[asterisk-users] Asterisk with OpenBTS and mobile phone
Ellen Apolinar
ellen.apolinar.td at googlemail.com
Wed Jul 18 05:37:31 CDT 2012
Hey Ioan,
thanks for your answer.
It helped a little bit but I have no idea what exactly could work wrong.
My new situation:
*CLI> originate SIP/123456789101112 application MusicOnHold
> == Using SIP RTP CoS mark 5
> -- Got SIP response 482 "Loop Detected" back from 192.168.0.102:5060
> [Jul 18 10:38:27] WARNING[4615]: chan_sip.c:3873 __sip_autodestruct:
> Autodestruct on dialog '
> 446588d34c8b0e2d1920fec416ef0b5d at 192.168.0.102:5060' with owner in place
> (Method: INVITE)
>
*CLI> sip show peers
> Name/username Host Dyn
> Forcerport ACL Port Status
> 123456789101112/6202 192.168.0.102
> N 5060 OK (1 ms)
> 6000/6000 192.168.0.102 D
> N 5061 Unmonitored
> 6001/6001 192.168.0.102 D
> N 5061 Unmonitored
>
*CLI> sip show channels
> Peer User/ANR Call ID Format
> Hold Last Message Expiry Peer
> 192.168.0.102 (None) 2dab9ef669bc9a4 0x0 (nothing)
> No Rx: OPTIONS <guest>
> 1 active SIP dialog
>
I thought with 6201 I could build a connection to Asterisk. In the
extensions.conf and in the Asterisk-GUI the numbers from 6000 - 6300 (not
all, just a frew of them) are shown so I choosed one of them like I did
with the softphones.
asterisk -rx doesn't work.
What do you think is wrong with my extensions.conf?
Best regards.
Ellen
On Fri, Jul 13, 2012 at 4:06 PM, Ioan Indreias <indreias at gmail.com> wrote:
> 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-users
>
> 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
>
> --
> _____________________________________________________________________
> -- 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-users
>
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