[asterisk-users] CallerPres SIP headers Analog Phone

[Ketema Harris]

Yes. I am able to match the *67 and appropriately set the  
SetCallerPres when SIP phones make calls because the *67 is passed  
through and can be matched.
However on my analog handset its as if the *67 is processed and  
discarded.  Here is my chan_dahdi.conf and a snippet of console output:

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=1.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

context=from_telco
group=0
echocancel=yes
signalling=fxs_ks
channel => 1

context=from_telco
group=0
echocancel=yes
signalling=fxs_ks
channel => 2

context=fax
group=1
echocancel=yes
signalling=fxo_ks
channel => 3

context=analog_phone
group=1
echocancel=yes
signalling=fxo_ks
channel => 4

extensions.conf snippet

[analog_phone]
immediate=no  ;This line tells asterisk to wait for input from the  
analog phone then continues on
include => default

[default]
exten => _*XX.,1,Goto(outbound,${EXTEN},1)
exten => _*XX.,n,Hangup()
exten => _X.,1,Goto(outbound,${EXTEN},1)
exten => _X.,n,Hangup()

[outbound]
exten => _*67NXXNXXXXXX,n,SIPAddHeader(Remote-Party-ID: <sip:XXXXXXXXXX at sipprovider:5060 
\;user=phone>\;party=calling\;screen=yes\;privacy=full)
exten => _*67NXXNXXXXXX,n,Noop(${CALLERID(num)})
exten => _*67NXXNXXXXXX,n,Dial(SIP/provider/${EXTEN:3})

Again for SIP handset the above works fine, but here is what an analog  
phone does:

-- Starting simple switch on 'DAHDI/4-1'  --THIS WHEN THE PHONE GOES  
OFFHOOK
     -- Disabling Caller*ID on DAHDI/4-1   --THIS IS AS SOON AS *67 is  
PRESSED
     -- Executing [XXXXXXXXXX at analog_phone:1] Goto("DAHDI/4-1",  
"outbound,XXXXXXXXXX,1") in new stack --THE REMAINING DIGITS ARE  
PASSED TO OUTBOUND
     -- Goto (outbound,XXXXXXXXXX,1)								     --BUT CAN'T MATCH  
because the *67 is STRIPPED
     -- Executing [XXXXXXXXXX at outbound:1] Dial("DAHDI/4-1", "SIP/ 
provider/XXXXXXXXXX") in new stack

I'd like to know if this is because of the phone handset, the analog  
card, or something else.  Obviously there has to be a way to either  
capture the *67 from the handset or utilize the fact that apparently  
the card is detecting that *67 was pressed, and get it to properly set  
the CallerPres()

Thanks

On Jul 22, 2009, at 11:36 AM, Philipp Kempgen wrote:

> Ketema Harris schrieb:
>> hello all...I have been trying to get a handle on CallerPres with an
>> analog handset.  I have usecallingpres=yes in my chan_dahdi.conf file
>> and when I dial *67 on my analog handset I see Disabling Caller*ID on
>> DAHDI/4-1 but when the call is then forwarded to my outbound SIP
>> provider the RPID header is not correct privacy=off;screen=no  
>> instead of
>> full and yes how can I correct this?
>
> I don't know if/how Asterisk handles/stores CLIR for analog handsets
> but SetCallerPres(prohib_passed_screen) does the trick when dialing
> to a SIP channel.
>
> Remote-Party-ID: ...;privacy:full;screen:yes
>
> You could add a *67 extension to your dialplan and store the CLIR
> state in AstDB for example.
>
>
>    Philipp Kempgen
> -- 
> AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
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