Asterisk with OpenBTS and mobile phone

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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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