Ellen Apolinar
2012-Jul-12 12:55 UTC
[asterisk-users] Asterisk with OpenBTS and mobile phone
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 at DLPN_DialPlan1:1] Macro("SIP/6000-00000013", > "stdexten,6201,SIP/6201") in new stack > -- Executing [s at 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 at macro-stdexten:2] GotoIf("SIP/6000-00000013", > "?5:3") in new stack > -- Goto (macro-stdexten,s,3) > -- Executing [s at 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 at macro-stdexten:4] Goto("SIP/6000-00000013", > "s-CHANUNAVAIL,1") in new stack > -- Goto (macro-stdexten,s-CHANUNAVAIL,1) > -- Executing [s-CHANUNAVAIL at macro-stdexten:1] > Goto("SIP/6000-00000013", "s-NOANSWER,1") in new stack > -- Goto (macro-stdexten,s-NOANSWER,1) > -- Executing [s-NOANSWER at 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. 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Ioan Indreias
2012-Jul-13 14:06 UTC
[asterisk-users] Asterisk with OpenBTS and mobile phone
On Thu, Jul 12, 2012 at 3:55 PM, Ellen Apolinar <ellen.apolinar.td at 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 at DLPN_DialPlan1:1] Macro("SIP/6000-00000013", >> "stdexten,6201,SIP/6201") in new stack >> -- Executing [s at 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 at macro-stdexten:2] GotoIf("SIP/6000-00000013", >> "?5:3") in new stack >> -- Goto (macro-stdexten,s,3) >> -- Executing [s at 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 at macro-stdexten:4] Goto("SIP/6000-00000013", >> "s-CHANUNAVAIL,1") in new stack >> -- Goto (macro-stdexten,s-CHANUNAVAIL,1) >> -- Executing [s-CHANUNAVAIL at macro-stdexten:1] >> Goto("SIP/6000-00000013", "s-NOANSWER,1") in new stack >> -- Goto (macro-stdexten,s-NOANSWER,1) >> -- Executing [s-NOANSWER at 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-usersYour 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