I'm trying to add a second extension to my setup. The second device is able to successfully connect to the Asterisk server. I am unable to contact extension 101 from 102 and vise-versa. Also are my context setup logically or is there a better fashion to organize them? My error is at the bottom. Here is the extension.conf [default] ; ; By default we include the demo. In a production system, you ; probably don't want to have the demo there. ; ;include => demo exten => 101,1,Dial(SIP/101&SIP/9046260705 at vitel-outbound,30) exten => 101,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?lbl_default_1:) exten => 101,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?lbl_default_1:) exten => 101,n(lbl_default_0),Hangup() exten => 101,n(lbl_default_1),Dial(SIP/9046260705 at vitel-outbond,30) exten => 101,n,Goto(lbl_default_0) exten => 102,1,Dial(SIP/102,20) exten => 102,n,Hangup ;This automatically calls the right mailbox using the ${CALLERIDNUM} variable in the current context (var ${CONTEXT}). exten=>*98,1,VoiceMailMain(${CALLERIDNUM}@${CONTEXT}) include => inbound include => outgoing [inbound] exten => 9045622082,1,Goto(default,101,1) [outgoing] ; The following gives an Unknown Caller ID ;exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=XXXXXXXXXX) ;exten => _1NXXNXXXXXX,2,Set(CALLERID(name)=XXXXXXXXXX) exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _1NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@vitel-outbound) exten => _NXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _NXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _NXXXXXX,n,Dial(SIP/1904${EXTEN}@vitel-outbound) exten => _NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) exten => _NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese") exten => _NXXNXXXXXX,n,Dial(SIP/1${EXTEN}@vitel-outbound) exten => _011.,1,Set(CALLERID(num)=9045622082) exten => _011.,n,Set(CALLERID(name)="Stephen Reese") exten => _011.,n,Dial(SIP/${EXTEN}@vitel-outbound) exten => _911,1,Set(CALLERID(num)=9045622082) exten => _911,n,Set(CALLERID(name)="Stephen Reese") exten => _911,n,Dial(SIP/911 at vitel-outbound) This is a call from extension 101 to 102 that fails with a busy signal. -- Executing [102 at default:1] Dial("SIP/101-08266f60", "'SIP/102',20") in new stack [Oct 19 15:28:28] WARNING[26596]: channel.c:3470 ast_request: No channel type registered for ''SIP' [Oct 19 15:28:28] WARNING[26596]: app_dial.c:1450 dial_exec_full: Unable to create channel of type ''SIP' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [102 at default:2] Hangup("SIP/101-08266f60", "") in new stack == Spawn extension (default, 102, 2) exited non-zero on 'SIP/101-08266f60'
What is vitel-outbound?? an IP address?? And what version of Asterisk is this? Regards, Juan On Sun, Oct 19, 2008 at 3:30 PM, Stephen Reese <rsreese at gmail.com> wrote:> I'm trying to add a second extension to my setup. The second device is > able to successfully connect to the Asterisk server. I am unable to > contact extension 101 from 102 and vise-versa. Also are my context > setup logically or is there a better fashion to organize them? My > error is at the bottom. > > Here is the extension.conf > > [default] > ; > ; By default we include the demo. In a production system, you > ; probably don't want to have the demo there. > ; > ;include => demo > > exten => 101,1,Dial(SIP/101&SIP/9046260705 at vitel-outbound,30) > exten => 101,n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?lbl_default_1:) > exten => 101,n,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?lbl_default_1:) > exten => 101,n(lbl_default_0),Hangup() > exten => 101,n(lbl_default_1),Dial(SIP/9046260705 at vitel-outbond,30) > exten => 101,n,Goto(lbl_default_0) > > exten => 102,1,Dial(SIP/102,20) > exten => 102,n,Hangup > > ;This automatically calls the right mailbox using the ${CALLERIDNUM} > variable in the current context (var ${CONTEXT}). > exten=>*98,1,VoiceMailMain(${CALLERIDNUM}@${CONTEXT}) > > include => inbound > include => outgoing > > [inbound] > exten => 9045622082,1,Goto(default,101,1) > > [outgoing] > ; The following gives an Unknown Caller ID > ;exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=XXXXXXXXXX) > ;exten => _1NXXNXXXXXX,2,Set(CALLERID(name)=XXXXXXXXXX) > > exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) > exten => _1NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese") > exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@vitel-outbound) > > exten => _NXXXXXX,1,Set(CALLERID(num)=9045622082) > exten => _NXXXXXX,n,Set(CALLERID(name)="Stephen Reese") > exten => _NXXXXXX,n,Dial(SIP/1904${EXTEN}@vitel-outbound) > > exten => _NXXNXXXXXX,1,Set(CALLERID(num)=9045622082) > exten => _NXXNXXXXXX,n,Set(CALLERID(name)="Stephen Reese") > exten => _NXXNXXXXXX,n,Dial(SIP/1${EXTEN}@vitel-outbound) > > exten => _011.,1,Set(CALLERID(num)=9045622082) > exten => _011.,n,Set(CALLERID(name)="Stephen Reese") > exten => _011.,n,Dial(SIP/${EXTEN}@vitel-outbound) > > exten => _911,1,Set(CALLERID(num)=9045622082) > exten => _911,n,Set(CALLERID(name)="Stephen Reese") > exten => _911,n,Dial(SIP/911 at vitel-outbound) > > This is a call from extension 101 to 102 that fails with a busy signal. > > -- Executing [102 at default:1] Dial("SIP/101-08266f60", > "'SIP/102',20") in new stack > [Oct 19 15:28:28] WARNING[26596]: channel.c:3470 ast_request: No > channel type registered for ''SIP' > [Oct 19 15:28:28] WARNING[26596]: app_dial.c:1450 dial_exec_full: > Unable to create channel of type ''SIP' (cause 66 - Channel not > implemented) > == Everyone is busy/congested at this time (1:0/0/1) > -- Executing [102 at default:2] Hangup("SIP/101-08266f60", "") in new > stack > == Spawn extension (default, 102, 2) exited non-zero on 'SIP/101-08266f60' > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Juan E. Rodr?guez Cel. 829-886-5565 Work: 809-724-9227 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081019/6e1f5331/attachment.htm
On Sun, Oct 19, 2008 at 4:11 PM, Juan Rodr?guez <jerdguez at gmail.com> wrote:> First, I think is better to to have SIP/vitel-outbound/${EXTEN} than > having SIP/${EXTEN}@vitel-outbound > And try issuing SIP SET DEBUG on the cli to see what happens when making the > call, post back what you see making calls from 101 to 102 and 102 to 101. > Having the sip.conf sould help on getting whats going on.Here are the relevant parts of the sip.conf [general] register => rsreese:test at inbound18.vitelity.net:5060/rsreese context=default ; Default context for incoming calls ;allowguest=no ; Allow or reject guest calls (default is yes) ;match_auth_username=yes ; if available, match user entry using the ; 'username' field from the authentication line ; instead of the From: field. allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) ; Default is enabled realm=ns1.neocipher.net ; Realm for digest authentication ; defaults to "asterisk". If you set a system name in ; asterisk.conf, it defaults to that system name ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port for unencrypted UDP ; and TCP sessions is 5060) ; bindport is the local UDP port that Asterisk will listen on bindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) ; You can specify port here too, like 123.123.123.123:5080 domain=neocipher.net [101] type=friend ; allows incoming and outgoing calls username=101 secret=pass mailbox=101 callerid=\"Stephen\" <101> host=dynamic nat=yes dtmfmode=rfc2833 canreinvite=no reinvite=no qualify=yes [102] type=friend ; allows incoming and outgoing calls username=102 secret=pass mailbox=102 callerid=\"Stephen\" <102> host=dynamic nat=yes dtmfmode=rfc2833 canreinvite=no reinvite=no qualify=yes [vitel-inbound] ;(exact format/casing required) type=friend host=inbound18.vitelity.net context=inbound ;(ext-did or from-trunk for A at H) username=rsreese secret=test allow=all ;insecure=very insecure = invite canreinvite=no [vitel-outbound] ;(exact format/casing required) type=friend host=outbound.vitelity.net context=inbound ;(ext-did or from-trunk for A at H) username=rsreese fromuser=rsreese trustrpid=yes sendrpid=yes secret=test allow=all canreinvite=no Here is the sip debug error: <------------> -- Executing [102 at default:1] Dial("SIP/102-08266f60", "'SIP/102',20") in new stack [Oct 19 16:21:21] WARNING[26690]: channel.c:3470 ast_request: No channel type registered for ''SIP' [Oct 19 16:21:21] WARNING[26690]: app_dial.c:1450 dial_exec_full: Unable to create channel of type ''SIP' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [102 at default:2] Hangup("SIP/102-08266f60", "") in new stack == Spawn extension (default, 102, 2) exited non-zero on 'SIP/102-08266f60' Scheduling destruction of SIP dialog 'NTQxOTRlZjI2MmEzMWYyOTliZmI2ZDJkMTVkOTYzZDQ.' in 32000 ms (Method: INVITE) ns1*CLI> <--- Reliably Transmitting (NAT) to 68.156.63.118:56558 ---> SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 68.156.63.118:56558;branch=z9hG4bK-d8754z-e0a3d830adb35401-1---d8754z-;received=68.156.63.118;rport=56558 From: <sip:102 at neocipher.net>;tag=7d39014c To: "102"<sip:102 at neocipher.net>;tag=as4f32f2a7 Call-ID: NTQxOTRlZjI2MmEzMWYyOTliZmI2ZDJkMTVkOTYzZDQ. CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:102 at 209.251.157.91> Content-Length: 0
On Sun, Oct 19, 2008 at 5:43 PM, Juan Rodr?guez <jerdguez at gmail.com> wrote:> Try reinstalling Asterisk, because in the channel.c this error is returned > if the channels TEC (in this case SIP) is not found. > Weird!! > Let me know if it works. > Regards, > JuanSo the extensions.conf and sip.conf look correct? I tried reinstalling and I still am unable to communicate between the two extensions.
Stephen: Your configuration files looks fine. Try from the CLI issuing "originate SIP/101 extension 102 at default", having the 101 online, then do that with "originate SIP/102 extension 101 at default". See what happens. If you got a CVS commit, commit again or try installing a release. http://downloads.digium.com/pub/asterisk/asterisk-1.6-current.tar.gz (for download) Regards, Juan On Sun, Oct 19, 2008 at 10:36 PM, Stephen Reese <rsreese at gmail.com> wrote:> On Sun, Oct 19, 2008 at 5:43 PM, Juan Rodr?guez <jerdguez at gmail.com> > wrote: > > Try reinstalling Asterisk, because in the channel.c this error is > returned > > if the channels TEC (in this case SIP) is not found. > > Weird!! > > Let me know if it works. > > Regards, > > Juan > > So the extensions.conf and sip.conf look correct? I tried reinstalling > and I still am unable to communicate between the two extensions. >-- Juan E. Rodr?guez Cel. 829-886-5565 Work: 809-724-9227 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081019/e4893fa8/attachment.htm
On Tue, Oct 21, 2008 at 9:56 AM, Juan Rodr?guez <jerdguez at gmail.com> wrote:> Try changing: > exten => 101,1,Dial(SIP/101/20) > to > exten => 101,1,Dial(SIP/101|20) or exten => 101,1,Dial(SIP/101,20) > > because exten => 101,1,Dial(SIP/101/20) means you are trying to contact ext. > 20 on through a trunk called 101.Oh, typo, but that still didn't cure it.... Successful call from from 101 to 102 == Using SIP RTP CoS mark 5 -- Executing [102 at default:1] Dial("SIP/101-08220318", "SIP/102,20") in new stack == Using SIP RTP CoS mark 5 -- Called 102 -- SIP/102-08221a78 is ringing -- SIP/102-08221a78 answered SIP/101-08220318 -- Packet2Packet bridging SIP/101-08220318 and SIP/102-08221a78 == Spawn extension (default, 102, 1) exited non-zero on 'SIP/101-08220318' Failed call from 102 to 101 == Using SIP RTP CoS mark 5 -- Executing [101 at default:1] Dial("SIP/102-08221a78", "SIP/101,20") in new stack == Using SIP RTP CoS mark 5 -- Called 101 -- Got SIP response 400 "Bad Request" back from 68.156.63.118 -- SIP/101-0821e110 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [101 at default:2] Hangup("SIP/102-08221a78", "") in new stack == Spawn extension (default, 101, 2) exited non-zero on 'SIP/102-08221a78'
And this phone are connected in a local LAN?? Because I see Asterisk receiving a "Bad request" from 68.156.63.118 If those phones are not in your local LAN, try with a soft phone first. Could be Zoiper or Xlite. Besides, use SIP SET DEBUG, for SIP debugging and try to see why is 101 sending a "400 Bad request" back to Asterisk. On Wed, Oct 22, 2008 at 9:10 PM, Stephen Reese <rsreese at gmail.com> wrote:> On Wed, Oct 22, 2008 at 8:15 PM, Juan Rodr?guez <jerdguez at gmail.com> > wrote: > > What kind of phone are you trying to connect to 101??? and from where? > > > > Both phones are Cisco, 101 is a 7960 and 102 is a 7912. 101 can > contact 102 by dialing 101 but not the other way around, I just get a > busy tone. >-- Juan E. Rodr?guez Cel. 829-886-5565 Work: 809-724-9227 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081023/1ffaf1bf/attachment.htm