[asterisk-users] Asterisk with OpenBTS and mobile phone
Ioan Indreias
indreias at gmail.com
Fri Jul 13 09:06:28 CDT 2012
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
>
> --
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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
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