Hi, first of all, sorry for this long thread... I have changed my extensions.conf like you suggested and delete the line with subscribecontext=notify. But unfortunately i still don't see subscribe request in the sip debug trace.
SIP Debugging enabled kingcoco*CLI> <-- SIP read from 192.168.204.5:6108: --- (0 headers 0 lines) Nat keepalive --- kingcoco*CLI> <-- SIP read from 192.168.204.100:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 70 Content-Length: 307 Via: SIP/2.0/UDP 192.168.204.100:5060;branch=z9hG4bK4c6bfc983 Call-ID: 7e6c264483fd010 From: OptiPoint410std <sip:[EMAIL PROTECTED]>;tag=c2a05e95916bbfa;epid=SC22390b To: sip:[EMAIL PROTECTED] CSeq: 1 INVITE Supported: timer Min-SE: 90 Supported: 100rel Allow-Events: talk, hold, conference Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,UPDATE Content-Type: application/sdp Contact: OptiPoint410std <sip:[EMAIL PROTECTED]:5060;transport=udp> Supported: replaces User-Agent: optiPoint 410_420/v4 4.1.66 v=0 o=MxSIP 0 1595508908 IN IP4 192.168.204.100 s=SIP Call c=IN IP4 192.168.204.100 t=0 0 m=audio 5004 RTP/AVP 9 8 0 18 4 101 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 --- (17 headers 14 lines)--- Using INVITE request as basis request - 7e6c264483fd010 Sending to 192.168.204.100 : 5060 (non-NAT) Found user '2001' Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 192.168.204.100:5004 Found description format G722 Found description format PCMA Found description format PCMU Found description format G729 Found description format G723 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 2002 in local (domain 192.168.204.223) list_route: hop: <sip:[EMAIL PROTECTED]:5060;transport=udp> Transmitting (no NAT) to 192.168.204.100:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.204.100:5060;branch=z9hG4bK4c6bfc983;received=192.168.204.100 From: OptiPoint410std <sip:[EMAIL PROTECTED]>;tag=c2a05e95916bbfa;epid=SC22390b To: sip:[EMAIL PROTECTED] Call-ID: 7e6c264483fd010 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- -- Executing Dial("SIP/2001-65fe", "SIP/2002|10|tr") in new stack We're at 192.168.204.223 port 10830 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 192.168.204.5:6108: INVITE sip:[EMAIL PROTECTED]:6108 SIP/2.0 Via: SIP/2.0/UDP 192.168.204.223:5060;branch=z9hG4bK153d90a3;rport From: "OptiPoint410std" <sip:[EMAIL PROTECTED]>;tag=as29a3f9ee To: <sip:[EMAIL PROTECTED]:6108> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 17 May 2006 08:58:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 246 v=0 o=root 24071 24071 IN IP4 192.168.204.223 s=session c=IN IP4 192.168.204.223 t=0 0 m=audio 10830 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 2002 Transmitting (no NAT) to 192.168.204.100:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.204.100:5060;branch=z9hG4bK4c6bfc983;received=192.168.204.100 From: OptiPoint410std <sip:[EMAIL PROTECTED]>;tag=c2a05e95916bbfa;epid=SC22390b To: sip:[EMAIL PROTECTED];tag=as5094780f Call-ID: 7e6c264483fd010 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- kingcoco*CLI> <-- SIP read from 192.168.204.5:6108: SIP/2.0 180 Ringing To: <sip:[EMAIL PROTECTED]:6108>;tag=0a630b27 From: "OptiPoint410std"<sip:[EMAIL PROTECTED]>;tag=as29a3f9ee Via: SIP/2.0/UDP 192.168.204.223:5060;branch=z9hG4bK153d90a3;rport=5060;received=192.168.204.223 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: <sip:[EMAIL PROTECTED]:6108> Content-Length: 0 --- (8 headers 0 lines)--- -- SIP/2002-7bc1 is ringing kingcoco*CLI> <-- SIP read from 192.168.204.5:6108: SIP/2.0 200 OK To: <sip:[EMAIL PROTECTED]:6108>;tag=0a630b27 From: "OptiPoint410std"<sip:[EMAIL PROTECTED]>;tag=as29a3f9ee Via: SIP/2.0/UDP 192.168.204.223:5060;branch=z9hG4bK153d90a3;rport=5060;received=192.168.204.223 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: <sip:[EMAIL PROTECTED]:6108> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Content-Length: 185 v=0 o=- 10603328 10603369 IN IP4 192.168.204.5 s=eyeBeam c=IN IP4 192.168.204.5 t=0 0 m=audio 8702 RTP/AVP 8 0 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (10 headers 9 lines)--- Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.204.5:8702 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: <sip:[EMAIL PROTECTED]:6108> set_destination: Parsing <sip:[EMAIL PROTECTED]:6108> for address/port to send to set_destination: set destination to 192.168.204.5, port 6108 Transmitting (no NAT) to 192.168.204.5:6108: ACK sip:[EMAIL PROTECTED]:6108 SIP/2.0 Via: SIP/2.0/UDP 192.168.204.223:5060;branch=z9hG4bK30ed871d;rport From: "OptiPoint410std" <sip:[EMAIL PROTECTED]>;tag=as29a3f9ee To: <sip:[EMAIL PROTECTED]:6108>;tag=0a630b27 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/2002-7bc1 answered SIP/2001-65fe We're at 192.168.204.223 port 13582 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.204.100:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.204.100:5060;branch=z9hG4bK4c6bfc983;received=192.168.204.100 From: OptiPoint410std <sip:[EMAIL PROTECTED]>;tag=c2a05e95916bbfa;epid=SC22390b To: sip:[EMAIL PROTECTED];tag=as5094780f Call-ID: 7e6c264483fd010 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 246 v=0 o=root 24071 24071 IN IP4 192.168.204.223 s=session c=IN IP4 192.168.204.223 t=0 0 m=audio 13582 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/2001-65fe and SIP/2002-7bc1 kingcoco*CLI> <-- SIP read from 192.168.204.100:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Max-Forwards: 70 Content-Length: 0 Via: SIP/2.0/UDP 192.168.204.100:5060;branch=z9hG4bK43676e2ac Call-ID: 7e6c264483fd010 From: OptiPoint410std <sip:[EMAIL PROTECTED]>;tag=c2a05e95916bbfa;epid=SC22390b To: sip:[EMAIL PROTECTED];tag=as5094780f CSeq: 1 ACK User-Agent: optiPoint 410_420/v4 4.1.66 --- (9 headers 0 lines)--- kingcoco*CLI> <-- SIP read from 192.168.204.5:6108: --- (0 headers 0 lines) Nat keepalive --- kingcoco*CLI> <-- SIP read from 192.168.204.5:6108: BYE sip:[EMAIL PROTECTED] SIP/2.0 To: "OptiPoint410std"<sip:[EMAIL PROTECTED]>;tag=as29a3f9ee From: <sip:[EMAIL PROTECTED]:6108>;tag=0a630b27 Via: SIP/2.0/UDP 192.168.204.5:6108;branch=z9hG4bK-d87543-421140965-1--d87543-;rport Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE Contact: <sip:[EMAIL PROTECTED]:6108> Max-Forwards: 70 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 0 --- (10 headers 0 lines)--- Sending to 192.168.204.5 : 6108 (NAT) Transmitting (NAT) to 192.168.204.5:6108: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.204.5:6108;branch=z9hG4bK-d87543-421140965-1--d87543-;received=192.168.204.5;rport=6108 From: <sip:[EMAIL PROTECTED]:6108>;tag=0a630b27 To: "OptiPoint410std"<sip:[EMAIL PROTECTED]>;tag=as29a3f9ee Call-ID: [EMAIL PROTECTED] CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- == Spawn extension (local, 2002, 1) exited non-zero on 'SIP/2001-65fe' set_destination: Parsing <sip:[EMAIL PROTECTED]:5060;transport=udp> for address/port to send to set_destination: set destination to 192.168.204.100, port 5060 Reliably Transmitting (no NAT) to 192.168.204.100:5060: BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.204.223:5060;branch=z9hG4bK2010e979;rport From: sip:[EMAIL PROTECTED];tag=as5094780f To: OptiPoint410std <sip:[EMAIL PROTECTED]>;tag=c2a05e95916bbfa;epid=SC22390b Contact: <sip:[EMAIL PROTECTED]> Call-ID: 7e6c264483fd010 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- kingcoco*CLI> <-- SIP read from 192.168.204.100:5060: SIP/2.0 200 OK Call-ID: 7e6c264483fd010 CSeq: 102 BYE From: sip:[EMAIL PROTECTED];tag=as5094780f To: OptiPoint410std <sip:[EMAIL PROTECTED]>;tag=c2a05e95916bbfa;epid=SC22390b Via: SIP/2.0/UDP 192.168.204.223:5060;branch=z9hG4bK2010e979;rport Content-Length: 0 Supported: replaces User-Agent: optiPoint 410_420/v4 4.1.66 --- (9 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' Destroying call '7e6c264483fd010' kingcoco*CLI> sip no deb <-- SIP read from 192.168.204.5:6108: --- (0 headers 0 lines) Nat keepalive --- > Try this: > > [local] > exten => 2001,1,Dial(SIP/2001,10,tr) > exten => 2001,hint,SIP/2001 > exten => 2002,1,Dial(SIP/2002,10,tr) > exten => 2002,hint,SIP/2002 > > cYa, > Avi __________________________________________________ Do You Yahoo!? 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