Hello, I tried to remotely anwer physical phones, which are in RINGING state, via event-socket. But the phones are not answering (tried Yealink an Aastra, see below). I send via event-socket sendevent NOTIFY. profile: internal. event-string: talk. user: 200. host: my.domain.com. content-type: application/simple-message-summary. State: ACTIVE I also set the "State" as described in http://wiki.freeswitch.org/wiki/Mod_event_socket#sendevent The following SIP is exchanged then (here Yealink): NOTIFY sip:200@192.168.1.129:5062 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.220;rport;branch=z9hG4bKy8arj9a0SXNZN. Max-Forwards: 70. From: <sip:200@my.domain.com>;tag=515t5XNrSKFQF. To: <sip:200@my.domain.com>. Call-ID: c74eacdc-cca4-1231-c19a-b3a15c58d7e1. CSeq: 52146637 NOTIFY. Contact: <sip:mod_sofia@192.168.1.220:5060>. User-Agent: Freeswitch. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Event: talk. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=noresource. Content-Type: application/simple-message-summary. Content-Length: 0. SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.220;rport;branch=z9hG4bKy8arj9a0SXNZN. From: <sip:200@my.domain.com>;tag=515t5XNrSKFQF. To: <sip:200@my.domain.com>;tag=669161475. Call-ID: c74eacdc-cca4-1231-c19a-b3a15c58d7e1. CSeq: 52146637 NOTIFY. User-Agent: Yealink SIP-T38G 38.70.0.125. Content-Length: 0. The call was not answered however and the Rsyslog does not show an entry to this message. I then tried an Aastra 6739i Phone and got the following: NOTIFY sip:208@192.168.1.118:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.1.220;rport;branch=z9hG4bKpeya6gXmHa3rr. Max-Forwards: 70. From: <sip:208@my.domain.com>;tag=SXXjXe2S7De0K. To: <sip:208@my.domain.com>. Call-ID: 835658b2-cca1-1231-c19a-b3a15c58d7e1. CSeq: 52145935 NOTIFY. Contact: <sip:mod_sofia@192.168.1.220:5060>. User-Agent: Freeswitch. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Event: talk. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=noresource. Content-Type: application/simple-message-summary. Content-Length: 0. . SIP/2.0 489 Bad Event. Via: SIP/2.0/UDP 192.168.1.220;rport=5060;branch=z9hG4bKpeya6gXmHa3rr;received=192.168.1.220. From: <sip:208@my.domain.com>;tag=SXXjXe2S7De0K. To: <sip:208@my.domain.com>;tag=891443321. Call-ID: 835658b2-cca1-1231-c19a-b3a15c58d7e1. CSeq: 52145935 NOTIFY. Server: Aastra 6739i/3.2.2.2044. Content-Length: 0. Aastra shows me in rsyslog: 00:53:57.550000 : (SIP) INFO: EvOnPacketReceived: CSipEngine::, Packet do not match an existing Dialog, method: 6 00:53:57.550000 : (SIP) FUNC: OnPacketReceived: CSipGenericRequest(0x821DBC60) 00:53:57.550000 : (SIP) FUNC: EvInvalidNotify: CSipSubscribeEventDispatcher(0x821DC770) Reason 8002c405: Enter So I think, the Subscription-State may not be right, so Subscription-State: active might be better, right? mod_sofia.c, around line 1429 (as of 23-Oct-2013) should set this to "active" as far as I can see, but this is not the case here. But how can I solve this? Maybe there is another way? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de