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
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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
???http://lists.digium.com/mailman/listinfo/asterisk-users
Your Mail works best with the New Yahoo Optimized IE8. Get it NOW!
http://downloads.yahoo.com/in/internetexplorer/
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20100213/c7231b4b/attachment.htm
Hi, I've experienced the same thing in the 1.6.2 release, with the 1.6.1 all work as expected. There is nothing in the changelog ... So, I think it's a bug ? -----Message d'origine----- De?: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] De la part de Loris Santamaria Envoy??: samedi 13 f?vrier 2010 04:09 ??: asterisk-users at lists.digium.com Objet?: [asterisk-users] Call Pickup with 1.6.2.1 and Snom 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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users