Ready for Stage 2

I run Asterisk 1.4.11 on a FreeBSD-6.2 server with a single X101P card and two IP phones on a 10/100 LAN. There are also several plain analog phones plugged into the existing POTS jacks around the house.

The POTS calls come and go on the * server without any problems these days. There were big problems before, but since 1.4.10 came out that seems to have settled down.

My * server will let the POTS calls ring for 20 seconds so I can try to answer the call on one of the regular analog phones. Then it will take the call and allow the user to dial one of the SIP extensions if they want. If they don’t select an extension it sends them to voicemail.

I want to be able to pickup the calls on one of the sip phones during the initial 20 second wait period. But my dialplan doesn’t exactly answer the phone until after this 20 second wait period.

[incoming]
exten => s,1,Wait,20 ; Wait 20 seconds
exten => s,n,Answer() ; Answer the call
exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=7) ; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(welcome) ; Play the initial welcome message
exten => s,n,WaitExten(7) ; Wait for an extension to be dialed.
exten => s,n,Voicemail(201@default,u) ; Send Caller to Voicemail

Since I need the analog phones more than the sip phones I’m wondering if this is possible. Can I get something to work so that I can pickup the incoming call from one of the sip phones?

Also, whenever I’m on a sip phone making a call through the X101P card on the POTS line and I hear the “beep” of another incoming call, I cannot accept it. This only works when I’m on one of the analog phones. Is there a way to get the sip phones working with Call Waiting? My zapata.conf has call waiting enabled and these phones are supposed to work with call waiting. But I cannot accept a second POTS call when on the SIP phones.

Any advice would be appreciated.

.

Try pickup groups

look at zapata.conf example is below;

[channels]
faxdetect=incoming

usecallerid=yes
callerid=asreceived
hidecallerid=no
sendcalleridafter=2

hanguponpolarityswitch=yes
busydetect=no
echocancel=256
immediate=no

musiconhold=default

rxgain=7.0
txgain=5.0

jitterbuffers=4

context=pcg-incoming
group=1
signalling=fxs_ks
channel=>1,2,3

In SIP.conf

[515]
tos=184
disallow=all
allow=ulaw
allow=g729
mailbox=515
type=friend
username=515
callerid=515
host=dynamic
context=internal
canreinvite=no
secret=1234
nat=no
qualify=yes
callgroup=1
pickupgroup=1

look in features.conf
pickup default is *8

So if you here the phone ringing dial *8 to pick it up asuming your belong to the correct pickup-group

Thanks for your reply. I took a look at your suggestion and then added the following lines to my zapata.conf file

[channels]
language=en
context=default
signalling=fxs_ks
channel=>1
disconnectsupervision=yes
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=no
transfer=yes
echocancel=yes
echotraining=yes
echocancelwhenbridged=yes
immediate=no
group=1

Here is the entry for the sip phone I want to pickup the call from:

[2000]
context=default
type=friend
username=2000
secret=xxxxxx
nat=yes
host=dynamic
canreinvite=no
qualify=yes
dtmfmode=inband
callgroup=1
pickupgroup=1

So I made an incoming call to the * server. After a ring or two I tried to pickup the call from the sip phone by dialing *8 but all I get is this on the CLI:

-- Starting simple switch on 'Zap/1-1'
-- Executing [s@default:1] Wait("Zap/1-1", "20") in new stack

[Oct 25 19:18:00] NOTICE[26421]: chan_sip.c:13754 handle_request_invite: Nothing to pick up for 8d19464684476598@10.0.0.198

asterisk# == Spawn extension (default, s, 1) exited non-zero on ‘Zap/1-1’
– Hungup ‘Zap/1-1’

Then I hear a busy signal in the sip phone handset.

You don’t have a callgroup & pickupgroup in zapata.conf
That is why asterisk reckons there is “Nothing to pick up”

[channels]
callgroup=1
pickupgroup=1

Good luck!