donkey*CLI> sip debug SIP Debugging enabled donkey*CLI> <-- SIP read from 209.183.xxx.xxx:5060: INVITE sip:4792425555@72.18.xxx.xxx:5060;transport=udp SIP/2.0 To: From: "Last name, First Name";tag=0cccea58 Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-57baf57b2a1a8b7606c7-1-cHAxMTg1YTU0MDFlMDE2OTRjODk1NQ..-d87543- Call-ID: 2a184c01f56b980ade15a75fb21ef295 CSeq: 315707884 INVITE Contact: "Last name, First Name" Expires: 180 Max-Forwards: 69 Content-Type: application/sdp Organization: Supported: 100rel Allow-Events: message-summary, refer, dialog, line-seize, presence Content-Length: 169 v=0 o=- 931650190 903309146 IN IP4 209.183.xxx.xxx s=- c=IN IP4 209.183.xxx.xxx t=0 0 m=audio 25496 RTP/AVP 2 0 101 a=ptime:20 a=rtpmap:101 telephone-event/8000 --- (14 headers 8 lines)--- Using INVITE request as basis request - 2a184c01f56b980ade15a75fb21ef295 Sending to 209.183.xxx.xxx : 5060 (non-NAT) Found peer 'myprovidertrunk' Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 209.183.xxx.xxx:25496 Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x14 (ulaw|g726)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 4792425555 in keysystem (domain 72.18.xxx.xxx) Reliably Transmitting (NAT) to 209.183.xxx.xxx:5060: SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-57baf57b2a1a8b7606c7-1-cHAxMTg1YTU0MDFlMDE2OTRjODk1NQ..-d87543-;received=209.183.xxx.xxx From: "Last name, First Name";tag=0cccea58 To: ;tag=as088d7c9c Call-ID: 2a184c01f56b980ade15a75fb21ef295 CSeq: 315707884 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- donkey*CLI> <-- SIP read from 209.183.xxx.xxx:5060: ACK sip:4792425555@72.18.xxx.xxx:5060;transport=udp SIP/2.0 To: ;tag=as088d7c9c From: "Last Name,First Name";tag=0cccea58 Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-57baf57b2a1a8b7606c7-1-cHAxMTg1YTU0MDFlMDE2OTRjODk1NQ..-d87543- Call-ID: 2a184c01f56b980ade15a75fb21ef295 CSeq: 315707884 ACK Content-Length: 0 --- (7 headers 0 lines)--- Destroying call '2a184c01f56b980ade15a75fb21ef295' donkey*CLI> <-- SIP read from 209.183.xxx.xxx:5060: INVITE sip:4792425555@72.18.xxx.xxx:5060;transport=udp SIP/2.0 To: From: "Last name, First Name";tag=83261338 Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9h;G4bK-d87543-3a7e9c743b67b052eb8e-1-cHA3YWRiMGM3NjgxY2VhYzBlMzk5Yw..-d87543- Call-ID: 6a2f8f2ef57d30cc18945606cc736a68 CSeq: 249514123 INVITE Contact: "Last name, First Name" Expires: 180 Max-Forwards: 69 Content-Type: application/sdp Organization: Supported: 100rel Allow-Events: message-summary, refer, dialog, line-seize, presence Content-Length: 169 v=0 o=- 931651729 903310705 IN IP4 209.183.xxx.xxx s=- c=IN IP4 209.183.xxx.xxx t=0 0 m=audio 29562 RTP/AVP 2 0 101 a=ptime:20 a=rtpmap:101 telephone-event/8000 --- (14 headers 8 lines)--- Using INVITE request as basis request - 6a2f8f2ef57d30cc18945606cc736a68 Sending to 209.183.xxx.xxx : 5060 (non-NAT) Found peer 'myprovidertrunk' Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 209.183.xxx.xxx:29562 Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x14 (ulaw|g726)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 4792425555 in keysystem (domain 72.18.xxx.xxx) Reliably Transmitting (NAT) to 209.183.xxx.xxx:5060: SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-3a7e9c743b67b052eb8e-1-cHA3YWRiMGM3NjgxY2VhYzBlMzk5Yw..-d87543-;received=209.183.xxx.xxx From: "Last name, First Name";tag=83261338 To: ;tag=as77270ab7 Call-ID: 6a2f8f2ef57d30cc18945606cc736a68 CSeq: 249514123 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- donkey*CLI> <-- SIP read from 209.183.xxx.xxx:5060: ACK sip:4792425555@72.18.xxx.xxx:5060;transport=udp SIP/2.0 To: ;tag=as77270ab7 From: "Last name, First Name";tag=83261338 Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-3a7e9c743b67b052eb8e-1-cHA3YWRiMGM3NjgxY2VhYzBlMzk5Yw..-d87543- Call-ID: 6a2f8f2ef57d30cc18945606cc736a68 CSeq: 249514123 ACK Content-Length: 0 --- (7 headers 0 lines)--- Destroying call '6a2f8f2ef57d30cc18945606cc736a68' donkey*CLI> <-- SIP read from 209.183.xxx.xxx:5060: INVITE sip:4792425555@72.18.xxx.xxx:5060;transport=udp SIP/2.0 To: From: "Last name, First Name";tag=fa63913c Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-d0118f257ca89f50c227-1-cHAwZTgwMTQ1MmM3YjMxMTVkMzA2Ng..-d87543- Call-ID: 62330166f22dcad1a1d3fca3041222b2 CSeq: 520618893 INVITE Contact: "Last name, First Name" Expires: 180 Max-Forwards: 69 Content-Type: application/sdp Organization: Supported: 100rel Allow-Events: message-summary, refer, dialog, line-seize, presence Content-Length: 169 v=0 o=- 931653070 903312054 IN IP4 209.183.xxx.xxx s=- c=IN IP4 209.183.xxx.xxx t=0 0 m=audio 30136 RTP/AVP 2 0 101 a=ptime:20 a=rtpmap:101 telephone-event/8000 --- (14 headers 8 lines)--- Using INVITE request as basis request - 62330166f22dcad1a1d3fca3041222b2 Sending to 209.183.xxx.xxx : 5060 (non-NAT) Found peer 'myprovidertrunk' Found RTP audio format 2 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 209.183.xxx.xxx:30136 Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x14 (ulaw|g726)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 4792425555 in keysystem (domain 72.18.xxx.xxx) Reliably Transmitting (NAT) to 209.183.xxx.xxx:5060: SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-d0118f257ca89f50c227-1-cHAwZTgwMTQ1MmM3YjMxMTVkMzA2Ng..-d87543-;received=209.183.xxx.xxx From: "Last name, First Name";tag=fa63913c To: ;tag=as350d0bdf Call-ID: 62330166f22dcad1a1d3fca3041222b2 CSeq: 520618893 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- donkey*CLI> <-- SIP read from 209.183.xxx.xxx:5060: ACK sip:4792425555@72.18.xxx.xxx:5060;transport=udp SIP/2.0 To: ;tag=as350d0bdf From: "Last name, First Name";tag=fa63913c Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-d0118f257ca89f50c227-1-cHAwZTgwMTQ1MmM3YjMxMTVkMzA2Ng..-d87543- Call-ID: 62330166f22dcad1a1d3fca3041222b2 CSeq: 520618893 ACK Content-Length: 0 --- (7 headers 0 lines)--- Destroying call '62330166f22dcad1a1d3fca3041222b2' donkey*CLI> <-- SIP read from 209.183.xxx.xxx:5060: OPTIONS sip:s@72.18.xxx.xxx:5060 SIP/2.0 To: From: ;tag=f53b2453 Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-298d9f69ad4aed1d6b57-1--d87543- Call-ID: 6744b26d1516e339047a8a560c27c0076a16013d CSeq: 1001 OPTIONS Max-Forwards: 70 Content-Length: 0 --- (8 headers 0 lines)--- Looking for s in default (domain 72.18.xxx.xxx) Transmitting (no NAT) to 209.183.xxx.xxx:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 209.183.xxx.xxx:5060;branch=z9hG4bK-d87543-298d9f69ad4aed1d6b57-1--d87543-;received=209.183.xxx.xxx From: ;tag=f53b2453 To: ;tag=as7094f8f2 Call-ID: 6744b26d1516e339047a8a560c27c0076a16013d CSeq: 1001 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '6744b26d1516e339047a8a560c27c0076a16013d' donkey*CLI> <-- SIP read from 63.211.239.18:5060: OPTIONS sip:s@72.18.xxx.xxx SIP/2.0 Via: SIP/2.0/UDP 63.211.239.18:5060;branch=z9hG4bK62a43d4c;rport From: "Unknown" ;tag=as270cc96b To: Contact: Call-ID: 65ddf76b0c09eec70b542af84e6789af@63.211.239.18 CSeq: 102 OPTIONS User-Agent: TELIAX GEAR v4.1.0x Max-Forwards: 70 Date: Wed, 20 Feb 2008 15:48:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- (13 headers 0 lines)--- Looking for s in default (domain 72.18.xxx.xxx) Transmitting (no NAT) to 63.211.239.18:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 63.211.239.18:5060;branch=z9hG4bK62a43d4c;rport;received=63.211.239.18 From: "Unknown" ;tag=as270cc96b To: ;tag=as5bb18aba Call-ID: 65ddf76b0c09eec70b542af84e6789af@63.211.239.18 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Destroying call '65ddf76b0c09eec70b542af84e6789af@63.211.239.18' donkey*CLI> sip no debug SIP Debugging Disabled ------------------------------------------------------------------------ ; ; Static extension configuration file, used by ; the pbx_config module. This is where you configure all your ; inbound and outbound calls in Asterisk. ; ; This configuration file is reloaded ; - With the "extensions reload" command in the CLI ; - With the "reload" command (that reloads everything) in the CLI ; ; The "General" category is for certain variables. ; [general] ; ; If static is set to no, or omitted, then the pbx_config will rewrite ; this file when extensions are modified. Remember that all comments ; made in the file will be lost when that happens. ; ; XXX Not yet implemented XXX ; static=yes ; ; if static=yes and writeprotect=no, you can save dialplan by ; CLI command 'save dialplan' too ; writeprotect=no ; ; If autofallthrough is set, then if an extension runs out of ; things to do, it will terminate the call with BUSY, CONGESTION ; or HANGUP depending on Asterisk's best guess (strongly recommended). ; ; If autofallthrough is not set, then if an extension runs out of ; things to do, asterisk will wait for a new extension to be dialed ; (this is the original behavior of Asterisk 1.0 and earlier). ; autofallthrough=yes ; ; If clearglobalvars is set, global variables will be cleared ; and reparsed on an extensions reload, or Asterisk reload. ; ; If clearglobalvars is not set, then global variables will persist ; through reloads, and even if deleted from the extensions.conf or ; one if its included files, will remain set to the previous value. ; clearglobalvars=no ; ; If priorityjumping is set to 'yes', then applications that support ; 'jumping' to a different priority based on the result of their operations ; will do so (this is backwards compatible behavior with pre-1.2 releases ; of Asterisk). Individual applications can also be requested to do this ; by passing a 'j' option in their arguments. ; priorityjumping=no ; ; You can include other config files, use the #include command ; (without the ';'). Note that this is different from the "include" command ; that includes contexts within other contexts. The #include command works ; in all asterisk configuration files. ;#include "filename.conf" ; The "Globals" category contains global variables that can be referenced ; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental ; variables, ; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid ; [globals] CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password ;IAXINFO=myuser:mypass TRUNK=Zap/G1 ; Trunk interface CELLTRUNK=Zap/2 ;FIREBALLIAXTRUNK=IAX2/kirkhamsystems@teliax KIRKHAMIAXTRUNK=IAX2/kirkhamsystems2@teliax2 RINGTIME=25 MAXPSTNCALLS=1 ; ; Note the 'g2' in the TRUNK variable above. It specifies which group (defined ; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use in ; the specified group. The four possible options are: ; ; g: select the lowest-numbered non-busy Zap channel ; (aka. ascending sequential hunt group). ; G: select the highest-numbered non-busy Zap channel ; (aka. descending sequential hunt group). ; r: use a round-robin search, starting at the next highest channel than last ; time (aka. ascending rotary hunt group). ; R: use a round-robin search, starting at the next lowest channel than last ; time (aka. descending rotary hunt group). ; TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) ;TRUNK=IAX2/user:pass@provider ; ; 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) ; ! - wildcard, causes the matching process to complete as soon as ; it can unambiguously determine that no other matches are possible ; ; 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. The priority ; "next" or "n" means the previous priority plus one, regardless of whether ; the previous priority was associated with the current extension or not. ; The priority "same" or "s" means the same as the previously specified ; priority, again regardless of whether the previous entry was for the ; same extension. Priorities may be immediately followed by a plus sign ; and another integer to add that amount (most useful with 's' or 'n'). ; Priorities may then also have an alias, or label, in ; parenthesis after their name which can be used in goto situations ; ; 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. ; ;[context] ;exten => someexten,priority[+offset][(alias)],application(arg1,arg2,...) ;exten => someexten,priority[+offset][(alias)],application,arg1|arg2... ; ; Timing list for includes is ; ;