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 : [email protected] [mailto:[email protected]] De la part de Loris Santamaria Envoyé : samedi 13 février 2010 04:09 À : [email protected] 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:[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 -- _____________________________________________________________________ -- 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
