Hello.
After doing much web searching and searching archives of this mailing
list, I see that my question has been asked at least 6 separate times
but no answers have been attached.
In a nutshell, is there a way for a SIP phone to easily hook flash a ZAP
analog trunk mid-call? (This is important when trying to make use of
features on a PSTN analog line such as call waiting, call forwarding,
3-way calling, etc.)
I've seen the *3 trick (mentioned in the comments here
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Flash ), but I
don't know in which context to put the extensions.conf stuff:
exten => s,n,Set(DYNAMIC_FEATURES=zapflash)
exten => s,n,Dial(SIP/xxxx,15,tw)
...to make it take effect while on a call.
My configuration:
- 1 X100P card with one analog line attached with call waiting, caller
ID, 3-way calling enabled on the line.
- 1 Snom 300 SIP desk phone
My Zapata.conf:
[trunkgroups]
; define any trunk groups
[channels]
; hardware channels
; default
usecallingpres=yes
usecallerid=yes
cidstart=ring
cidsignalling=bell
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=no
callreturn=no
immediate=no ; for Caller ID, allows time for telco to send digits
hidecallerid=no
echocancel=yes
echotraining=yes
useincomingcalleridonzaptransfer=yes
context=from-pstn ; Incoming calls go to [from-pstn]
signalling=fxs_ks ; Use FXS signalling for an FXO channel
group=0 ; Use with Zap/g0
channel => 1 ; PSTN attached to port 1
Thank you very much for your time. Hopefully we can get an answer to
this and put it on the Wiki for all to see!
Sincerely,
Sean M. Pappalardo
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