sip.conf ; AstBill-0.9.0.8 ; Created: Mon Oct 24 16:47:33 BST 2005 ; sip_custom.conf ; This is where you put your own STATIC custom SIP configuration ; ; Remember Asterisk commands: reload or extensions reload [70555] type=peer username=70555 fromuser=70555 context=default secret=65703 host=83.143.155.28 ;callerid=Barbara <70125> ;context=default dtmfmode=rfc2833 ;mailbox=70125 [linksys] context=office type=friend username=111 secret=111 host=192.168.5.24 dtmfmode=rfc2833 ;disallow=all ;allow=alaw qualify=yes [222] type=friend username=222 secret=222 callerid=222 host=dynamic context=office dtmfmode=rfc2833 ;disallow=all ;allow=alaw qualify=yes extensions.conf [general] static=yes writeprotect=yes [globals] ; Any category other than "General" and "Globals" represent ; extension contexts, which are collections of extensions. ; ; Extension names may be numbers, letters, or combinations ; thereof. If an extension name is prefixed by a '_' ; character, it is interpreted as a pattern rather than a ; literal. In patterns, some characters have special meanings: ; ; X - any digit from 0-9 ; Z - any digit from 1-9 ; N - any digit from 2-9 ; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9) ; . - wildcard, matches anything remaining (e.g. _9011. matches ; anything starting with 9011 excluding 9011 itself) ; ; For example the extension _NXXXXXX would match normal 7 digit dialings, ; while _1NXXNXXXXXX would represent an area code plus phone number ; preceeded by a one. ; ; Each step of an extension is ordered by priority, which must ; always start with 1 to be considered a valid extension. ; ; Contexts contain several lines, one for each step of each ; extension, which can take one of two forms as listed below, ; with the first form being preferred. One may include another ; context in the current one as well, optionally with a ; date and time. Included contexts are included in the order ; they are listed. ; [default] exten => #,1,Hangup ; Hang them up. ; A timeout and "invalid extension rule" exten => t,1,Goto(i,1) ; If they take too long, give up exten => i,1,Playback(invalid) ; "That's not valid, try again" exten => i,2,Hangup ; Hang them up. exten => h,1,DeadAGI(agistardead.agi) exten => h,1,Hangup exten => 198,1,Ringing exten => 198,2,Wait(2) exten => 198,3,VoicemailMain exten => 199,1,Ringing exten => 199,2,Wait(2) exten => 199,3,AGI(agistarvoicemail.agi) exten => 199,4,VoicemailMain(s${VOICEMAIL}) exten => 500,1,Ringing exten => 500,2,Wait(2) exten => 500,3,DateTime exten => 500,4,Hangup ; Hang them up. *CLI> sip debug SIP Debugging enabled The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. *CLI> <--- SIP read from 192.168.5.1:5060 ---> INVITE sip:421818@192.168.5.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK37537693b718f1e1 Max-Forwards: 70 To: From: ;tag=xIzN3EDMxID Call-ID: 243D1B916121212@192.168.5.1 CSeq: 1 INVITE Contact: Session-Expires: 180;refresher=uac Supported: timer Content-Type: application/sdp Content-Length: 234 v=0 o=222 0 0 IN IP4 192.168.5.1 s=- c=IN IP4 192.168.5.1 t=0 0 m=audio 50364 RTP/AVP 0 18 100 a=rtpmap:0 PCMU/8000 a=sendrecv a=rtpmap:18 G729/8000 a=sendrecv a=ptime:20 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 <-------------> --- (12 headers 13 lines) --- Sending to 192.168.5.1 : 5060 (no NAT) Using INVITE request as basis request - 243D1B916121212@192.168.5.1 <--- Reliably Transmitting (no NAT) to 192.168.5.1:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK37537693b718f1e1;received=192.168.5.1 From: ;tag=xIzN3EDMxID To: ;tag=as2b2032ac Call-ID: 243D1B916121212@192.168.5.1 CSeq: 1 INVITE User-Agent: Noname Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="017ee81a" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '243D1B916121212@192.168.5.1' in 32000 ms (Method: INVITE) Found user '222' <--- SIP read from 192.168.5.1:5060 ---> ACK sip:421818@192.168.5.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK37537693b718f1e1 Max-Forwards: 70 To: ;tag=as2b2032ac From: ;tag=xIzN3EDMxID Call-ID: 243D1B916121212@192.168.5.1 CSeq: 1 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 192.168.5.1:5060 ---> INVITE sip:421818@192.168.5.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bKe00036c520796f26 Max-Forwards: 70 To: From: ;tag=xIzN3EDMxID Call-ID: 243D1B916121212@192.168.5.1 CSeq: 2 INVITE Contact: Session-Expires: 180;refresher=uac Supported: timer Proxy-Authorization: Digest username="222", realm="asterisk", nonce="017ee81a", uri="sip:421818@192.168.5.100", response="b33f1a33af044df4210111e68b360d53", algorithm=MD5, nc=0000001 Content-Type: application/sdp Content-Length: 234 v=0 o=222 0 0 IN IP4 192.168.5.1 s=- c=IN IP4 192.168.5.1 t=0 0 m=audio 50364 RTP/AVP 0 18 100 a=rtpmap:0 PCMU/8000 a=sendrecv a=rtpmap:18 G729/8000 a=sendrecv a=ptime:20 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 <-------------> --- (13 headers 13 lines) --- Sending to 192.168.5.1 : 5060 (no NAT) Using INVITE request as basis request - 243D1B916121212@192.168.5.1 Found user '222' Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 100 Peer audio RTP is at port 192.168.5.1:50364 Found description format PCMU for ID 0 Found description format G729 for ID 18 Found description format telephone-event for ID 100 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.5.1:50364 Looking for 421818 in office (domain 192.168.5.100) list_route: hop: <--- Transmitting (no NAT) to 192.168.5.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bKe00036c520796f26;received=192.168.5.1 From: ;tag=xIzN3EDMxID To: Call-ID: 243D1B916121212@192.168.5.1 CSeq: 2 INVITE User-Agent: Noname Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [421818@office:1] Dial("SIP/222-081fa5b0", "SIP/421818@linksys|20") in new stack Audio is at 192.168.5.100 port 10554 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.5.24:5060: INVITE sip:421818@192.168.5.24 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK1c3c210b;rport From: "222" ;tag=as0387a7e6 To: Contact: Call-ID: 68a154c6376ccb1126896235676c7712@192.168.5.100 CSeq: 102 INVITE User-Agent: Noname Max-Forwards: 70 Date: Wed, 03 Oct 2007 08:53:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 289 v=0 o=root 17570 17570 IN IP4 192.168.5.100 s=session c=IN IP4 192.168.5.100 t=0 0 m=audio 10554 RTP/AVP 0 3 8 100 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 421818@linksys <--- SIP read from 192.168.5.24:5060 ---> SIP/2.0 404 Not Found To: ;tag=be972e5e3c29a9a0i0 From: "222" ;tag=as0387a7e6 Call-ID: 68a154c6376ccb1126896235676c7712@192.168.5.100 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK1c3c210b Server: Linksys/SPA3102-3.3.6(GW) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (no NAT) to 192.168.5.24:5060: ACK sip:421818@192.168.5.24 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK1c3c210b;rport From: "222" ;tag=as0387a7e6 To: ;tag=be972e5e3c29a9a0i0 Contact: Call-ID: 68a154c6376ccb1126896235676c7712@192.168.5.100 CSeq: 102 ACK User-Agent: Noname Max-Forwards: 70 Content-Length: 0 --- -- SIP/linksys-08201780 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [421818@office:2] Congestion("SIP/222-081fa5b0", "") in new stack <--- Transmitting (no NAT) to 192.168.5.1:5060 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bKe00036c520796f26;received=192.168.5.1 From: ;tag=xIzN3EDMxID To: ;tag=as4b788a4f Call-ID: 243D1B916121212@192.168.5.1 CSeq: 2 INVITE User-Agent: Noname Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 X-Asterisk-HangupCause: Unallocated (unassigned) number X-Asterisk-HangupCauseCode: 1 <------------> == Spawn extension (office, 421818, 2) exited non-zero on 'SIP/222-081fa5b0' <--- SIP read from 192.168.5.1:5060 ---> ACK sip:421818@192.168.5.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bKe00036c520796f26 Max-Forwards: 70 To: ;tag=as4b788a4f From: ;tag=xIzN3EDMxID Call-ID: 243D1B916121212@192.168.5.1 CSeq: 2 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '68a154c6376ccb1126896235676c7712@192.168.5.100' Method: INVITE Really destroying SIP dialog '243D1B916121212@192.168.5.1' Method: ACK Reliably Transmitting (no NAT) to 192.168.5.24:5060: OPTIONS sip:192.168.5.24 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK307c546f;rport From: "asterisk" ;tag=as09b8ae40 To: Contact: Call-ID: 19a7fdd35da6d38b3b75d7c0683fec6e@192.168.5.100 CSeq: 102 OPTIONS User-Agent: Noname Max-Forwards: 70 Date: Wed, 03 Oct 2007 08:53:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Reliably Transmitting (no NAT) to 192.168.5.1:5060: OPTIONS sip:222@192.168.5.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK5a09b74f;rport From: "asterisk" ;tag=as504711e1 To: Contact: Call-ID: 7ae227663b122c0d19488512369c7ee8@192.168.5.100 CSeq: 102 OPTIONS User-Agent: Noname Max-Forwards: 70 Date: Wed, 03 Oct 2007 08:53:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 192.168.5.24:5060 ---> SIP/2.0 404 Not Found To: ;tag=be972e5e3c29a9a0i0 From: "asterisk" ;tag=as09b8ae40 Call-ID: 19a7fdd35da6d38b3b75d7c0683fec6e@192.168.5.100 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK307c546f Server: Linksys/SPA3102-3.3.6(GW) Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Really destroying SIP dialog '19a7fdd35da6d38b3b75d7c0683fec6e@192.168.5.100' Method: OPTIONS <--- SIP read from 192.168.5.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.5.100:5060;branch=z9hG4bK5a09b74f To: From: "asterisk" ;tag=as504711e1 Call-ID: 7ae227663b122c0d19488512369c7ee8@192.168.5.100 CSeq: 102 OPTIONS Supported: timer,100rel Allow: INVITE,ACK,OPTIONS,BYE,CANCEL Contact: Content-Length: 0 Accept-Encoding: identity Accept-Language: en <-------------> --- (12 headers 0 lines) --- Really destroying SIP dialog '7ae227663b122c0d19488512369c7ee8@192.168.5.100' Method: OPTIONS