Hey Ioan, thanks for your answer.
It helped a little bit but I have no idea what exactly could work wrong. My new situation: *CLI> originate SIP/123456789101112 application MusicOnHold > == Using SIP RTP CoS mark 5 > -- Got SIP response 482 "Loop Detected" back from 192.168.0.102:5060 > [Jul 18 10:38:27] WARNING[4615]: chan_sip.c:3873 __sip_autodestruct: > Autodestruct on dialog ' > 446588d34c8b0e2d1920fec416ef0b5d@192.168.0.102:5060' with owner in place > (Method: INVITE) > *CLI> sip show peers > Name/username Host Dyn > Forcerport ACL Port Status > 123456789101112/6202 192.168.0.102 > N 5060 OK (1 ms) > 6000/6000 192.168.0.102 D > N 5061 Unmonitored > 6001/6001 192.168.0.102 D > N 5061 Unmonitored > *CLI> sip show channels > Peer User/ANR Call ID Format > Hold Last Message Expiry Peer > 192.168.0.102 (None) 2dab9ef669bc9a4 0x0 (nothing) > No Rx: OPTIONS <guest> > 1 active SIP dialog > I thought with 6201 I could build a connection to Asterisk. In the extensions.conf and in the Asterisk-GUI the numbers from 6000 - 6300 (not all, just a frew of them) are shown so I choosed one of them like I did with the softphones. asterisk -rx doesn't work. What do you think is wrong with my extensions.conf? Best regards. Ellen On Fri, Jul 13, 2012 at 4:06 PM, Ioan Indreias <indre...@gmail.com> wrote: > On Thu, Jul 12, 2012 at 3:55 PM, Ellen Apolinar > <ellen.apolinar...@googlemail.com> wrote: > > Hello mailinglist, > > > > I want to connect Asterisk with OpenBTS and make a call with a mobile > phone. > > > > I use: > > Ubuntu 11.10 + Kernel 3.0.22 > > GnuRadio 3.3.0 > > Asterisk 1.8.13 > > OpenBTS 2.8 > > Nokia Mobile Phone > > > > OpenBTS works and I can send sms from the OpenBTS server to the > > mobile phone. What I also need is a call between Asterisk and OpenBTS. > > > > I have also two soft phones which works with Asterisk. And also OpenBSC > > is working with Asterisk successfully (OpenBSC is another project). > > > > Perhaps you can help me because I think it is an issue with Asterisk. > > > > > > sip.conf: > >> > >> ;SIP-Phones (Twinkle) > >> [user1] > >> callerid = 6000 > >> username = 6000 > >> secret = 6000 > >> canreinvite = no > >> type = friend > >> context = phones > >> allow = all > >> host = dynamic > >> dtmfmode = info > >> > >> [user2] > >> callerid = 6001 > >> username = 6001 > >> secret = 6001 > >> canreinvite = no > >> type = friend > >> context = phones > >> allow = all > >> host = dynamic > >> dtmfmode = info > >> > >> ; Mobile phone > >> [123456789101112] > >> callerid = 6201 > >> username = 6201 > >> secret = 6201 > >> canreinvite = no > >> type = friend > >> context = sip_external > >> ;context = open-bts > >> disallow = all > >> allow = gsm > >> host = 192.168.0.102 > >> domain = 192.168.0.102 > >> dtmfmode = info > > > > > > extensions.conf > >> > >> [internal] > >> exten => s,1,Verbose(1|Echo test application) > >> exten => s,n,Echo() > >> exten => s,n,Hangup() > >> exten => 6000,1,Verbose(1|Extension 6000) > >> exten => 6000,n,Dial(SIP/user1,30) > >> exten => 6000,n,Hangup() > >> exten => 6001,1,Verbose(1|Extension 6001) > >> exten => 6001,n,Dial(SIP/user2,30) > >> exten => 6001,n,Hangup() > >> > >> [phones] > >> include => internal > >> include => default > >> > >> [open-bts] > >> exten => 6002,1,Playback(demo-echotest) > >> exten => 6002,n,Echo > >> exten => 6002,n,Playback(demo-echodone) > >> exten => 6002,n,HangUp > >> > >> [sip_external] > >> exten => 6201,1,Macro(dialGSM,123456789101112) > >> > >> [macro-dialGSM] > >> exten => s,1,Dial(SIP/${ARG1},20) > >> exten => s,n,Goto(s-${DIALSTATUS},1) > >> exten => s-CANCEL,1,Hangup > >> exten => s-NOANSWER,1,Hangup > >> exten => s-BUSY,1,Busy(30) > >> exten => s-CONGESTION,1,Congestion (30) > >> exten => s-CHANUNAVAIL,1,Read(extension_digits,pbx-invalid) > >> exten => s-CHANUNAVAIL,n,GoTo(open-bts,${extension_digits},1) > > > > I have tried both contexts, [open-bts] and [sip_external] and both don't > > work > > > > > > If I want to call the mobile phone (6201) with a Twinkle soft phone > (6000) > > I get following message in the CLI-window from Asterisk: > >> > >> == Using SIP RTP CoS mark 5 > >> -- Executing [6201@DLPN_DialPlan1:1] Macro("SIP/6000-00000013", > >> "stdexten,6201,SIP/6201") in new stack > >> -- Executing [s@macro-stdexten:1] Set("SIP/6000-00000013", > >> "__DYNAMIC_FEATURES=") in new stack > >> [Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:468 ast_yyerror: > >> ast_yyerror(): syntax error: syntax error, unexpected '=', expecting > $end; > >> Input: > >> = 1 > >> ^ > >> [Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:472 ast_yyerror: If > you > >> have questions, please refer to > >> https://wiki.asterisk.org/wiki/display/AST/Channel+Variables > >> -- Executing [s@macro-stdexten:2] GotoIf("SIP/6000-00000013", > >> "?5:3") in new stack > >> -- Goto (macro-stdexten,s,3) > >> -- Executing [s@macro-stdexten:3] Dial("SIP/6000-00000013", > >> "SIP/6201,20,") in new stack > >> [Jul 12 12:14:29] WARNING[7092]: app_dial.c:2274 dial_exec_full: > >> Unable to create channel of type 'SIP' (cause 20 - Unknown) > >> == Everyone is busy/congested at this time (1:0/0/1) > >> -- Executing [s@macro-stdexten:4] Goto("SIP/6000-00000013", > >> "s-CHANUNAVAIL,1") in new stack > >> -- Goto (macro-stdexten,s-CHANUNAVAIL,1) > >> -- Executing [s-CHANUNAVAIL@macro-stdexten:1] > >> Goto("SIP/6000-00000013", "s-NOANSWER,1") in new stack > >> -- Goto (macro-stdexten,s-NOANSWER,1) > >> -- Executing [s-NOANSWER@macro-stdexten:1] > >> VoiceMail("SIP/6000-00000013", "6201,u") in new stack > >> -- <SIP/6000-00000013> Playing 'vm-theperson.gsm' (language > 'en') > >> -- <SIP/6000-00000013> Playing 'digits/6.gsm' (language 'en') > >> -- <SIP/6000-00000013> Playing 'digits/2.gsm' (language 'en') > >> -- <SIP/6000-00000013> Playing 'digits/0.gsm' (language 'en') > >> -- <SIP/6000-00000013> Playing 'digits/1.gsm' (language 'en') > >> -- <SIP/6000-00000013> Playing 'vm-isunavail.gsm' (language > 'en') > >> -- <SIP/6000-00000013> Playing 'vm-intro.gsm' (language 'en') > >> == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero > >> on 'SIP/6000-00000013' in macro 'stdexten' > >> == Spawn extension (DLPN_DialPlan1, 6201, 1) exited non-zero on > >> 'SIP/6000-00000013' > > > > > > > > *CLI> sip show peers > >> > >> Name/username Host > Dyn > >> Forcerport ACL Port Status > >> 123456789101112/6201 192.168.0.102 > >> N 5060 Unmonitored > >> 6000/6000 192.168.0.102 > D > >> N 5061 Unmonitored > >> 6001/6001 192.168.0.102 > D > >> N 5061 Unmonitored > >> (...) > >> user1/6000 (Unspecified) > D > >> N 0 Unmonitored > >> user2/6001 (Unspecified) > D > >> N 0 Unmonitored > > > > > > *CLI> sip show peer 123456789101112 > >> > >> * Name : 123456789101112 > >> Secret : <Set> > >> MD5Secret : <Not set> > >> Remote Secret: <Not set> > >> Context : sip_external > >> Subscr.Cont. : device-hints > >> Language : > >> AMA flags : Unknown > >> Transfer mode: open > >> CallingPres : Presentation Allowed, Not Screened > >> Callgroup : > >> Pickupgroup : > >> MOH Suggest : > >> Mailbox : > >> VM Extension : asterisk > >> LastMsgsSent : 32767/65535 > >> Call limit : 0 > >> Max forwards : 0 > >> Dynamic : No > >> Callerid : "" <6201> > >> MaxCallBR : 384 kbps > >> Expire : -1 > >> Insecure : no > >> Force rport : Yes > >> ACL : No > >> DirectMedACL : No > >> T.38 support : No > >> T.38 EC mode : Unknown > >> T.38 MaxDtgrm: -1 > >> DirectMedia : No > >> PromiscRedir : No > >> User=Phone : No > >> Video Support: No > >> Text Support : No > >> Ign SDP ver : No > >> Trust RPID : No > >> Send RPID : No > >> Subscriptions: Yes > >> Overlap dial : No > >> DTMFmode : info > >> Timer T1 : 500 > >> Timer B : 32000 > >> ToHost : 192.168.0.102 > >> Addr->IP : 192.168.0.102:5060 > >> Defaddr->IP : (null) > >> Prim.Transp. : UDP > >> Allowed.Trsp : UDP > >> Def. Username: 6201 > >> SIP Options : (none) > >> Codecs : 0x80030c7fffff > >> > (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719) > >> Codec Order : (none) > >> Auto-Framing : No > >> Status : Unmonitored > >> Useragent : > >> Reg. Contact : > >> Qualify Freq : 60000 ms > >> Sess-Timers : Accept > >> Sess-Refresh : uas > >> Sess-Expires : 1800 secs > >> Min-Sess : 90 secs > >> RTP Engine : asterisk > >> Parkinglot : > >> Use Reason : No > >> Encryption : No > > > > > > Asterisk log file (path: /var/log/asterisk/cdr-csv/Master.csv): > >> > >> "","6000","6201","DLPN_DialPlan1","""6000"" > >> <6000>","SIP/6000-00000013","","VoiceMail","6201,u","2012-07-12 > >> 10:14:29","2012-07-12 10:14:29","2012-07-12 > >> 10:14:35",6,6,"ANSWERED","DOCUMENTATION","1342088069.31","" > > > > > > > > > > If you need more informations write me and I will give you. It would be > very > > appreciated if some of you can help me or has an idea how I can fix this > > erorr. > > > > Best regards and thanks for helping. > > Ellen > > > > -- > > _____________________________________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > Your extensions.conf looks to be incomplete. Any way, dialling > SIP/6201 failed as 6201 is not a valid SIP account (you probably like > to dial SIP/123456789101112 > > Please try the following command: > asterisk -rx "originate SIP/123456789101112 application MusicOnHold" > and check asterisk logs. It should dial to the mobile phone and > connect to the MOH application. > > HTH, > Ioan > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users