[asterisk-users] Asterisk with OpenBTS and mobile phone

Ellen Apolinar ellen.apolinar.td at googlemail.com
Mon Jul 23 11:24:11 CDT 2012


Hey mailinglist,

my problem still exists and I need a little bit help.

When I start Asterisk, I do the following:

> asterisk -rvvvvv
> originate SIP/IMSI123456789101112 application MusicOnHold
>

Perhaps this will help you:

 *CLI> sip show peers

> Name/username              Host                                    Dyn
> Forcerport ACL Port     Status
> 6000/6000                  192.168.0.102                            D
> N             5061     Unmonitored
> 6001/6001                  192.168.0.102                            D
> N             5061     Unmonitored
> ...
> IMSI123456789101112        192.168.0.102
> N             5060     OK (1 ms)
> 36 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 33
> offline]
>

*CLI> sip set debug peer IMSI123456789101112

> <--- SIP read from UDP:192.168.0.102:5060 --->
> SIP/2.0 200 OK
>

*CLI> console dial 6202

> -- Executing [6202 at local:1] Macro("ALSA/default",
> "dialGSM,IMSI123456789101112") in new stack
>     -- Executing [s at macro-dialGSM:1] Dial("ALSA/default", "SIP/
> IMSI123456789101112 at 192.168.0.102") in new stack
>   == Using SIP RTP CoS mark 5
>   == Using SIP RTP CoS mark 5
> *    -- Called SIP/IMSI123456789101112 at 192.168.0.102*
> *  == Everyone is busy/congested at this time (1:0/1/0)*
>     -- Executing [s at macro-dialGSM:2] Goto("ALSA/default",
> "s-CONGESTION,1") in new stack
>     -- Goto (macro-dialGSM,s-CONGESTION,1)
>     -- Auto fallthrough, channel 'ALSA/default' status is 'CONGESTION'
>  << Hangup on console >>
>

*CLI> sip show users

> Username                   Secret           Accountcode
> Def.Context      ACL  ForcerPort
> ...
> 6001                       6001
> DLPN_DialPlan1   No   Yes
> 6000                       6000
> DLPN_DialPlan1   No   Yes
> ...
> IMSI123456789101112
> sip_external     No   Yes
>

*CLI> sip show channels

> Peer             User/ANR         Call ID          Format
> Hold     Last Message    Expiry     Peer
> 192.168.0.102    (None)           3beb558b219a72c  0x0 (nothing)
> No       Rx: OPTIONS                <guest>
> 1 active SIP dialog
>


*CLI> console dial 6202

>     -- Executing [6202 at local:1] Macro("ALSA/default",
> "dialGSM,IMSI123456789101112") in new stack
>     -- Executing [s at macro-dialGSM:1] Dial("ALSA/default", "SIP/
> IMSI123456789101112 at 192.168.0.102") in new stack
>   == Using SIP RTP CoS mark 5
>     -- Called SIP/IMSI123456789101112 at 192.168.0.102
>   == Using SIP RTP CoS mark 5
> *  == Everyone is busy/congested at this time (1:0/1/0)*
>     -- Executing [s at macro-dialGSM:2] Goto("ALSA/default",
> "s-CONGESTION,1") in new stack
>     -- Goto (macro-dialGSM,s-CONGESTION,1)
>     -- Auto fallthrough, channel 'ALSA/default' status is 'CONGESTION'
>  << Hangup on console >>
>

*CLI> sip show registry

> Host                                    dnsmgr Username       Refresh
> State                Reg.Time
> *0 SIP registrations.*
>


So I have no idea how to solve this and it would be appreciated if someone
of this mailinglist is able to help me.

Best regards and thank you for reading.

Ellen




On Wed, Jul 18, 2012 at 12:37 PM, Ellen Apolinar <
ellen.apolinar.td at googlemail.com> wrote:

> 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
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120723/5c3edee7/attachment-0001.htm>


More information about the asterisk-users mailing list