The extensions.conf file you provided looks suspiciously like the asterisk configs/extensions.conf.sample file.
Did you create a dialplan for your specific configuration or did you just copy the sample file? ----- Original Message ----- From: "Angus Comber" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Sent: Sunday, July 24, 2005 2:50 PM Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202 > I think the 777 may be a bit of a Red Herring. I dialed 777 as a test. I > can't dial 202 from 200 if I actually dial 202! > > My extensions.conf file: > > > ; > ; 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 > > ; 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 > variable > ; ${${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/g2 ; Trunk interface > ; > ; 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:[EMAIL PROTECTED] > > ; > ; 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. > ; > ;[context] > ;exten => someexten,priority,application(arg1,arg2,...) > ;exten => someexten,priority,application,arg1|arg2... > ; > ; Timing list for includes is > ; > ; <time range>|<days of week>|<days of month>|<months> > ; > ;include => daytime|9:00-17:00|mon-fri|*|* > ; > ; ignorepat can be used to instruct drivers to not cancel dialtone upon > ; receipt of a particular pattern. The most commonly used example is > ; of course '9' like this: > ; > ;ignorepat => 9 > ; > ; so that dialtone remains even after dialing a 9. > ; > > ; > ; Here are the entries you need to participate in the IAXTEL > ; call routing system. Most IAXTEL numbers begin with 1-700, but > ; there are exceptions. For more information, and to sign > ; up, please go to www.gnophone.com or www.iaxtel.com > ; > [iaxtel700] > exten => _91700XXXXXXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED]) > > ; > ; The SWITCH statement permits a server to share the dialplain with > ; another server. Use with care: Reciprocal switch statements are not > ; allowed (e.g. both A -> B and B -> A), and the switched server needs > ; to be on-line or else dialing can be severly delayed. > ; > [iaxprovider] > ;switch => IAX2/user:[EMAIL PROTECTED]/mycontext > > [trunkint] > ; > ; International long distance through trunk > ; > exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _9011.,2,Congestion > > [trunkld] > ; > ; Long distance context accessed through trunk > ; > exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _91NXXNXXXXXX,2,Congestion > > [trunklocal] > ; > ; Local seven-digit dialing accessed through trunk interface > ; > exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _9NXXXXXX,2,Congestion > > [trunktollfree] > ; > ; Long distance context accessed through trunk interface > ; > exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _91800NXXXXXX,2,Congestion > exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _91888NXXXXXX,2,Congestion > exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _91877NXXXXXX,2,Congestion > exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) > exten => _91866NXXXXXX,2,Congestion > > [international] > ; > ; Master context for international long distance > ; > ignorepat => 9 > include => longdistance > include => trunkint > > [longdistance] > ; > ; Master context for long distance > ; > ignorepat => 9 > include => local > include => trunkld > > [local] > ; > ; Master context for local, toll-free, and iaxtel calls only > ; > ignorepat => 9 > include => default > include => parkedcalls > include => trunklocal > include => iaxtel700 > include => trunktollfree > include => iaxprovider > ; > ; You can use an alternative switch type as well, to resolve > ; extensions that are not known here, for example with remote > ; IAX switching you transparently get access to the remote > ; Asterisk PBX > ; > ; switch => IAX2/user:[EMAIL PROTECTED]/local > > [macro-stdexten]; > ; > ; Standard extension macro: > ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well > ; ${ARG2} - Device(s) to ring > ; > exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum > exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status > (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) > > exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to > voicemail w/ unavail announce > exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start > > exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy > announce > exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start > > exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer > > exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into > VoicemailMain > > [demo] > ; > ; We start with what to do when a call first comes in. > ; > exten => s,1,Wait,1 ; Wait a second, just for fun > exten => s,2,Answer ; Answer the line > exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds > exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds > exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message > exten => s,6,BackGround(demo-instruct) ; Play some instructions > > exten => 2,1,BackGround(demo-moreinfo) ; Give some more information. > exten => 2,2,Goto(s,6) > > exten => 3,1,SetLanguage(fr) ; Set language to french > exten => 3,2,Goto(s,5) ; Start with the congratulations > > exten => 1000,1,Goto(default,s,1) > ; > ; We also create an example user, 1234, who is on the console and has > ; voicemail, etc. > ; > exten => 1234,1,Playback(transfer,skip) ; "Please hold while..." > ; (but skip if channel is not up) > exten => 1234,2,Macro(stdexten,1234,${CONSOLE}) > > exten => 1235,1,Voicemail(u1234) ; Right to voicemail > > exten => 1236,1,Dial(Console/dsp) ; Ring forever > exten => 1236,2,Voicemail(u1234) ; Unless busy > > ; > ; # for when they're done with the demo > ; > exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo" > exten => #,2,Hangup ; Hang them up. > > ; > ; A timeout and "invalid extension rule" > ; > exten => t,1,Goto(#,1) ; If they take too long, give up > exten => i,1,Playback(invalid) ; "That's not valid, try again" > > ; > ; Create an extension, 500, for dialing the > ; Asterisk demo. > ; > exten => 500,1,Playback(demo-abouttotry); Let them know what's going on > exten => 500,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; Call the > Asterisk demo > exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo site > exten => 500,4,Goto(s,6) ; Return to the start over message. > > ; > ; Create an extension, 600, for evaulating echo latency. > ; > exten => 600,1,Playback(demo-echotest) ; Let them know what's going on > exten => 600,2,Echo ; Do the echo test > exten => 600,3,Playback(demo-echodone) ; Let them know it's over > exten => 600,4,Goto(s,6) ; Start over > > ; > ; Give voicemail at extension 8500 > ; > exten => 8500,1,VoicemailMain > exten => 8500,2,Goto(s,6) > ; > ; Here's what a phone entry would look like (IXJ for example) > ; > ;exten => 1265,1,Dial(Phone/phone0,15) > ;exten => 1265,2,Goto(s,5) > > ;[mainmenu] > ; > ; Example "main menu" context with submenu > ; > ;exten => s,1,Answer > ;exten => s,2,Background(thanks) ; "Thanks for calling press 1 for sales, 2 > for support, ..." > ;exten => 1,1,Goto(submenu,s,1) > ;exten => 2,1,Hangup > ;include => default > ; > ;[submenu] > ;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback > ;exten => s,2,Wait,2 > ;exten => s,3,Background(submenuopts) ; "Thanks for calling the sales > department. Press 1 for steve, 2 for..." > ;exten => 1,1,Goto(default,steve,1) > ;exten => 2,1,Goto(default,mark,2) > > [default] > ; > ; By default we include the demo. In a production system, you > ; probably don't want to have the demo there. > ; > include => demo > > ; > ; Extensions like the two below can be used for FWD, Nikotel, sipgate etc. > ; Note that you must have a [sipprovider] section in sip.conf whereas > ; the otherprovider.net example does not require such a peer definition > ; > ;exten => _41X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,r) > ;exten => _42X.,1,Dial(SIP/user:[EMAIL PROTECTED]:[EMAIL PROTECTED],30,rT) > > ; Real extensions would go here. Generally you want real extensions to be 4 > or 5 > ; digits long (although there is no such requirement) and start with a > single > ; digit that is fairly large (like 6 or 7) so that you have plenty of room > to > ; overlap extensions and menu options without conflict. You can alias them > with > ; names, too and use global variables > > ;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1 ; Channel hints for presence > ;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer > ;exten => 6245,1,Dial(${HINT},20,rtT) ; Use hint as listed > ;exten => 6361,1,Dial(IAX2/JaneDoe,,rm) ; ring without time limit > ;exten => 6389,1,Dial(MGCP/aaln/[EMAIL PROTECTED]) > ;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK} > > ;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is > something like Zap/2 > ;exten => mark,1,Goto(6275|1) ; alias mark to 6275 > ;exten => 6536,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil > ;exten => wil,1,Goto(6236|1) > ; > ; Some other handy things are an extension for checking voicemail via > ; voicemailmain > ; > ;exten => 8500,1,VoicemailMain > ;exten => 8500,2,Hangup > ; > ; Or a conference room (you'll need to edit meetme.conf to enable this room) > ; > ;exten => 8600,1,Meetme(1234) > ; > ; Or playing an announcement to the called party, as soon it answers > ; > ;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg)) > ; > ; For more information on applications, just type "show applications" at > your > ; friendly Asterisk CLI prompt. > ; > ; 'show application <command>' will show details of how you > ; use that particular application in this file, the dial plan. > ; > > > > > ----- Original Message ----- > From: "dbruce" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Sent: Sunday, July 24, 2005 8:39 PM > Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202 > > > > Marc: My answer is not incorrect... it is incomplete. > > > > The OP stipulated 2 extensions 200 and 202... and provided a sip debug > > indicating a call from 200 to 777. > > > > I pointed out the obvious. > > > > If the OP is dialing 202 on the phone, and the phone is dialing 777, then > > he > > needs to look at the dialplan configuration of the phone. If he is dialing > > 777 on the phone and expecting to reach 202, then he will need to have > > translations in the asterisk dialplan. But, the question was "what should > > I > > be looking at?"... Using just the information provided, and the fact that > > he > > is new to asterisk... without any further information... the first thing > > he > > should be looking at is why the phone is trying to reach 777 when he wants > > to reach 202... Many new users do not realize the complexity of the SIP > > protocol, and only really look at the trace in a general manner... such > > as: > > INVITE > > 407 Proxy Authentication Required > > ACK > > INVITE > > 404 Not Found > > ACK > > > > The idea was to provide a clue... not to provide a complete working > > dialplan > > and phone configuration. Providing new users with "the complete package" > > is > > a dis-service to them. They will only learn from thier mistakes and > > experiences.. providing clues allows them to expand their experience and > > build their confidence... It requires them to look at the details and > > learn > > to analyse them. > > > > Regards, > > Derek > > > > > > ----- Original Message ----- > > From: "Marc Storck" <[EMAIL PROTECTED]> > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > <[email protected]> > > Sent: Sunday, July 24, 2005 12:53 PM > > Subject: Re: [Asterisk-Users] Why can't sip/200 call sip/202 > > > > > >> Derek: you reply is uncorrect. If Angus has the extension 777 in his > >> dialplan/extensions.conf which will dial 202. The name of the peer has > >> absolutely nothing to do with which number/name he would have to dial. > >> Without dialplan he will be unable to call any extension even 202, as > >> 202 is only the name of the peer. > >> > >> Angus: please paste your extensions.conf to pastebin.ca > >> > >> Regards, > >> > >> Marc > >> > >> dbruce wrote: > >> > It appears from the debug that extension 200 is trying to call 777, not > >> > 202. Your Asterisk server can't find an extension 777 and returns "404 > >> > not found". That will explain why you can't call extension 777 from > >> > extension 200. If you want to call extension 202, you will need to dial > >> > 202 on extension 200, not 777. > >> > > >> > Regards, > >> > Derek > >> > > >> > > >> > ----- Original Message ----- > >> > *From:* Angus Comber <mailto:[EMAIL PROTECTED]> > >> > *To:* [email protected] > >> > <mailto:[email protected]> > >> > *Sent:* Sunday, July 24, 2005 11:51 AM > >> > *Subject:* [Asterisk-Users] Why can't sip/200 call sip/202 > >> > > >> > I have 2 sip accounts setup - 200 and 202. If I do sip show peers > >> > I > >> > get: > >> > > >> > sip show peers > >> > Name/username Host Dyn Nat ACL Mask > >> > Port Status > >> > 202/202 192.168.0.6 D 255.255.255.255 > >> > 5060 Unmonitored > >> > 201/201 (Unspecified) D 255.255.255.255 > >> > 5060 Unmonitored > >> > 200/200 192.168.0.3 D 255.255.255.255 > >> > 5060 Unmonitored > >> > > >> > 200 is a Grandstream GXP200 IP Phone and 202 is a Grandstream BT100 > >> > IP phone. > >> > > >> > relevant bit of sip.conf: > >> > > >> > [200] > >> > username=200 > >> > type=friend > >> > secret=1234 > >> > port=5060 > >> > nat=never > >> > dtmfmode=rfc2833 > >> > context=default > >> > callerid="Angus Comber" <200> > >> > host=dynamic > >> > disallow=all > >> > allow=ulaw > >> > allow=alaw > >> > allow=g723.1 > >> > allow=g729 > >> > > >> > [202] > >> > username=202 > >> > type=friend > >> > secret=1234 > >> > port=5060 > >> > nat=never > >> > dtmfmode=rfc2833 > >> > context=default > >> > callerid="Sam Comber" <202> > >> > host=dynamic > >> > disallow=all > >> > allow=ulaw > >> > allow=alaw > >> > allow=g723.1 > >> > allow=g729 > >> > > >> > > >> > But whenever I try to dial between phones I get this: > >> > > >> > > >> > Sip read: > >> > > >> > 0 headers, 0 lines > >> > > >> > > >> > Sip read: > >> > INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 > >> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1 > >> > From: "Angus Comber" > >> > <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 > >> > To: <sip:[EMAIL PROTECTED];user=phone> > >> > Contact: <sip:[EMAIL PROTECTED];user=phone> > >> > Supported: replaces, timer > >> > Call-ID: [EMAIL PROTECTED] > >> > <mailto:[EMAIL PROTECTED]> > >> > CSeq: 45925 INVITE > >> > User-Agent: Grandstream GXP2000 1.0.1.9 > >> > Max-Forwards: 70 > >> > Allow: > >> > > > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK > >> > Content-Type: application/sdp > >> > Content-Length: 258 > >> > > >> > v=0 > >> > o=200 8000 8000 IN IP4 192.168.0.3 > >> > s=SIP Call > >> > c=IN IP4 192.168.0.3 > >> > t=0 0 > >> > m=audio 5004 RTP/AVP 18 0 8 101 > >> > a=sendrecv > >> > a=rtpmap:18 G729/8000 > >> > a=rtpmap:0 PCMU/8000 > >> > a=rtpmap:8 PCMA/8000 > >> > a=ptime:20 > >> > a=rtpmap:101 telephone-event/8000 > >> > a=fmtp:101 0-11 > >> > > >> > 13 headers, 13 lines > >> > Using latest request as basis request > >> > Sending to 192.168.0.3 : 5060 (non-NAT) > >> > Reliably Transmitting (no NAT): > >> > SIP/2.0 407 Proxy Authentication Required > >> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1 > >> > From: "Angus Comber" > >> > <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 > >> > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be > >> > Call-ID: [EMAIL PROTECTED] > >> > <mailto:[EMAIL PROTECTED]> > >> > CSeq: 45925 INVITE > >> > User-Agent: Asterisk PBX > >> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > >> > Contact: <sip:[EMAIL PROTECTED]> > >> > Proxy-Authenticate: Digest realm="asterisk", nonce="0c555366" > >> > Content-Length: 0 > >> > > >> > > >> > to 192.168.0.3:5060 > >> > Scheduling destruction of call '[EMAIL PROTECTED]' > >> > <mailto:'[EMAIL PROTECTED]'> in 15000 ms > >> > Found user '200' > >> > > >> > > >> > Sip read: > >> > ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 > >> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKa6cf8b6a7c7198a1 > >> > From: "Angus Comber" > >> > <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 > >> > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be > >> > Contact: <sip:[EMAIL PROTECTED];user=phone> > >> > Call-ID: [EMAIL PROTECTED] > >> > <mailto:[EMAIL PROTECTED]> > >> > CSeq: 45925 ACK > >> > User-Agent: Grandstream GXP2000 1.0.1.9 > >> > Max-Forwards: 70 > >> > Allow: > >> > > > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK > >> > Content-Length: 0 > >> > > >> > > >> > 11 headers, 0 lines > >> > > >> > > >> > Sip read: > >> > INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 > >> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 > >> > From: "Angus Comber" > >> > <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 > >> > To: <sip:[EMAIL PROTECTED];user=phone> > >> > Contact: <sip:[EMAIL PROTECTED];user=phone> > >> > Supported: replaces, timer > >> > Proxy-Authorization: Digest username="200", realm="asterisk", > >> > algorithm=MD5, uri="sip:[EMAIL PROTECTED];user=phone", > >> > nonce="0c555366", response="ee6088fb4e50da5fe412913ae40dd45c" > >> > Call-ID: [EMAIL PROTECTED] > >> > <mailto:[EMAIL PROTECTED]> > >> > CSeq: 45926 INVITE > >> > User-Agent: Grandstream GXP2000 1.0.1.9 > >> > Max-Forwards: 70 > >> > Allow: > >> > > > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK > >> > Content-Type: application/sdp > >> > Content-Length: 258 > >> > > >> > v=0 > >> > o=200 8000 8001 IN IP4 192.168.0.3 > >> > s=SIP Call > >> > c=IN IP4 192.168.0.3 > >> > t=0 0 > >> > m=audio 5004 RTP/AVP 18 0 8 101 > >> > a=sendrecv > >> > a=rtpmap:18 G729/8000 > >> > a=rtpmap:0 PCMU/8000 > >> > a=rtpmap:8 PCMA/8000 > >> > a=ptime:20 > >> > a=rtpmap:101 telephone-event/8000 > >> > a=fmtp:101 0-11 > >> > > >> > 14 headers, 13 lines > >> > Using latest request as basis request > >> > Sending to 192.168.0.3 : 5060 (non-NAT) > >> > Found user '200' > >> > Found RTP audio format 18 > >> > Found RTP audio format 0 > >> > Found RTP audio format 8 > >> > Found RTP audio format 101 > >> > Peer audio RTP is at port 192.168.0.3:5004 > >> > Found description format G729 > >> > Found description format PCMU > >> > Found description format PCMA > >> > Found description format telephone-event > >> > Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x10c > >> > (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c > > (ulaw|alaw|g729) > >> > Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), > >> > combined > >> > - 0x1 (g723) > >> > Looking for 777 in default > >> > Reliably Transmitting (no NAT): > >> > SIP/2.0 404 Not Found > >> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 > >> > From: "Angus Comber" > >> > <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 > >> > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be > >> > Call-ID: [EMAIL PROTECTED] > >> > <mailto:[EMAIL PROTECTED]> > >> > CSeq: 45926 INVITE > >> > User-Agent: Asterisk PBX > >> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > >> > Contact: <sip:[EMAIL PROTECTED]> > >> > Content-Length: 0 > >> > > >> > > >> > to 192.168.0.3:5060 > >> > > >> > > >> > Sip read: > >> > ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 > >> > Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bKe570dfe4b8441304 > >> > From: "Angus Comber" > >> > <sip:[EMAIL PROTECTED];user=phone>;tag=a1afaf4fdb0ac845 > >> > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as668982be > >> > Contact: <sip:[EMAIL PROTECTED];user=phone> > >> > Proxy-Authorization: Digest username="200", realm="asterisk", > >> > algorithm=MD5, uri="sip:[EMAIL PROTECTED];user=phone", > >> > nonce="0c555366", response="7fcb1024a81b3ea3bcc56baeca4bac3e" > >> > Call-ID: [EMAIL PROTECTED] > >> > <mailto:[EMAIL PROTECTED]> > >> > CSeq: 45926 ACK > >> > User-Agent: Grandstream GXP2000 1.0.1.9 > >> > Max-Forwards: 70 > >> > Allow: > >> > > > INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK > >> > Content-Length: 0 > >> > > >> > > >> > 12 headers, 0 lines > >> > Destroying call '[EMAIL PROTECTED]' > >> > <mailto:'[EMAIL PROTECTED]'> > >> > > >> > > >> > How can I troubleshoot? What should I be looking at? > >> > > >> > Angus > >> > > >> > > >> > ------------------------------------------------------------------------ > >> > > >> > _______________________________________________ > >> > Asterisk-Users mailing list > >> > [email protected] > >> > http://lists.digium.com/mailman/listinfo/asterisk-users > >> > To UNSUBSCRIBE or update options visit: > >> > http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > >> > > >> > ------------------------------------------------------------------------ > >> > > >> > _______________________________________________ > >> > Asterisk-Users mailing list > >> > [email protected] > >> > http://lists.digium.com/mailman/listinfo/asterisk-users > >> > To UNSUBSCRIBE or update options visit: > >> > http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> -- > >> CTO Marc Storck > >> MS Networks SA [EMAIL PROTECTED] > >> IT Service Provider http://www.msnetworks.lu > >> 15, route d'Esch Phone: +352 2727 3030 > >> L-4450 Belvaux Fax: +352 2727 3060 > >> > >> --------------- MS Networks powered service --------------- > >> http://www.LuxAdmin.com Hosting and housing solutions > >> ----------------------------------------------------------- > >> > >> _______________________________________________ > >> Asterisk-Users mailing list > >> [email protected] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > > Asterisk-Users mailing list > > [email protected] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
