[asterisk-users] No CallerID Transfer Problem

[Ken Williams]

Actually, the code below works perfectly to fix the transfer disconnect
problem.  I was asking of other, better ways, aside from manually
defining on all incoming calls a dummy CID.

To answer Steve's question, using a single TDM400 card for the incoming
PSTN (it's one line, a remote office that most of their communication is
done over IAX back to our main location).  The three handsets are
Grandstream GXP-2000 (let the flaming begin, we currently have about 40
GXP-2000's in production and yes, we've had strange issues, but they're
working quite well now).

Anyway, it's really not a huge deal, but I had work arounds.  I'd prefer
the 'usecallerid=no' type route instead of making a fix in the dialplan,
that's all I was looking for.

Ken 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric
Wieling
Sent: Friday, April 25, 2008 7:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No CallerID Transfer Problem

Try removing the quotes from the Caller*ID info.

Steve Davies wrote:
> 2008/4/24 Ken Williams <ken at intermountainelectronics.com>:
>> Came upon a problem today that I thought I'd see if it's by design, 
>> if I'm missing an option somewhere, or if my fix is the way to fix
it.
>>
>> We setup a remote location with a server, same as we've done with 
>> others, but for some reason when they would transfer an outside call 
>> anywhere it would pause for a few seconds and hang up the line.
>>
>> Well, after spending most of the day on it, it turns out it's because

>> they don't have callerID on the PSTN lines coming in through zaptel.

>> My first thought was, set "usecallerid=no" and all would be well, but

>> this didn't do any good.  After playing a bit longer I just set the
following:
>>
>>         exten => 900,2,set(CALLERID(num)="606-555-1212")
>>         exten => 900,3,set(CALLERID(name)="Outside Call")
>>         exten => 
>> 900,4,Dial(${DIALEXTENSIONS},${RINGTIMER},${DIAL_OPTIONS})
>>
>> Now all works well.
>>
>> So is there another option somewhere to keep asterisk from killing a 
>> transfer without callerid?  This happened on both 1.4.17 & 1.4.18.1.
>>
>> Thanks,
>> Ken
> 
> Can I guess that they are using snom phones with firmware 7.1.30? I 
> encountered exactly that bug here, but only if I enabled "sendrpid" in

> the sip.conf of the asterisk system. Downgrading to a more-stable 
> 6.5.x snom firmware, or disabling "sendrpid" for all of the snom 
> devices fixed this in our case. (Roll on the next snom firmware
> release!)
> 
> If not, then can I suggest that you provide more detail of equipment 
> involved - PCI cards, handsets etc etc?
> 
> Hope that helps,
> Steve
> 
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