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:[email protected]"> <dialog id="35505" call-id="pickup-3c26701519b8-5xxapzoav2u4" direction="recipient"> <remote> <identity display="Lab 1">sip:[email protected]</identity> <target uri="sip:[email protected]"/> </remote> <local> <identity>sip:[email protected]</identity> <target uri="sip:[email protected]"/> </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:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.40.24.175:5060;branch=z9hG4bK-qoz3zjhmyfcw;rport From: "Lab 4" <sip:[email protected]>;tag=o28fq65rfu To: "Lab 1" <sip:[email protected]> Call-ID: 3c2672b3f35a-dpd0zv11yegl CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:[email protected]: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 35...@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 35...@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(35...@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:[email protected] Links Global Services, C.A. http://www.lgs.com.ve Tel: 0286 952.06.87 Cel: 0414 095.00.10 sip:[email protected] ------------------------------------------------------------ -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
