[Asterisk-Users] Problem with call pickup
Title: Message I have configured call pickup, and this works fine. Although there are 2 problems, perhaps anyone would know a solution to this; - When I pickup a call from another set, the *8 code keeps being displayed in my screen (Snom 220). I would like it to show the phonenumber of the person calling me. - When a caller that I've answered through Call-Pickup disconnects, my phone does not close the connection but acts like there is still someone on the otherside. (Logging shows dat de Zap/channel has cleared, but not the SIP/channel) I use Asterisk 1.0.2-BRIstuffed-0.2.0-RC2 Any help would be greatly appreciated... Ramon ___ 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
Re: [Asterisk-Users] Problem with call pickup
Ramon Peek wrote: - When I pickup a call from another set, the *8 code keeps being displayed in my screen (Snom 220). I would like it to show the phonenumber of the person calling me. This is correct. You are placing a call to *8 which just happens to connect you to caller. As far as your phone is concerned it is talking to someone at *8. ___ 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] Problem with call pickup
Title: Message
I know that my
phone displays *8 because I dailed that.
But it's
definitly not what I would want, or most other people.
Any other
ordinary PBX would show the CID of the caller, but because this is a SIP-based
system we get this problem.
I was thinking
more in line of an alternate call-pickup procedure to realize this
option.
My idea would
be:
exten =
*8,1,SetVar(PICKEXT=${CALLERIDNUM})
exten =
*8,2,HangUp
exten =
*8,3, Here come the lines that will deflect the call to
$PICKEXT
Why
deflect?, well when a call is deflected CID information is also
transferred.
The
effectwould bethat when dialing *8the connection would be
closed, but immediatly after that your phone will ring showing you all the
information you need.. even before pickup.
You could
call this function remote deflecting???
This
function does not exist in * as far as I know, but perhaps there is some
work-around to this???
Anyone?!??!
I have configured
call pickup, and this works fine.
Although there are 2
problems, perhaps anyone would know a solution to this;
- When I pickup a
call from another set, the *8 code keeps being displayed in my screen (Snom
220).
I would like
it to show the phonenumber of the person calling me.
- When a caller that
I've answered through Call-Pickup disconnects, my phone does not close the
connection but acts like there is still someone on the otherside. (Logging shows
dat de Zap/channel has cleared, but not the SIP/channel)
I use Asterisk
1.0.2-BRIstuffed-0.2.0-RC2
Any help would be
greatly appreciated...
Ramon
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[Asterisk-Users] Problem with call pickup -or- what stupid mistake have I made?
For some reason, I can't get call pickup to work between Sip phones or between Sip and Zap phones. All phones are in the same call group and pickup group (1). The source code was downloaded and built as of today 11/15/03. Here's what's in sip.conf: [general] port=5060 bindaddr=192.168.17.2 tos=lowdelay disallow=all allow=ulaw context=aliens ; ; SIP Entry for sipura line 1 ; This phone is allowed to dial extensions and local phone numbers ; [101] type=friend host=dynamic context=house-toll reinvite=no canreinvite=no qualify=300 secret=x callerid=Sipura Line 1 101 username=101 callgroup=1 pickupgroup=1 [EMAIL PROTECTED] nat=0 ; Sample for sipura line 2 ; This phone is allowed to dial extensions and local phone numbers ; [102] type=friend host=dynamic context=house-toll reinvite=no canreinvite=no qualify=300 secret=y callerid=Sipura Line 2 102 username=102 callgroup=1 pickupgroup=1 [EMAIL PROTECTED] nat=0 Here's what's in zapata.conf: ; ; zapata.conf ; [channels] ; ; 2 ea. X100P plugged into PSTN ; context=from-pstn-xx signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes rxgain=1.5 txgain=1.5 immediate=no busydetect=no callprogress=no musiconhold=default usecallerid=no ;callerid=asreceived group=1 channel=1 ; context=from-pstn-xx channel=2 ; ; TDM100B Port #1 plugged into analog Phone ; This phone is allowed to dial extensions and local and long distance numbers ; amaflags=documentation context=house-admin signalling=fxo_ks callwaiting=yes callwaitingcallerid=no threewaycalling=yes cancallforward=yes transfer=yes echocancel=yes echocancelwhenbridged=yes echotraining=no relaxdtmf=yes rxgain=1 txgain=1 immediate=no musiconhold=default usecallerid=yes callerid=TDM100 Zap Phone 103 callgroup=1 pickupgroup=1 group=2 channel=3 Here's the verbose output from the console when *8# is dialed on a sip phone: -- Starting simple switch on 'Zap/3-1' -- Executing SetVar(Zap/3-1, ALERT_INFO=Bellcore-r3) in new stack -- Executing Dial(Zap/3-1, SIP/[EMAIL PROTECTED]:5060|20) in new stack -- Called [EMAIL PROTECTED]:5060 -- SIP/192.168.17.6-67bd is ringing NOTICE[1142127920]: File chan_sip.c, Line 5090 (handle_request): Nothing to pick up == Spawn extension (house-admin, 101, 2) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
