  years since my last call so I have no idea what the general
 support staff is
  like.
 
  I could not get any questions answered by the tech that took
 hours to call
  me back to tell me to read the readme.  That would be all well
 and good if I
  didn't pay money.
 
  He could not explain Digium's math as far as faxing and failed
 to offer to
  get back to me with any kind of answer.
 
  Maybe someone on the list can make sense of this Enron style of
 accounting:
 
  voipgw01*CLI fax show stats
  voipgw01*CLI
  FAX Statistics:
  ---
 
  Current Sessions : 1
  Transmit Attempts: 0
  Receive Attempts : 336
  Completed FAXes  : 320
  Failed FAXes : 57
 
  Digium G.711
  Licensed Channels: 4
  Max Concurrent   : 1
  Success  : 0
  Switched to T.38 : 0
  Canceled : 0
  No FAX   : 1
  Partial  : 0
  Negotiation Failed   : 0
  Train Failure: 3
  Protocol Error   : 0
  IO Partial   : 0
  IO Fail  : 0
  voipgw01*CLI
  Digium T.38
  Licensed Channels: 4
  Max Concurrent   : 4
  Success  : 175
  Canceled : 0
  No FAX   : 6
  Partial  : 19
  Negotiation Failed   : 0
  Train Failure: 83
  Protocol Error   : 33
  IO Partial   : 0
  IO Fail  : 0
 
  Thanks,
  Steve Totaro


 wow definitely the acccounting engine is broken ...

 I can only make sense of this

  Receive Attempts : 336
  Completed FAXes  : 320
  Failed FAXes : 57

 1) your receive app was called 336 times but the fax hanged up before
 negotiating
 2) you had 320 of this completed (partially or fully)
 3) but 57 out of 320 failed to transmit entirely

 57/320=17.8% which is too high for a commercial product IHMO

 Martin


 Considering that this is a direct cross connect from Leve3's cage to 
 my my cage in the same DC at an Equinix facility, 100Mb DIA w/EIPT 
 VoIP service, I would expect nearly 100% success.

 Considering the circuit was just turned up and there is no data except 
 Level3's phone traffic.  They are our carrier, RespOrg, origination 
 and termination, no 3rd parties, all on net.

 I could understand if it was a peaked out DIA circuit to some cut rate 
 VoIP provider, but not under perfect circumstances.

 Thanks,
 Steve Totaro
Were these all test calls made from a well defined source? It takes 
*two* correctly working FAX terminals to make a successful call. Its 
easy to get a high failure rate for silly reasons. In volume testing of 
spandsp and iaxmodem we had times where a high percentage of calls 
failed, which turned out to be just one rouge machine calling over and 
over again trying to achieve success. On the other hand, failures 
between known good FAX terminals should be far below 1%.

Steve


-- 
_
-- 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] Questions About Fax for Asterisk

[Steve Totaro]

Yes, I purchased licenses for Fax for Asterisk and yes I called tech support
and had the WORST experience I have ever had with any technical support
call.

I am running Asterisk 1.6.2.6 and:

FAX For Asterisk Components:
Applications: 1.6.2.0_1.2.0
voipgw01Digium FAX Driver: 1.6.2.0_1.2.0 (optimized for c3_2_32)

The guy was arrogant and absolutely a jerk and I don't like to call people
names, but call it as I see it.  This has not been my experience the five or
six times I have had to call Digium over the years, but it has been many
years since my last call so I have no idea what the general support staff is
like.

I could not get any questions answered by the tech that took hours to call
me back to tell me to read the readme.  That would be all well and good if I
didn't pay money.

He could not explain Digium's math as far as faxing and failed to offer to
get back to me with any kind of answer.

Maybe someone on the list can make sense of this Enron style of accounting:

voipgw01*CLI fax show stats
voipgw01*CLI
FAX Statistics:
---

Current Sessions : 1
Transmit Attempts: 0
Receive Attempts : 336
Completed FAXes  : 320
Failed FAXes : 57

Digium G.711
Licensed Channels: 4
Max Concurrent   : 1
Success  : 0
Switched to T.38 : 0
Canceled : 0
No FAX   : 1
Partial  : 0
Negotiation Failed   : 0
Train Failure: 3
Protocol Error   : 0
IO Partial   : 0
IO Fail  : 0
voipgw01*CLI
Digium T.38
Licensed Channels: 4
Max Concurrent   : 4
Success  : 175
Canceled : 0
No FAX   : 6
Partial  : 19
Negotiation Failed   : 0
Train Failure: 83
Protocol Error   : 33
IO Partial   : 0
IO Fail  : 0

Thanks,
Steve Totaro
-- 
_
-- 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] Questions about asterisk and spa2102

[Kosa]

Hi there! First mail on the list :)

1.- is it possible to use an spa2102 to make and revice calls from a
normal phone? I mean, I know I can use it to connect an analog to an
asterisk server, but I want to know if it can be used to connect
asterisk to the analog phoneline.

2.-  I'm trying to unlock the spa2102 with no succes at the moment, any
links or hint will be very appreciated.


I'm and absolute newbie on asterisk, btw.


Thanx!

Kosa

- Un mundo mejor es posible -


-- 
_
-- 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] Questions about asterisk and spa2102

[John Novack]

Kosa wrote:
 Hi there! First mail on the list :)

 1.- is it possible to use an spa2102 to make and revice calls from a normal 
 phone? I mean, I know I can use it to connect an analog to an asterisk 
 server, but I want to know if it can be used to connect asterisk to the 
 analog phoneline.
   
In simple terms:

FXS ports provide battery to analog phones, provide ringing to analog 
phones when so instructed, and provide dialtone to analog phones, then 
forward the dialed number as data to a server.
FXO  ports expect to see battery from analog exchange lines, supply a 
loop closure to request service from an exchange, in some cases will 
pulse dial a string of digits, in all cases send a string of DTMF 
digits, and detect a ringing voltage from an exchange, forwarding 
received information as data to a server.

Some devices have both types of connections.

If you want an external device to do both, it will need both types of 
ports, one cannot be both.
 2.-  I'm trying to unlock the spa2102 with no succes at the moment, any
 links or hint will be very appreciated.

   
Why waste your ( valuable? ) time??
New unlocked similar devices are available from multiple sources.
Many of these are born locked to a specific service, and cannot be changed.


 I'm and absolute newbie on asterisk, btw.


 Thanx!

 Kosa

 - Un mundo mejor es posible -


   

-- 
Dog is my co-pilot


-- 
_
-- 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] Questions about asterisk and spa2102

[Steve Edwards]

On Fri, 29 Jan 2010, Kosa wrote:

 1.- is it possible to use an spa2102 to make and revice calls from a 
 normal phone? I mean, I know I can use it to connect an analog to an 
 asterisk server, but I want to know if it can be used to connect 
 asterisk to the analog phoneline.

The 2102 is an FXS (station) device. It connects to things like a 
telephone or a fax machine.

The 3102 is an FXS and FXO (office) device. You can plug in a telephone 
and the wire coming out of the wall. I have the predecessor, the 3000. It 
has the neat feature that if it loses power it will bridge the FXS and 
FXO ports so the telephone can still be used. I don't know if current 
models still have this feature.

 2.- I'm trying to unlock the spa2102 with no succes at the moment, any 
 links or hint will be very appreciated.

I did this many years ago with some PAP2s that were locked to Vonage. 
Definitely not worth the effort it took to configure my name servers to 
pretend they were Vonage's so they could resolve Vonage's names to my 
local IP addresses and setting up the Ethernet interface aliases to 
Vonage's IP addresses.

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

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

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

Re: [asterisk-users] Questions about static

[Dovey Forman]

We have swapped out the phone multiple times for the user.
Only one user.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cb
Sent: Wednesday, November 25, 2009 11:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about static

On Nov 25, 2009, at 3:07 PM, Dovey Forman wrote:

 Would be a cause of static for inbound/outbound and ext to ext calls?

 Its voip both in and out.

 We swapped, phones, cordes, switches etc...

 Typically a reboot of the phone resolves the problem.person also
 swears there is nothing on or near their desk to cause interference
 (microwave, cell phone is purse).

Only one user? Did you check to see if it is a bad handset cord?

-chris
www.mythtech.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

___
-- 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] Questions about static

[Dovey Forman]

It’s a single user and we have swapped everything.

The phone is an Aastra 6731i and its PoE.



*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Michael Wyres
*Sent:* Wednesday, November 25, 2009 6:27 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Questions about static



Is it a single user?  Or every single phone?



If it’s a single user, and you can get hold of a UPS with power conditioning
on it, try plugging the various devices into it – there might be some dirty
power coming along.





*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dovey Forman
*Sent:* Thursday, 26 November 2009 07:08
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Questions about static



Using an Asterisk system running 1.2 with Aastra phones.

Would be a cause of static for inbound/outbound and ext to ext calls?



Its voip both in and out.



We swapped, phones, cordes, switches etc…..



Typically a reboot of the phone resolves the problem…person also swears
there is nothing on or near their desk to cause interference (microwave,
cell phone is purse).



Strange……



Thanks

--Dovey

IMPORTANT NOTICE TO RECIPIENT



Computer viruses - It is your responsibility to scan this email and
any attachments for viruses and defects and rely on those scans as
Communications Design  Management Pty Limited (CDM) does not accept
any liability for loss or damage arising from receipt or use of this
email or any attachments.



Confidentiality - This email and any attachments are intended for the
named recipient only and may contain personal information, be it
confidential or subject to privilege, none of which are lost or waived
because this email may have been sent to you in error. If you are not
the named addressee please let CDM know by return email, permanently
delete it from your system and destroy all copies and do not use or
disclose the contents.



Copyright - This email is subject to copyright and no part of it maybe
reproduced in any manner without the written permission of the
copyright owner.



Privacy - Within the jurisdiction of Australian law, personal
information in this email must be dealt with in compliance with the
Australian Federal Privacy Act 1988.
___
-- 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] Questions about static

[Noah Miller]

 We have swapped out the phone multiple times for the user.
 Only one user.

Bad PoE port on the switch?

How about local interference that the user cannot control?  Does the
same phone experience static when moved elsewhere?

Do you have a power brick for the phone so you can try it as non-PoE?

Is the static consistent or intermittent?


- Noah

___
-- 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] Questions about static

[Dovey Forman]

We swapped PoE switches, phones, cable and switch ports multiple times.
What do you mean by local interference? Cell phone? The person swears
nothing is near the phone.

Its very strange.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah Miller
Sent: Friday, November 27, 2009 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about static

 We have swapped out the phone multiple times for the user.
 Only one user.

Bad PoE port on the switch?

How about local interference that the user cannot control?  Does the
same phone experience static when moved elsewhere?

Do you have a power brick for the phone so you can try it as non-PoE?

Is the static consistent or intermittent?


- Noah

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

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

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

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

Re: [asterisk-users] Questions about static

[Don Kelly]

Could the static be in the user's hearing aid?

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman
Sent: Friday, November 27, 2009 12:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about static

We swapped PoE switches, phones, cable and switch ports multiple times.
What do you mean by local interference? Cell phone? The person swears
nothing is near the phone.

Its very strange.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah Miller
Sent: Friday, November 27, 2009 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about static

 We have swapped out the phone multiple times for the user.
 Only one user.

Bad PoE port on the switch?

How about local interference that the user cannot control?  Does the
same phone experience static when moved elsewhere?

Do you have a power brick for the phone so you can try it as non-PoE?

Is the static consistent or intermittent?


- Noah

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

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

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

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


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

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

Re: [asterisk-users] Questions about static

[Noah Miller]

 We swapped PoE switches, phones, cable and switch ports multiple times.
 What do you mean by local interference? Cell phone? The person swears
 nothing is near the phone.

There are lots of things that can cause interference.  Radios,
elevators, bad electrical wiring, you name it.  Is the static still
there when you move the identical phone elsewhere?  If not, then the
static is most probably caused by some local interference where the
user is.


- Noah

___
-- 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] Questions about Voicemail

[Robert Lister]

On Mon, 2009-11-23 at 14:46 -0500, Dovey Forman wrote:
 Regarding the email to multiple receipients, it is available on an ad-hoc
 basis from the phone?
 
 IE; call into the voicemail system, enter x digit to send a voicemail to
 multiple users, record the message, then enter the destination mailboxes,
 separated by  # sign...

You can enable an option in the voicemail that allows the prompt:

'To send a message to another user'...

sendvoicemail=yes ; Allow the user to compose and send a voicemail
  ; while  inside VoiceMailMain() [option 5 from
  ; mailbox's advanced menu]. If set to 'no', 
  ; option 5 will not be listed.

This would enable the option from within the vm app, but you want to do
a dynamic list of mailboxes to deliver to, so by the time we get here,
I think it's going to be to late to to anything useful (since we already
called the voicemail app.)

You could write some dialplan magic with a while loop, so that the user
can dial a specific extn (maybe call it 'group message') and then it
will prompt for a mailbox number, followed by #, or just # to end.

Then it could build this list of mailboxes as a variable before calling
the voicemail app.

I can attempt to build an example if you are interested.

Rob



___
-- 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] Questions about Voicemail

[Dovey Forman]

Rob;

That would be great. You could send directly to me @ dovey.for...@idt.net
or respond to this list.

I appreciate it!

--Dovey

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Lister
Sent: Wednesday, November 25, 2009 11:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about Voicemail

On Mon, 2009-11-23 at 14:46 -0500, Dovey Forman wrote:
 Regarding the email to multiple receipients, it is available on an
ad-hoc
 basis from the phone?

 IE; call into the voicemail system, enter x digit to send a voicemail to
 multiple users, record the message, then enter the destination
mailboxes,
 separated by  # sign...

You can enable an option in the voicemail that allows the prompt:

'To send a message to another user'...

sendvoicemail=yes ; Allow the user to compose and send a voicemail
  ; while  inside VoiceMailMain() [option 5 from
  ; mailbox's advanced menu]. If set to 'no',
  ; option 5 will not be listed.

This would enable the option from within the vm app, but you want to do
a dynamic list of mailboxes to deliver to, so by the time we get here,
I think it's going to be to late to to anything useful (since we already
called the voicemail app.)

You could write some dialplan magic with a while loop, so that the user
can dial a specific extn (maybe call it 'group message') and then it
will prompt for a mailbox number, followed by #, or just # to end.

Then it could build this list of mailboxes as a variable before calling
the voicemail app.

I can attempt to build an example if you are interested.

Rob



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

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

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

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

[asterisk-users] Questions about static

[Dovey Forman]

Using an Asterisk system running 1.2 with Aastra phones.

Would be a cause of static for inbound/outbound and ext to ext calls?



Its voip both in and out.



We swapped, phones, cordes, switches etc…..



Typically a reboot of the phone resolves the problem…person also swears
there is nothing on or near their desk to cause interference (microwave,
cell phone is purse).



Strange……



Thanks

--Dovey
___
-- 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] Questions about static

[Michael Wyres]

Is it a single user?  Or every single phone?

If it's a single user, and you can get hold of a UPS with power conditioning on 
it, try plugging the various devices into it - there might be some dirty power 
coming along.


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman
Sent: Thursday, 26 November 2009 07:08
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Questions about static

Using an Asterisk system running 1.2 with Aastra phones.
Would be a cause of static for inbound/outbound and ext to ext calls?

Its voip both in and out.

We swapped, phones, cordes, switches etc.

Typically a reboot of the phone resolves the problem...person also swears there 
is nothing on or near their desk to cause interference (microwave, cell phone 
is purse).

Strange..

Thanks
--Dovey
IMPORTANT NOTICE TO RECIPIENT

Computer viruses - It is your responsibility to scan this email and any 
attachments for viruses and defects and rely on those scans as Communications 
Design  Management Pty Limited (CDM) does not accept any liability for loss or 
damage arising from receipt or use of this email or any attachments.

Confidentiality - This email and any attachments are intended for the named 
recipient only and may contain personal information, be it confidential or 
subject to privilege, none of which are lost or waived because this email may 
have been sent to you in error. If you are not the named addressee please let 
CDM know by return email, permanently delete it from your system and destroy 
all copies and do not use or disclose the contents.

Copyright - This email is subject to copyright and no part of it maybe 
reproduced in any manner without the written permission of the copyright owner.

Privacy - Within the jurisdiction of Australian law, personal information in 
this email must be dealt with in compliance with the Australian Federal Privacy 
Act 1988.
___
-- 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] Questions about static

[cb]

On Nov 25, 2009, at 3:07 PM, Dovey Forman wrote:

 Would be a cause of static for inbound/outbound and ext to ext calls?

 Its voip both in and out.

 We swapped, phones, cordes, switches etc…..

 Typically a reboot of the phone resolves the problem…person also  
 swears there is nothing on or near their desk to cause interference  
 (microwave, cell phone is purse).

Only one user? Did you check to see if it is a bad handset cord?

-chris
www.mythtech.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

[asterisk-users] Questions about Voicemail

[Dovey Forman]

I am sorry if this is not the correct place to post a question.



I am wondering if there is way in asterisk 1.2 to:



1.   Send a voicemail (from the phone) to multiple recipients.

2.   Require (as an admin) for users 1st logging into their voicemail to
change their greeting and/or password.



Thanks

--Dovey
___
-- 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] Questions about Voicemail

[Robert Lister]

On Mon, 2009-11-23 at 10:37 -0500, Dovey Forman wrote:
 I am sorry if this is not the correct place to post a question.

 I am wondering if there is way in asterisk 1.2 to:

 1.  Send a voicemail (from the phone) to multiple recipients.

Yes I believe so.

1. The voicemail app allows delivery to multiple destinations at once:

 - example :

exten = 100,1,VoiceMail(u101102103)

2. Create an e-mail alias/list and deliver the voicemail via e-mail to
that alias.


 2.  Require (as an admin) for users 1st logging into their
 voicemail to change their greeting and/or password.

There is a user option forcegreetings:

forcegreetings = [yes|no]

Sets whether the user will be forced to record a new greeting
when logging in to the system for the first time. Default: no

Example:

forcegreetings = no 


Not sure about the forced change PIN, but it should be easy enough to
write a little command wrapper around it and prompt for PIN via the
dialplan.


Rob





___
-- 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] Questions about Voicemail

[Dovey Forman]

Regarding the email to multiple receipients, it is available on an ad-hoc
basis from the phone?

IE; call into the voicemail system, enter x digit to send a voicemail to
multiple users, record the message, then enter the destination mailboxes,
separated by  # sign...

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Lister
Sent: Monday, November 23, 2009 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Questions about Voicemail

On Mon, 2009-11-23 at 10:37 -0500, Dovey Forman wrote:
 I am sorry if this is not the correct place to post a question.

 I am wondering if there is way in asterisk 1.2 to:

 1.  Send a voicemail (from the phone) to multiple recipients.

Yes I believe so.

1. The voicemail app allows delivery to multiple destinations at once:

 - example :

exten = 100,1,VoiceMail(u101102103)

2. Create an e-mail alias/list and deliver the voicemail via e-mail to
that alias.


 2.  Require (as an admin) for users 1st logging into their
 voicemail to change their greeting and/or password.

There is a user option forcegreetings:

forcegreetings = [yes|no]

Sets whether the user will be forced to record a new greeting
when logging in to the system for the first time. Default: no

Example:

forcegreetings = no


Not sure about the forced change PIN, but it should be easy enough to
write a little command wrapper around it and prompt for PIN via the
dialplan.


Rob





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

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

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

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

Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

[Olivier]

2009/11/11 Tzafrir Cohen tzafrir.co...@xorcom.com

 On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote:
  2009/11/10 Tzafrir Cohen tzafrir.co...@xorcom.com
 
   On Tue, Nov 10, 2009 at 06:06:12PM +0100, Olivier wrote:
Hello,
   
1. How can specify in /etc/dahdi/genconf_parameters file that a port
 from
   a
B410P board is to be disabled.
  
   There's currently no way to do that.
  
   It should be trivial to implment. The more difficult part of it would
 be
   how to define exactly what spans / channels to disable.
  
   But why do you need that?
  
 
  I don't really know why I thought I needed that feature but as some
 gateways
  implement this feature (the ability to enable or disable each port), I
 must
  have told myself I may have missed here.
 
  On a general point of view, as most Dahdi cards have a light showing
 nearby
  port status, it should ideally possible, to turn this light off when a
 port
  is disabled.
 
  But I must also add it doesn't seem very important to me to have this
  implemented.

 dahdi_genconf is an optional tool. Ideally it should need no
 configuration at all and generate configuration that Just Works (though
 the fact that it can do that indicates that the current defaults are
 broken).

 It should not be another configuration layer. If the configuration it
 has generated is not good enough, you can also manually edit it.

 
 
Playing with comments (see bellow) doesn't help : file
/etc/asterisk/dahdi-channels.conf is filled with 4 ports data.
   
pri_termtype
SPAN/1  TE
SPAN/2  TE
SPAN/3  TE
#   SPAN/4  TE
  
   Currently pri_termtype is the only directive in dahdi_genconf that uses
   this list syntax. I'm not very happy with it.
  
   I'm not exactly sure if there should be some sort of generic way of
   adding per-span (span? channel? how do you define a span?) definitions.
   Think of ssh_config.
  
 
 
  What about adding per-span section headers like Asterisk .conf files ?
  [span1]
  group_lines 1
  pri_termtype
  SPAN/1  TE
  SPAN/2  TE
 
  [span2]
  group_lines 2
  pri_termtype
  SPAN/2  TE

 This implies you will know span numbers in advance. I would like better
 ways to specify configuration.


Really ?
I used this [span1] header as an example. Using any other string would be
fine for me as what matters, if I'm not mistaken, is the group_lines number
:

[foo]
group_lines 1
pri_termtype
 SPAN/1  TE
 SPAN/2  TE

[bar]
group_lines 2



 
 
  I don't think we need to define any further what a span is, beside that
  rules that applied to the whole genconf_parameters (no more than 1
  group_lines statement) should apply to each section.

 --
Tzafrir Cohen
 icq#16849755  
 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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

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

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

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

Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

[Tzafrir Cohen]

On Thu, Nov 12, 2009 at 01:43:48PM +0100, Olivier wrote:
 2009/11/11 Tzafrir Cohen tzafrir.co...@xorcom.com
 
  On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote:

   What about adding per-span section headers like Asterisk .conf files ?
   [span1]
   group_lines 1
   pri_termtype
   SPAN/1  TE
   SPAN/2  TE
  
   [span2]
   group_lines 2
   pri_termtype
   SPAN/2  TE
 
  This implies you will know span numbers in advance. I would like better
  ways to specify configuration.
 
 
 Really ?
 I used this [span1] header as an example. Using any other string would be
 fine for me as what matters, if I'm not mistaken, is the group_lines number
 :
 
 [foo]
 group_lines 1
 pri_termtype
  SPAN/1  TE
  SPAN/2  TE
 
 [bar]
 group_lines 2

How can you tell which spans / channels will use each section?

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

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

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

Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

[Olivier]

2009/11/12 Tzafrir Cohen tzafrir.co...@xorcom.com

 On Thu, Nov 12, 2009 at 01:43:48PM +0100, Olivier wrote:
  2009/11/11 Tzafrir Cohen tzafrir.co...@xorcom.com
 
   On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote:

What about adding per-span section headers like Asterisk .conf files
 ?
[span1]
group_lines 1
pri_termtype
SPAN/1  TE
SPAN/2  TE
   
[span2]
group_lines 2
pri_termtype
SPAN/2  TE
  
   This implies you will know span numbers in advance. I would like better
   ways to specify configuration.
  
 
  Really ?
  I used this [span1] header as an example. Using any other string would be
  fine for me as what matters, if I'm not mistaken, is the group_lines
 number
  :
 
  [foo]
  group_lines 1
  pri_termtype
   SPAN/1  TE
   SPAN/2  TE
 
  [bar]
  group_lines 2

 How can you tell which spans / channels will use each section?


My understanding of Dahdi is that I mostly need a group number to use with
Dial application :
Dial(DAHDI/g1/0123456789).





To get that dahdi-channels.conf file generated with dahdi_genconf, the only
missing feature (if my understanding is correct) is to be able to group
together a couple of ports so that I could either include in my diaplans,
lines such as Dial(DAHDI/g1/0123456789) or Dial(DAHDI/g2/9876543210).

So with a /etc/dahdi/genconf_parameters like this ...

[foo]
group_lines 1
pri_termtype
  SPAN/1TE


[bar]
group_lines 2
pri_termtype
  SPAN/2TE


... I think we've got everything needed to generate a
/etc/asterisk/dahdi-channels.conf file this :

; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS
group=1,11
context=remote
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel = 1-2
context = default
group = 63

; Span 2: B4/0/2 B4XXP (PCI) Card 0 Span 2 AMI/CCS
group=2,12
context=remote
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel = 4-5
context = default
group = 63


So I don't understand where I would have to tell which spans / channels
will use each section. The only purpose of sections within
genconf_parameters is to set the scope of parameters like group_lines.

Am I correct to think I can't today generate
/etc/asterisk/dahdi-channels.conf files in which 2 groups of BRI ports are
defined ?



 --
   Tzafrir Cohen
 icq#16849755  
 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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

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

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

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

Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

[Olivier]

2009/11/10 Tzafrir Cohen tzafrir.co...@xorcom.com

 On Tue, Nov 10, 2009 at 06:06:12PM +0100, Olivier wrote:
  Hello,
 
  1. How can specify in /etc/dahdi/genconf_parameters file that a port from
 a
  B410P board is to be disabled.

 There's currently no way to do that.

 It should be trivial to implment. The more difficult part of it would be
 how to define exactly what spans / channels to disable.

 But why do you need that?


I don't really know why I thought I needed that feature but as some gateways
implement this feature (the ability to enable or disable each port), I must
have told myself I may have missed here.

On a general point of view, as most Dahdi cards have a light showing nearby
port status, it should ideally possible, to turn this light off when a port
is disabled.

But I must also add it doesn't seem very important to me to have this
implemented.


  Playing with comments (see bellow) doesn't help : file
  /etc/asterisk/dahdi-channels.conf is filled with 4 ports data.
 
  pri_termtype
  SPAN/1  TE
  SPAN/2  TE
  SPAN/3  TE
  #   SPAN/4  TE

 Currently pri_termtype is the only directive in dahdi_genconf that uses
 this list syntax. I'm not very happy with it.

 I'm not exactly sure if there should be some sort of generic way of
 adding per-span (span? channel? how do you define a span?) definitions.
 Think of ssh_config.



What about adding per-span section headers like Asterisk .conf files ?
[span1]
group_lines 1
pri_termtype
SPAN/1  TE
SPAN/2  TE

[span2]
group_lines 2
pri_termtype
SPAN/2  TE


I don't think we need to define any further what a span is, beside that
rules that applied to the whole genconf_parameters (no more than 1
group_lines statement) should apply to each section.


 
  2. How can specify groups in /etc/dahdi/genconf_parameters ?
  I would like to group SPAN/1 and SPAN/2 into group 1 and SPAN/3 into
 group
  2.
  I was unsuccessful with :
 
  group_lines 1
  pri_termtype
  SPAN/1  TE
  SPAN/2  TE
  group_lines 2
  pri_termtype
  SPAN/3  TE
 
  3. After a dahdi_genconf commande, generated
  /etc/asterisk/dahdi-channels.conf is like this :
  ; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS
  group=5,11
  context=remote
  switchtype = euroisdn
  signalling = bri_cpe_ptmp
  channel = 1-2
  context = default
  group = 63
 
  I can see 2 group= and context= lines. What is the difference between
  them ?
  Shall I care to have them both ?

 The second ones are not really needed. Unless you want to assume less of
 the configuration below.

 TODO: implement a [section] syntax there to care even less about such
 inclusion.

 --
   Tzafrir Cohen
 icq#16849755  
 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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

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

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

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

Re: [asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

[Tzafrir Cohen]

On Wed, Nov 11, 2009 at 08:24:53PM +0100, Olivier wrote:
 2009/11/10 Tzafrir Cohen tzafrir.co...@xorcom.com
 
  On Tue, Nov 10, 2009 at 06:06:12PM +0100, Olivier wrote:
   Hello,
  
   1. How can specify in /etc/dahdi/genconf_parameters file that a port from
  a
   B410P board is to be disabled.
 
  There's currently no way to do that.
 
  It should be trivial to implment. The more difficult part of it would be
  how to define exactly what spans / channels to disable.
 
  But why do you need that?
 
 
 I don't really know why I thought I needed that feature but as some gateways
 implement this feature (the ability to enable or disable each port), I must
 have told myself I may have missed here.
 
 On a general point of view, as most Dahdi cards have a light showing nearby
 port status, it should ideally possible, to turn this light off when a port
 is disabled.
 
 But I must also add it doesn't seem very important to me to have this
 implemented.

dahdi_genconf is an optional tool. Ideally it should need no
configuration at all and generate configuration that Just Works (though
the fact that it can do that indicates that the current defaults are
broken).

It should not be another configuration layer. If the configuration it
has generated is not good enough, you can also manually edit it.

 
 
   Playing with comments (see bellow) doesn't help : file
   /etc/asterisk/dahdi-channels.conf is filled with 4 ports data.
  
   pri_termtype
   SPAN/1  TE
   SPAN/2  TE
   SPAN/3  TE
   #   SPAN/4  TE
 
  Currently pri_termtype is the only directive in dahdi_genconf that uses
  this list syntax. I'm not very happy with it.
 
  I'm not exactly sure if there should be some sort of generic way of
  adding per-span (span? channel? how do you define a span?) definitions.
  Think of ssh_config.
 
 
 
 What about adding per-span section headers like Asterisk .conf files ?
 [span1]
 group_lines 1
 pri_termtype
 SPAN/1  TE
 SPAN/2  TE
 
 [span2]
 group_lines 2
 pri_termtype
 SPAN/2  TE

This implies you will know span numbers in advance. I would like better
ways to specify configuration.

 
 
 I don't think we need to define any further what a span is, beside that
 rules that applied to the whole genconf_parameters (no more than 1
 group_lines statement) should apply to each section.

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

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

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

[asterisk-users] Questions about Dahdi's /etc/dahdi/genconf_parameters

[Olivier]

Hello,

1. How can specify in /etc/dahdi/genconf_parameters file that a port from a
B410P board is to be disabled.
Playing with comments (see bellow) doesn't help : file
/etc/asterisk/dahdi-channels.conf is filled with 4 ports data.

pri_termtype
SPAN/1  TE
SPAN/2  TE
SPAN/3  TE
#   SPAN/4  TE

2. How can specify groups in /etc/dahdi/genconf_parameters ?
I would like to group SPAN/1 and SPAN/2 into group 1 and SPAN/3 into group
2.
I was unsuccessful with :

group_lines 1
pri_termtype
SPAN/1  TE
SPAN/2  TE
group_lines 2
pri_termtype
SPAN/3  TE

3. After a dahdi_genconf commande, generated
/etc/asterisk/dahdi-channels.conf is like this :
; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS
group=5,11
context=remote
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel = 1-2
context = default
group = 63

I can see 2 group= and context= lines. What is the difference between
them ?
Shall I care to have them both ?

Regards
___
-- 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] Questions about Dahdi's /etc/dahdi/genconf_parameters

[Tzafrir Cohen]

On Tue, Nov 10, 2009 at 06:06:12PM +0100, Olivier wrote:
 Hello,
 
 1. How can specify in /etc/dahdi/genconf_parameters file that a port from a
 B410P board is to be disabled.

There's currently no way to do that.

It should be trivial to implment. The more difficult part of it would be
how to define exactly what spans / channels to disable.

But why do you need that?

 Playing with comments (see bellow) doesn't help : file
 /etc/asterisk/dahdi-channels.conf is filled with 4 ports data.
 
 pri_termtype
 SPAN/1  TE
 SPAN/2  TE
 SPAN/3  TE
 #   SPAN/4  TE

Currently pri_termtype is the only directive in dahdi_genconf that uses
this list syntax. I'm not very happy with it.

I'm not exactly sure if there should be some sort of generic way of
adding per-span (span? channel? how do you define a span?) definitions.
Think of ssh_config.

 
 2. How can specify groups in /etc/dahdi/genconf_parameters ?
 I would like to group SPAN/1 and SPAN/2 into group 1 and SPAN/3 into group
 2.
 I was unsuccessful with :
 
 group_lines 1
 pri_termtype
 SPAN/1  TE
 SPAN/2  TE
 group_lines 2
 pri_termtype
 SPAN/3  TE
 
 3. After a dahdi_genconf commande, generated
 /etc/asterisk/dahdi-channels.conf is like this :
 ; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS
 group=5,11
 context=remote
 switchtype = euroisdn
 signalling = bri_cpe_ptmp
 channel = 1-2
 context = default
 group = 63
 
 I can see 2 group= and context= lines. What is the difference between
 them ?
 Shall I care to have them both ?

The second ones are not really needed. Unless you want to assume less of
the configuration below.

TODO: implement a [section] syntax there to care even less about such
inclusion.

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

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

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

[asterisk-users] Questions about app_jack.c [solved]

[Fabien COMTE]

Hello,
I corrected a bug and did some little optimizations in app_jack.c.

It works great now.

I propose this new file based on revision 140568.

Fabien

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2007 - 2008, Russell Bryant
 *
 * Russell Bryant russ...@digium.com
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */

/*!
 * \file
 * \brief Jack Application
 *
 * \author Russell Bryant russ...@digium.com
 *
 * This is an application to connect an Asterisk channel to an input
 * and output jack port so that the audio can be processed through
 * another application, or to play audio from another application.
 *
 * \arg http://www.jackaudio.org/
 *
 * \note To install libresample, check it out of the following repository:
 * code$ svn co
http://svn.digium.com/svn/thirdparty/libresample/trunk/code
 *
 * \ingroup applications
 */

/*** MODULEINFO
dependjack/depend
dependresample/depend
 ***/

#include asterisk.h

ASTERISK_FILE_VERSION(__FILE__, $Revision: 140568 $)

#include limits.h

#include jack/jack.h
#include jack/ringbuffer.h

#include libresample.h

#include asterisk/module.h
#include asterisk/channel.h
#include asterisk/strings.h
#include asterisk/lock.h
#include asterisk/app.h
#include asterisk/pbx.h
#include asterisk/audiohook.h

#define RESAMPLE_QUALITY 1

#define RINGBUFFER_SIZE 16384

/*! \brief Common options between the Jack() app and JACK_HOOK() function */
#define COMMON_OPTIONS \
s(name) - Connect to the specified jack server name.\n \
i(name) - Connect the output port that gets created to the
specified\n \
jack input port.\n \
o(name) - Connect the input port that gets created to the
specified\n \
jack output port.\n \
n - Do not automatically start the JACK server if it is not
already\n \
running.\n \
c(name) - By default, Asterisk will use the channel name for the jack
client\n \
name.  Use this option to specify a custom client name.\n

static char *jack_app = JACK;
static char *jack_synopsis = 
JACK (Jack Audio Connection Kit) Application;
static char *jack_desc = 
JACK([options])\n
  When this application is executed, two jack ports will be created; one
input\n
and one output.  Other applications can be hooked up to these ports to
access\n
the audio coming from, or being sent to the channel.\n
  Valid options:\n
COMMON_OPTIONS
;

struct jack_data {
AST_DECLARE_STRING_FIELDS(
AST_STRING_FIELD(server_name);
AST_STRING_FIELD(client_name);
AST_STRING_FIELD(connect_input_port);
AST_STRING_FIELD(connect_output_port);
);
jack_client_t *client;
jack_port_t *input_port;
jack_port_t *output_port;
jack_ringbuffer_t *input_rb;
jack_ringbuffer_t *output_rb;
void *output_resampler;
double output_resample_factor;
void *input_resampler;
double input_resample_factor;
unsigned int stop:1;
unsigned int has_audiohook:1;
unsigned int no_start_server:1;
/*! Only used with JACK_HOOK */
struct ast_audiohook audiohook;
};

static const struct {
jack_status_t status;
const char *str;
} jack_status_table[] = {
{ JackFailure,Failure },
{ JackInvalidOption,  Invalid Option },
{ JackNameNotUnique,  Name Not Unique },
{ JackServerStarted,  Server Started },
{ JackServerFailed,   Server Failed },
{ JackServerError,Server Error },
{ JackNoSuchClient,   No Such Client },
{ JackLoadFailure,Load Failure },
{ JackInitFailure,Init Failure },
{ JackShmFailure, Shared Memory Access Failure },
{ JackVersionError,   Version Mismatch },
};

static const char *jack_status_to_str(jack_status_t status)
{
int i;

for (i = 0; i  ARRAY_LEN(jack_status_table); i++) {
if (jack_status_table[i].status == status)
return jack_status_table[i].str;
}

return Unknown Error;
}

static void log_jack_status(const char *prefix, jack_status_t status)
{
struct ast_str *str = ast_str_alloca(512);
int i, first = 0;

for (i = 0; i  (sizeof(status) * 8); i++) {
if (!(status  (1  i)))
continue;

if (!first) {
ast_str_set(str, 0, %s, jack_status_to_str((1 
i)));
first = 1;
} else
ast_str_append(str, 

[asterisk-users] Questions about app_jack.c

[Fabien COMTE]

Hello,

My configuration is :
Card 0 - kernel dummy sound card
Card 1 - my soundcard

I have a jackd running in background. My jackd launch command is :
jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0
--capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2
--outchannels 2 --dither triangular 

1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to the
kernel dummy sound card (allow me dial command). I do a call with a
JACK_HOOK from app_jack.so, sound is sent but no one is received.

My extensions.conf :
exten = _0.,1,Answer 
exten =
_0.,n,Set(JACK_HOOK(manipulate,c(asterisk))i(from_voip:input)o(to_voip:outpu
t)))=on)
exten = _0.,n,Dial(SIP/freephonie-out/${EXTEN:1})

Asterisk command :
console dial 0

2) Jackd works well with anothers applications when I force them to use jack
as input/output. - probably not a jack configuration problem.

3) If I kill jackd and I use chan_alsa.so with the real soundcard, it works.
- probably not a network or sip configuration problem.


4) If I replace f_buf[i] = s_buf[i] * (1.0 / SHRT_MAX); with f_buf[i] =
0.5 * sin(0.3454 * ((float) i)); in app_jack.c and I retry the test 2, I
get test sound. 
It looks like no sound was read in channel...

Do you have any idea ?

Fabien





___
-- 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] Questions on X100P/X101P cards

[Tzafrir Cohen]

On Thu, May 07, 2009 at 07:24:21AM +0200, Massimo Nuvoli wrote:
 John Novack ha scritto:
  
 
  Not sure how you would do that, as the X100 card is an FXO card,
  won't provide either battery or dial tone to the cordless. What you
  will want for that is an FXS card or ATA. The X100 card will
  connect to a central office line, and with the later software echo
  cancel works OK. Not nearly as bad as some have made it out to be,
  though for US/Canada lines.  Not suitable for UK and others
 
 The problem is: analog line is a delicated environment where
 impedance, volts, and line quality are some of the critical components.
 
 I found my X100 cards failing in production, no software component can
  solve the line impedance or other physical things. I try but no way
 out.

Some X100P cards (e.g.: those that are based on SI3034, but not those
basedon SI3035) support programmable impedance settings. Sadly the
wcfxo driver does not support it.

Fixing it should mostly be a matter of lifting some code from wctdm.c
and adapting it. Shouldn't be much of an issue. Anybody wants to try
that?

(The cards I have at home are SI3035, sadly)

A more interesting task would be to add support for some newer
(soft/win-) modems. Anybody wants to try that?

The wcfxo driver needs some love and care. Don't expect Digium to do
that for you. They have more important stuff to do. Go and write your
own device drivers.

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

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

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

Re: [asterisk-users] Questions on X100P/X101P cards

[Vincent]

On Thu, 7 May 2009 09:32:19 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
Some X100P cards (e.g.: those that are based on SI3034, but not those
basedon SI3035) support programmable impedance settings. Sadly the
wcfxo driver does not support it.

Fixing it should mostly be a matter of lifting some code from wctdm.c
and adapting it. Shouldn't be much of an issue. Anybody wants to try
that?

(The cards I have at home are SI3035, sadly)

A more interesting task would be to add support for some newer
(soft/win-) modems. Anybody wants to try that?

The wcfxo driver needs some love and care. Don't expect Digium to do
that for you. They have more important stuff to do. Go and write your
own device drivers.

Thanks guys. So, provided the card has the right DAA chips to match
the country in which it is used (FCC or CTR21), all it takes to use
this hardware to handle a POTS line is patching Zaptel? IOW, the
hardware itself is good enough for SOHO use?

According to the following document, NovaVox (which no longer sells
X100P cards) provides a Zaptel patch for cards sold by X100P.com to
support non-FCC countries and UK CID:

This document describes how to configure an Open Source IP PBX with
an X100P Special Edition (SE) FXO PCI card installed to support Caller
ID received from a UK BT PSTN line. The configuration requires
implementing a patch for Asterisk®/Zaptel that was originally written
for the UK but has also been known to work in other countries.[...]

The Zaptel wcfxo driver has two user configurable modes of operation,
FCC to support US line standards and CTR21 to support European line
standards. The Silicon labs Si3012/Si3035 DAA chip used in the
original Digium X100P card and low cost X100P clone cards only
supports FCC mode. However, the Si3014/Si3034 DAA chip used on the
X100P SE supports global line standards.

X100P SE Setup Guide - Global Line Standards
http://novavox.co.uk/support/x100p.html
Richard Spencer supp...@novavox.co.uk


___
-- 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] Questions on X100P/X101P cards

[Tzafrir Cohen]

On Thu, May 07, 2009 at 11:47:03AM +0200, Vincent wrote:
 On Thu, 7 May 2009 09:32:19 +0300, Tzafrir Cohen
 tzafrir.co...@xorcom.com wrote:
 Some X100P cards (e.g.: those that are based on SI3034, but not those
 basedon SI3035) support programmable impedance settings. Sadly the
 wcfxo driver does not support it.
 
 Fixing it should mostly be a matter of lifting some code from wctdm.c
 and adapting it. Shouldn't be much of an issue. Anybody wants to try
 that?
 
 (The cards I have at home are SI3035, sadly)
 
 A more interesting task would be to add support for some newer
 (soft/win-) modems. Anybody wants to try that?
 
 The wcfxo driver needs some love and care. Don't expect Digium to do
 that for you. They have more important stuff to do. Go and write your
 own device drivers.
 
 Thanks guys. So, provided the card has the right DAA chips to match
 the country in which it is used (FCC or CTR21), all it takes to use
 this hardware to handle a POTS line is patching Zaptel? IOW, the
 hardware itself is good enough for SOHO use?

The problem with X100P and UK calelr ID is, if I understand it
correctly, that the card does not detect polarity reversal.

But doesn't it provide you with the raw amperage and voltage of the
line? This should allow detecting polarity reversal. And once that is
done, the driver can send up a polarity reversal event, and no change in
Asterisk is required.

This is all within wcfxo.c and thus has no effect on the performance of
other DAHDI devices.

It sounds so simple that there must have been a good reason why
something more complicated has been required.

 
 According to the following document, NovaVox (which no longer sells
 X100P cards) provides a Zaptel patch for cards sold by X100P.com to
 support non-FCC countries and UK CID:

Quoting later on:

| The DAA chip used in the X100P SE card is a Si3014/Si3034, which does
| support polarity reversal detection. However, because the X100P card 
| Zaptel driver does not include any polarity reversal detection code, 
| the X100P SE polarity detection feature cannot be used. An alternative 
| is to use a patch that uses a history buffer to store the CID value 
| written by Tony Hoyle. By the time the first ring arrives, the buffer 
| has a history of what was received immediately before so the CID 
| information can be extracted.

So, anybody wants to work on the code of wcfxo to add polarity
detection?

CPU is cheap (certainly so when referring to wcfxo, which will only
driver very few channels on a system), but this is still no excuse to
waste it.

Alternatively, some code to manually detect polarity reversal by
sampling amperage and voltage of the card may also help.

 
 This document describes how to configure an Open Source IP PBX with
 an X100P Special Edition (SE) FXO PCI card installed to support Caller
 ID received from a UK BT PSTN line. The configuration requires
 implementing a patch for Asterisk®/Zaptel that was originally written
 for the UK but has also been known to work in other countries.[...]
 
 The Zaptel wcfxo driver has two user configurable modes of operation,
 FCC to support US line standards and CTR21 to support European line
 standards. The Silicon labs Si3012/Si3035 DAA chip used in the
 original Digium X100P card and low cost X100P clone cards only
 supports FCC mode. However, the Si3014/Si3034 DAA chip used on the
 X100P SE supports global line standards.

This is the first thing I mentioned. Should be a relatively trivial
change in the driver.

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

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

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

Re: [asterisk-users] Questions on X100P/X101P cards

[Tzafrir Cohen]

On Thu, May 07, 2009 at 11:47:03AM +0200, Vincent wrote:

 X100P SE Setup Guide - Global Line Standards
 http://novavox.co.uk/support/x100p.html
 Richard Spencer supp...@novavox.co.uk

Another thing: their global-line-standard should basically (if
properly written) resolve http://bugs.digium.com/view.php?id=11057 . 
Though I guess the new code will actually be in DAHDI, as Zaptel is
frozen.

The driver already checks at init time the chip type and thus it is easy
to give different initialization / behaviour for si3034 and si3035.

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

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

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

Re: [asterisk-users] Questions on X100P/X101P cards

[Jonathan Moore]

On Wed, May 6, 2009 at 10:53 PM, John Novack
jnov...@stromberg-carlson.org wrote:
 Not sure how you would do that, as the X100 card is an FXO card, won't
 provide either battery or dial tone to the cordless.
 What you will want for that is an FXS card or ATA.
 The X100 card will connect to a central office line, and with the later
 software echo cancel works OK. Not nearly as bad as some have made it
 out to be, though for US/Canada lines.  Not suitable for UK and others

Ah, yes. Thanks for correcting me on that, I was getting some things
mixed up in my head.

-jonathan

___
-- 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] Questions on X100P/X101P cards

[Jon Pounder]

Jonathan Moore wrote:
 On Wed, May 6, 2009 at 10:53 PM, John Novack
 jnov...@stromberg-carlson.org wrote:
   
 Not sure how you would do that, as the X100 card is an FXO card, won't
 provide either battery or dial tone to the cordless.
 What you will want for that is an FXS card or ATA.
 The X100 card will connect to a central office line, and with the later
 software echo cancel works OK. Not nearly as bad as some have made it
 out to be, though for US/Canada lines.  Not suitable for UK and others
 

   
yeah I agree with the above - I never really found echo to ever be a 
problem, my only complaint was on some less than stellar cpu's I was 
having dtmf recognition problems.

 Ah, yes. Thanks for correcting me on that, I was getting some things
 mixed up in my head.

 -jonathan

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

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

   


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

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

Re: [asterisk-users] Questions on X100P/X101P cards

[Vincent]

On Thu, 07 May 2009 10:16:55 -0400, Jon Pounder j...@inline.net
wrote:
yeah I agree with the above - I never really found echo to ever be a 
problem, my only complaint was on some less than stellar cpu's I was 
having dtmf recognition problems.

BTW, can someone explain to a libart major like me (;-)) where echo
comes on in a telephone conversation? I seem to recall it's due to the
length of the line between the CO and the local party, but I'm not
sure.

Is there a way to keep track of this issue, and overtime, to configure
it to answer a call by expecting such and such echo, and thus, avoid
starting sampling from scratch every time?

Thank you.


___
-- 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] Questions on X100P/X101P cards

[Vincent]

On Thu, 7 May 2009 13:40:20 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
Another thing: their global-line-standard should basically (if
properly written) resolve http://bugs.digium.com/view.php?id=11057 . 
Though I guess the new code will actually be in DAHDI, as Zaptel is
frozen.

Ah yes, I seem to remember Zaptel had to change their name to DAHDI
for some reason.

Is there a mailing list to ask for more information about
Zaptel/DAHDI?
http://lists.digium.com/mailman/listinfo/


___
-- 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] Questions on X100P/X101P cards

[Wilton Helm]

BTW, can someone explain to a libart major like me (;-)) where echo
comes on in a telephone conversation? I seem to recall it's due to the
length of the line between the CO and the local party, but I'm not
sure.

Yes, I'll tackle that.  It takes a finite amount of time for the electrical
signal originating in one phone to arrive at another phone over whatever
path it is taking.  If the path is copper, that time can be fairly small.
If the path is satellite, the time will have to exceed the 1/4 second round
trip to the bird.  If there are SIP packets involved, the time must be
larger than twice the packet size because of the time taken to collect the
data in the packet and then to serialize it at the other and after it
arrives.  If the path involves the internet, there is the path delay there
to be added in (ping will give you an idea of what that is, but it can often
be 50 - 200 ms).

All of this constitutes a delay.  It can be a bit annoying in its own right
because one person asks a question, and twice the delay time elapses before
they start hearing the answer.  However, if there are POTS analog circuits
involved anywhere, a second factor comes into play.  A POTS analog circuit
is two wires, which carry an electrical representation of sound.  Both sides
of the conversation are carried over the same wire. (its called a 2 wire
circuit.  There are also four wire circuits where each direction travels on
a separate pair of wires.  They don't have echo problems.  Digital circuits
also have separate paths for each direction, so are immune to echo)

The problem with a two wire circuit is how to separate the sound going in
both directions.  That is done by something called a 2 wire to 4 wire
converter, also commonly known as a hybrid.  It basically works by
subtracting out what it knows is being sent at the near end from what it
sees on the wire.  If that subtraction is perfect, only what came from the
other end is left and that is presented to the listener.  In the real world,
this isn't perfectly possible, but it can be done fairly well.

However, there is a side effect that comes with the transition from two wire
to four wire.  Some of the signal originating at one end of the wire gets to
the other end and is reflected back.  For an analogy, tie the end of a long
rope to a pipe, stretch it out and snap the other end.  You will see a wave
travel to the pipe and then come back.  If you were able to attach the rope
to the pipe with a suitable dashpot or something that would fully absorb the
wave, nothing would come back.  This reflection from the other end is the
cause of echo.  If the path is terminated in exactly the correct impedance,
there would be no echo.  However, for real circuits over the range of
frequencies that make up sound, that impedance is a complex quantity, and
cannot be exactly matched.

The bottom line is that any circuit with one or more 2 wire analog portions
is going to have some echo.  Since most of the circuits provided by a phone
company are POTS, they are two wire analog from the subscriber to the CO.
If the subscriber equipment is Asterisk, then a 2 wire to 4 wire conversion
and digitization takes place there.  Likewise virtually all telco links are
digital and a conversion takes place in the switch in the CO.  Then at the
other end the process is repeated.  That makes a total of 4 interfaces where
echo can originate in a typical phone call.  If part of the call is SIP, or
internet or satellite, the delay is large enough to guarantee it will be
noticeable.  Since there are several interfaces there can be several echoes.

Another example that illustrates the concept is a speaker phone.  If the
person on the other end is using a speakerphone, then some of what you say
comes out of the speakerphone, bounces off the walls of the room, gets
picked up in the mic and comes back to you.  Again, if the delay is very
large, it will be an echo by the time it gets back to you.  Speakerphones
(if they are full duplex--i.e. allow both parties to talk at once) have to
have echo cancellers to prevent this from happening.


Is there a way to keep track of this issue, and overtime, to configure
it to answer a call by expecting such and such echo, and thus, avoid
starting sampling from scratch every time?

Unfortunately not.  If you've followed the discussion to this point, you
understand that the magnitude (loudness) of the echo depends on the
impedance mismatch which is unique to the circuitry at each end (for a
typical call) of the call.  The delay time is unique to the call path, which
is likely different for each call, and in the case of internet calls, can
vary within the call.  The echo canceller must constantly do pattern
matching to recognize changes and adjust for them.  Its job is to
subtracting out a signal of exactly the same amplitude as the echo, but of
the opposite polarity and delayed by exactly the path delay the echo is
travelling through.  Since there can easily be four or more 

[asterisk-users] Questions on X100P/X101P cards

[Vincent]

Hello,

I'm looking for a dirt cheap solution for SOHO use to handle at most
a couple of POTS lines, and I notice that X10?P cards go for $15 on
eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma.

I have a couple of questions about those cheap FXO cards:

1. Are they all glorified softmodems, ie. none has an on-board CPU or
DSP and outsources all processing to the computer's CPU?

2. Are they all bad, no matter what chipset is used (Intel, Motoral,
Ambient)? If not, which offer good enough quality to handle a single
POTS line?

3. Why are they often bad quality? Because the driver itself is badly
written? Because PC's don't have enough speed to handle the tasks
using their own CPU (hard to believe, but I don't know)?

Thank you.


___
-- 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] Questions on X100P/X101P cards

[Alan Lord (News)]

On 06/05/09 13:43, Vincent wrote:
 Hello,

   I'm looking for a dirt cheap solution for SOHO use to handle at most
 a couple of POTS lines, and I notice that X10?P cards go for $15 on
 eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma.

 I have a couple of questions about those cheap FXO cards:

 1. Are they all glorified softmodems, ie. none has an on-board CPU or
 DSP and outsources all processing to the computer's CPU?

 2. Are they all bad, no matter what chipset is used (Intel, Motoral,
 Ambient)? If not, which offer good enough quality to handle a single
 POTS line?

 3. Why are they often bad quality? Because the driver itself is badly
 written? Because PC's don't have enough speed to handle the tasks
 using their own CPU (hard to believe, but I don't know)?

Hi Vincent,

I bought a cheap eBay X100p card over a year ago. When I first tried 
it was appalling. I couldn't get rid of the echo and noise no matter what.

I then came across OSLEC (at the time a new Free Echo Canceller). A bit 
of hacking to get it to work and hey-presto! No more echo.

I have been using the same card ever since with no noticeable issues.

I think OSLEC is now the default EC for many distributions so I would 
have thought you will be fine although, of course, YMMV.

For a cheap backup to your VOIP service they do the job. I wouldn't use 
them for a proper system though.

HTH

Al


___
-- 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] Questions on X100P/X101P cards

[Vincent]

On Wed, 06 May 2009 14:02:20 +0100, Alan Lord (News)
alansli...@gmail.com wrote:
For a cheap backup to your VOIP service they do the job. I wouldn't use 
them for a proper system though.

Thanks for the feedback. I have two more questions:
1. Can the OSLEC echo canceller run OK on an 1.6GHz Intel Atom not
doing much more than this and running Asterisk?
2. Is it good enough to handle a single FXO line for professional use?
3. Can you give me a pointer about which X100p you bought on eBay?
AFAIK, there are three chipsets : Intel, Motorola, and Ambient.

Using a $15 card over an $80 card is not insignificant because I could
then sell a small server for $99.

Thank you.


___
-- 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] Questions on X100P/X101P cards

[Gordon Henderson]

On Wed, 6 May 2009, Vincent wrote:

 On Wed, 06 May 2009 14:02:20 +0100, Alan Lord (News)
 alansli...@gmail.com wrote:
 For a cheap backup to your VOIP service they do the job. I wouldn't use
 them for a proper system though.

 Thanks for the feedback. I have two more questions:
 1. Can the OSLEC echo canceller run OK on an 1.6GHz Intel Atom not
 doing much more than this and running Asterisk?

The OSLEC benchmark tell me it can run 14 concurrent instances on a 550MHz 
VIA C3 processor. On a 1.6GHz Atom, it tells me it can do 80 concurrent 
instances, so I think the overhead of OSLEC is the least of your problems 
there.

 2. Is it good enough to handle a single FXO line for professional use?

I use OSLEC in my standard PBX products - Not with x100p cards though, but 
with Digium and OpenVox cards. However I'm in the UK - your lines may be 
different...

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