[asterisk-users] Asterisk with OpenBTS and mobile phone
Ellen Apolinar
ellen.apolinar.td at googlemail.com
Thu Jul 12 07:55:05 CDT 2012
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
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