I don't see anything apparent in for the SIP phones that would indicate that *8 is anything but VoiceMailMain(). I looked in extensions.conf for it, as well as sip.conf (the config that I pasted in a previous e-mail. It does indeed appear that a literal *8 is being passed to asterisk.
Here's the relevant context of extensions.conf: [internal] ;downstairs office exten => 7191,1,Dial(Zap/1) ;cordless exten => 7192,1,Dial(Zap/2) ;second cordless exten => 7193,1,Dial(SIP/ht3861) ;call a couple phones exten => 7194,1,Dial(SIP/gxp2&SIP/user) ;goto voicemail exten => *8,1,VoiceMailMain() include => local include => ld The sip.conf bits are contained in my initial post. Here's the debug output from the console, it's somewhat long. Could the key line be (towards the bottom) this? [Jan 28 20:39:10] NOTICE[31924]: chan_sip.c:13519 handle_request_invite: Nothing to pick up for OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. Here's the full output: <--- SIP read from 192.168.1.165:5806 ---> INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.165:5806;branch=z9hG4bK-d87543-42396923770b6e49-1--d87543-;rport Max-Forwards: 70 Contact: <sip:[EMAIL PROTECTED]:5806> To: "*8"<sip:[EMAIL PROTECTED]> From: "SIP User"<sip:[EMAIL PROTECTED]>;tag=2a71c861 Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 328 v=0 o=- 6 2 IN IP4 192.168.1.165 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.165 t=0 0 m=audio 35172 RTP/AVP 107 119 0 98 8 3 101 a=alt:1 1 : UGnBtE65 8rk0z5iz 192.168.1.165 35172 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (12 headers 13 lines) --- Sending to 192.168.1.165 : 5806 (NAT) Using INVITE request as basis request - OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. Found user 'user' for 'user' ord*CLI> <--- Reliably Transmitting (no NAT) to 192.168.1.165:5806 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.165:5806;branch=z9hG4bK-d87543-42396923770b6e49-1--d87543-;received=192.168.1.165;rport=5806 From: "SIP User"<sip:[EMAIL PROTECTED]>;tag=2a71c861 To: "*8"<sip:[EMAIL PROTECTED]>;tag=as31c6aff1 Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="69d017e6" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.' in 32000 ms (Method: INVITE) ord*CLI> <--- SIP read from 192.168.1.165:5806 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.165:5806;branch=z9hG4bK-d87543-42396923770b6e49-1--d87543-;rport To: "*8"<sip:[EMAIL PROTECTED]>;tag=as31c6aff1 From: "SIP User"<sip:[EMAIL PROTECTED]>;tag=2a71c861 Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. CSeq: 1 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- ord*CLI> <--- SIP read from 192.168.1.165:5806 ---> INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.165:5806;branch=z9hG4bK-d87543-dc51447ec8163e0a-1--d87543-;rport Max-Forwards: 70 Contact: <sip:[EMAIL PROTECTED]:5806> To: "*8"<sip:[EMAIL PROTECTED]> From: "SIP User"<sip:[EMAIL PROTECTED]>;tag=2a71c861 Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1006e stamp 34025 Authorization: Digest username="user",realm="asterisk",nonce="69d017e6",uri="sip:[EMAIL PROTECTED]",response="b82d4700cc1f72ef2711df0b597e7184",algorithm=MD5 Content-Length: 328 v=0 o=- 6 2 IN IP4 192.168.1.165 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.165 t=0 0 m=audio 35172 RTP/AVP 107 119 0 98 8 3 101 a=alt:1 1 : UGnBtE65 8rk0z5iz 192.168.1.165 35172 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (13 headers 13 lines) --- Sending to 192.168.1.165 : 5806 (NAT) Using INVITE request as basis request - OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. Found user 'user' for 'user' Found RTP audio format 107 Found RTP audio format 119 Found RTP audio format 0 Found RTP audio format 98 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.165:35172 Found description format BV32 for ID 107 Found description format BV32-FEC for ID 119 Found description format iLBC for ID 98 Found description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.165:35172 Looking for *8 in internal (domain 192.168.1.2) list_route: hop: <sip:[EMAIL PROTECTED]:5806> ord*CLI> <--- Transmitting (no NAT) to 192.168.1.165:5806 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.165:5806;branch=z9hG4bK-d87543-dc51447ec8163e0a-1--d87543-;received=192.168.1.165;rport=5806 From: "SIP User"<sip:[EMAIL PROTECTED]>;tag=2a71c861 To: "*8"<sip:[EMAIL PROTECTED]> Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 <------------> [Jan 28 20:39:10] NOTICE[31924]: chan_sip.c:13519 handle_request_invite: Nothing to pick up for OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. ord*CLI> <--- Reliably Transmitting (no NAT) to 192.168.1.165:5806 ---> SIP/2.0 503 Unavailable Via: SIP/2.0/UDP 192.168.1.165:5806;branch=z9hG4bK-d87543-dc51447ec8163e0a-1--d87543-;received=192.168.1.165;rport=5806 From: "SIP User"<sip:[EMAIL PROTECTED]>;tag=2a71c861 To: "*8"<sip:[EMAIL PROTECTED]>;tag=as235f0837 Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 <------------> ord*CLI> <--- SIP read from 192.168.1.165:5806 ---> ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.165:5806;branch=z9hG4bK-d87543-dc51447ec8163e0a-1--d87543-;rport To: "*8"<sip:[EMAIL PROTECTED]>;tag=as235f0837 From: "SIP User"<sip:[EMAIL PROTECTED]>;tag=2a71c861 Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI. CSeq: 2 ACK Content-Length: 0 <-------------> --- (7 headers 0 lines) --- Really destroying SIP dialog 'OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.' Method: ACK On Mon, Jan 29, 2007 at 09:59:14AM +0800, Leo Ann Boon wrote: > check that your phone is not using *8 in its own dial plan. Also, do a > sip debug and see that the phone is actually sending *8 to asterisk. > > Leo > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
