[asterisk-users] Call Pickup with 1.6.2.1 and Snom

cool dude cool_dudeoflko at yahoo.co.in
Sat Feb 13 00:51:16 CST 2010 


hi friend need ur help in dial plan, i want to allow exten 2000 to 2005 can make call outside and exten 2006 to 2010 can not make call outside. heres my dial plan.
 
sip.conf
 
[general]
port = 5060
bindaddr = 0.0.0.0
context = others
[2000]
type=friend
context=outside
secret=1234
host=dynamic

[2001]
type=friend
context=outside
secret=1234
host=dynamic

[2002]
type=friend
context=outside
secret=1234
host=dynamic

[2003]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2004]
type=friend
contex=outside
secret=1234
host=dynamic

[2005]
type=friend
contex=outside
secret=1234
host=dynamic
 
[2006]
type=friend
contex=internal
secret=1234
host=dynamic

[2007]
type=friend
contex=internal
secret=1234
host=dynamic
[2008]
type=friend
contex=internal
secret=1234
host=dynamic

[2009]
type=friend
contex=internal
secret=1234
host=dynamic

[2010]
type=friend
contex=internal
secret=1234
host=dynamic

############################################################################################
vi /etc/asterisk/extensions.conf
[from-zaptel]
exten => s,1,wait(2)
exten => s,n,dial(sip/2000)
exten => s,n,dial(sip/2001)
exten => s,n,Playback(tt-weasels)
[others]
include => internal
include => outside
[inside]
exten => _20XX,1,Dial(SIP/${EXTEN})
exten => _20XX,n,VoiceMail(${EXTEN}@others,u)
exten => _20XX,n,Hangup()
[outside]
exten => 2001,1,Dial(Zap/1-1/${EXTEN})
exten => 2001,n,Hangup
exten => 2002,1,Dial(Zap/1-1/${EXTEN})
exten => 2002,n,Hangup
exten => 2003,1,Dial(Zap/1-1/${EXTEN})
exten => 2003,n,Hangup
exten => 2004,1,Dial(Zap/1-1/${EXTEN})
exten => 2004,n,Hangup
exten => 2005,1,Dial(Zap/1-1/${EXTEN})
exten => 2005,n,Hangup

this is the log when i am calling from exten 2000 to outside

Connected to Asterisk 1.4.29 currently running on localhost (pid = 2243)
Verbosity is at least 3
[Feb 13 12:05:47] NOTICE[2482]: chan_sip.c:15124 handle_request_invite: Call from '2002' to extension '919369613616' rejected because extension not found.
 
 
any help n support will be highly appreciated
--- On Sat, 13/2/10, Loris Santamaria <loris at lgs.com.ve> wrote:


From: Loris Santamaria <loris at lgs.com.ve>
Subject: [asterisk-users] Call Pickup with 1.6.2.1 and Snom
To: asterisk-users at lists.digium.com
Date: Saturday, 13 February, 2010, 8:39 AM


Hi,

I've used various patches with asterisk 1.4 to have support for call
pickup and notification with good results.

Now I'm trying vanilla 1.6.2 with its official support for "dialog-info
+xml" notifications with no success. This is what i'm doing:

- Phone A has a key configured as type "extension" pointing to Phone B.
- In sip.conf I added notifycid=ignore-context
- Phone A and B and C are in the same callgroup and pickupgroup
- Phone A and B and C are in the same context

Phone C calls Phone B and asterisk generates a notification for phone A:

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="11" state="full" entity="sip:35505 at 10.40.23.179">
<dialog id="35505" call-id="pickup-3c26701519b8-5xxapzoav2u4" direction="recipient">
<remote>
<identity display="Lab 1">sip:35501 at 10.40.23.179</identity>
<target uri="sip:35501 at 10.40.23.179"/>
</remote>
<local>
<identity>sip:35505 at 10.40.23.179</identity>
<target uri="sip:35505 at 10.40.23.179"/>
</local>
<state>early</state>
</dialog>
</dialog-info>

With this notification, Phone A shows on the screen that Phone C is
calling Phone B, and the function key blinks. If one presses the
blinking function key, the phone generates an Invite with replaces, to
try to pickup the call:

INVITE sip:35501 at 10.40.23.179 SIP/2.0
Via: SIP/2.0/UDP 10.40.24.175:5060;branch=z9hG4bK-qoz3zjhmyfcw;rport
From: "Lab 4" <sip:35504 at 10.40.23.179>;tag=o28fq65rfu
To: "Lab 1" <sip:35501 at 10.40.23.179>
Call-ID: 3c2672b3f35a-dpd0zv11yegl
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:35504 at 10.40.24.175:5060>;flow-id=1
Replaces: pickup-3c26701519b8-5xxapzoav2u4
P-Key-Flags: keys="3"
User-Agent: snom320/7.1.39
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 368

Then asterisk receives the pickup request:

[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag: <no from
tag> Totag: <no to tag>
[Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35505 at RedEdelca
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no NAT)
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method INVITE - callid 3c2672b3f35a-dpd0zv11yegl
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis request - 3c2672b3f35a-dpd0zv11yegl
[...]
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: INVITE part of call transfer. Replaces [pickup-3c26701519b8-5xxapzoav2u4]
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Invite/replaces: Will use Replace-Call-ID : pickup-3c26701519b8-5xxapzoav2u4 Fromtag: <no from
tag> Totag: <no to tag>
[Feb 11 10:44:13] NOTICE[4659] chan_sip.c: Trying to pick up 35505 at RedEdelca
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Sending to 10.40.24.175 : 5060 (no NAT)
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Initializing initreq for method INVITE - callid 3c2672b3f35a-dpd0zv11yegl
[Feb 11 10:44:13] VERBOSE[4659] chan_sip.c: Using INVITE request as basis request - 3c2672b3f35a-dpd0zv11yegl
[...]
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: About to call Pickup(35505 at PICKUPMARK)
[Feb 11 10:44:13] DEBUG[4649] devicestate.c: Changing state for SIP/35504 - state 2 (In use)
[Feb 11 10:44:13] DEBUG[4649] devicestate.c: device 'SIP/35504' state '2'
[Feb 11 10:44:13] NOTICE[4659] app_directed_pickup.c: No target channel found for 35505.
[Feb 11 10:44:13] DEBUG[4659] channel.c: Hanging up channel 'SIP/35504-0000000f'
[Feb 11 10:44:13] DEBUG[4659] chan_sip.c: Hangup call SIP/35504-0000000f, SIP callid 3c2672b3f35a-dpd0zv11yegl

After this obviously phone A hasn't picked up the call, and Phone B
keeps ringing.

Did I miss something in the dialplan or is it a bug?

-- 
Loris Santamaria   linux user #70506   xmpp:loris at lgs.com.ve
Links Global Services, C.A.            http://www.lgs.com.ve
Tel: 0286 952.06.87  Cel: 0414 095.00.10  sip:103 at lgs.com.ve
------------------------------------------------------------
-O9 -omg-optimize -fomit-instructions



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