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
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