asterisk users - Feb 2005 - astGUIclient users should not upgrade to Asterisk 1.0.5

Hello,

Just confirmed this on my end, because of the massive changes that have been
made to callerID handling in asterisk 1.0.5 many of the features of the
astGUIclient suite will not work on this new version. The latest stable
version recommended is Asterisk 1.0.3. We will work on trying to find ways
around the new callerID rules that the asterisk developers have put in place
and hope to have the next release version of the suite be compatible with
asterisk v1.0.5

Here's a more in-depth explanation of the problem:

In Asterisk v1.0.5 when you place an outbound phonecall(Zap, SIP, IAX2),
once the call picks up, Asterisk will change the callerid to the number that
you just dialed, no matter if you set a custom callerID for that call. When
the callerID is changed it is noted in the manager output only by a uniqueid
of the call. Sadly this does not help very much when you place calls to the
Local/ trunk, which can often result in 3 separate uniqueids for a single
call. That used to only be fully traceable with the custom callerID that you
used to be able to define when the call was started. Now we will need to
figure out some way of tracing all uniqueIDs for every call and try to
determine which call instance a specific call event is referring to even
though it may now have the same callerID. I am also hoping that we may be
able to get asterisk to keep a custom callerid value in the calleridname
field. If all else fails, we may have to resort to including an asterisk
source patch that would reinstate custom callerid values throughout a call's
life, in effect disabling some of these new callerid changes.

Thanks,

MATT---

I'm not a astGUIclient user, but I'm puzzled by the following statement:

mattf <mattf@vicimarketing.com> wrote:
> 
> In Asterisk v1.0.5 when you place an outbound phonecall(Zap, SIP, IAX2),
> once the call picks up, Asterisk will change the callerid to the number
that
> you just dialed, no matter if you set a custom callerID for that call.
What you've said there suggests that the CallerID is being set to the
DESTINATION number, which sounds to me not what CallerID should be at all.
CallerID normally indicates the source of a call.

OK, as I wrote that and re-read what you said, it occurred to me that
perhaps what you are saying is that you are using "Action: Originate"
in
the manager API. I believe that having successfully made the call to the
"Channel" parameter, Asterisk then needs to connect that call to the
Context / Exten / Priority specified, *as if it were an incoming call*. I
guess that would explain why the CallerID is being set to the number that
was called by the channel, as that is where the notional "incoming"
call
would have been coming from.

I must admit, I've had similar problems to those you've described, when
using Local channels for this purpose, as the failure status for failed
calls (busy, etc) comes in on the wrong channel. I have to work out a way
of tracking those over the next day or two!

Cheers
Tony
-- 
Tony Mountifield
Work: tony@softins.co.uk - http://www.softins.co.uk
Play: tony@mountifield.org - http://tony.mountifield.org

Please help,
    anyone out there have a fix for the 403 forbidden error...i am 
running asterisk with AMP but cannot get my budget tone 100 phones to 
register with my sip server....  i think the problem lies with the fact 
that the sip_additional.conf creates the call plan [ext-local] and not 
[from-sip]..I am a newbie to unix as well as sql administration and dont 
know how to manually add the [from-sip] dial plan to the amp 
database.... it could be a multitude of problems and what I think is the 
problem might not be causing it..i am really lost and have searched for 
fixes to my problem online and saw people with the same problem but what 
they did to fix it is not helping me..i keep getting the same 
error...please help....have been stuck on this problem for several days 
and dont know what to do now other than to just run asterisk without  
the nice amp web interface....the problem is probably a quick and easy 
fix for any one of you out there but I cannot seem to get it working 
seamlessly....PLEASE HELP

Thanks in advance,

Ken

I have added a simple patch to the bugnote for this issue:
http://bugs.digium.com/bug_view_page.php?bug_id=0003490

All it really does is delete the code in app_dial.c that wipes out the
callerID. But astGUIclient now runs properly on Asterisk 1.0.5 with this
patch applied. I will also post the patch on the astGUIclient web site.

Still I do believe that this feature is not a bad one, just very poorly
implemented. It really should be an OPTIONAL dial flag not a manditory
hard-coded feature.

MATT---


-----Original Message-----
From: Nicol?s Gudi?o [mailto:asternic@gmail.com]
Sent: Wednesday, February 02, 2005 11:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: astGUIclient users should not upgrade
to Asterisk 1.0.5


Hello,
> I'm not a astGUIclient user, but I'm puzzled by the following
statement:
> 
> mattf <mattf@vicimarketing.com> wrote:
> >
> > In Asterisk v1.0.5 when you place an outbound phonecall(Zap, SIP,
IAX2),
> > once the call picks up, Asterisk will change the callerid to the
number
that
> > you just dialed, no matter if you set a custom callerID for that call.
> 
> What you've said there suggests that the CallerID is being set to the
> DESTINATION number, which sounds to me not what CallerID should be at all.
> CallerID normally indicates the source of a call.
Just wanted to say that Flash Operator Panel users will have the same
problem. I'm puzzled too. IMHO there's something missing or wrong in
the new callerid handling.

If you trace the manager events and try to match the callerid via
Uniqueid, you will notice that the only way to have a match is *after*
the call is bridged. That means that you cannot  find the callerid of
a call before you pick up the phone. At least thats what I'm seing on
Asterisk 1.0.5. (did not try with HEAD) So, the callerid is plain
useless (Users expect to see the callerid before picking it up, dont't
they?)

It would be nice to have the callerid available on the manager when a
phone is RINGING and before picking it up.

I did not look at the Local channels, and it seems that it makes
things harder.. but I still think that we do not have to code
workarounds on manager based applications. We need to have an event in
the manager informing the callerid of the caller in the RINGING event
or associated directly with the Uniqueid of the callee.

Personally I had to downgrade app_dial.c to a previous releaes to get
the callerid as before.

Just my 2 cents...


-- 
Nicol?s Gudi?o
Buenos Aires - Argentina
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Hello,

Mark has changed app_dial.c in CVS-HEAD to allow for a 'p' dial flag
that
will allow for the original callerid to not be altered. If you are using
CVS-HEAD, simply put the p flag at the end of your Dial string in your
extensions.conf file for 91NXXNXXXXXX Dial TRUNK step(or wherever you dial
out). Or if you are using Asterisk 1.0.5 simply use the patch mentioned
before to eliminate callerid altering completely.

Thanks Mark!

MATT---


-----Original Message-----
From: mattf [mailto:mattf@vicimarketing.com]
Sent: Wednesday, February 02, 2005 12:32 PM
To: 'Nicol?s Gudi?o'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] Re: astGUIclient users should not upgrade
to Asterisk 1.0.5


I have added a simple patch to the bugnote for this issue:
http://bugs.digium.com/bug_view_page.php?bug_id=0003490

All it really does is delete the code in app_dial.c that wipes out the
callerID. But astGUIclient now runs properly on Asterisk 1.0.5 with this
patch applied. I will also post the patch on the astGUIclient web site.

Still I do believe that this feature is not a bad one, just very poorly
implemented. It really should be an OPTIONAL dial flag not a manditory
hard-coded feature.

MATT---


-----Original Message-----
From: Nicol?s Gudi?o [mailto:asternic@gmail.com]
Sent: Wednesday, February 02, 2005 11:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: astGUIclient users should not upgrade
to Asterisk 1.0.5


Hello,
> I'm not a astGUIclient user, but I'm puzzled by the following
statement:
> 
> mattf <mattf@vicimarketing.com> wrote:
> >
> > In Asterisk v1.0.5 when you place an outbound phonecall(Zap, SIP,
IAX2),
> > once the call picks up, Asterisk will change the callerid to the
number
that
> > you just dialed, no matter if you set a custom callerID for that call.
> 
> What you've said there suggests that the CallerID is being set to the
> DESTINATION number, which sounds to me not what CallerID should be at all.
> CallerID normally indicates the source of a call.
Just wanted to say that Flash Operator Panel users will have the same
problem. I'm puzzled too. IMHO there's something missing or wrong in
the new callerid handling.

If you trace the manager events and try to match the callerid via
Uniqueid, you will notice that the only way to have a match is *after*
the call is bridged. That means that you cannot  find the callerid of
a call before you pick up the phone. At least thats what I'm seing on
Asterisk 1.0.5. (did not try with HEAD) So, the callerid is plain
useless (Users expect to see the callerid before picking it up, dont't
they?)

It would be nice to have the callerid available on the manager when a
phone is RINGING and before picking it up.

I did not look at the Local channels, and it seems that it makes
things harder.. but I still think that we do not have to code
workarounds on manager based applications. We need to have an event in
the manager informing the callerid of the caller in the RINGING event
or associated directly with the Uniqueid of the callee.

Personally I had to downgrade app_dial.c to a previous releaes to get
the callerid as before.

Just my 2 cents...


-- 
Nicol?s Gudi?o
Buenos Aires - Argentina
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users