Asterisk With OpenBTS And Mobile Phone

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Hello people,

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.

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:

*CLI> sip show peers
*CLI> sip show peer 123456789101112

Asterisk log file (path: /var/log/asterisk/cdr-csv/Master.csv):

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.

3 thoughts on - Asterisk With OpenBTS And Mobile Phone

  • 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

  • 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 Music

  • –002354790f40d6153304c581ae03
    Content-Type: text/plain; charset=ISO-8859-1

    Hey mailinglist,

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

    When I start Asterisk, I do the following:

    Perhaps this will help you:

    *CLI> sip show peers

    *CLI> sip set debug peer IMSI123456789101112

    *CLI> console dial 6202

    *CLI> sip show users

    *CLI> sip show channels

    *CLI> console dial 6202

    *CLI> sip show registry

    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

    –002354790f40d6153304c581ae03
    Content-Type: text/html; charset=ISO-8859-1
    Content-Transfer-Encoding: quoted-printable

    Hey mailinglist,

    my problem still exists and I need a little bit hel=
    p.

    When I start Asterisk, I do the following:

    asterisk -rvvvvv
    originate SIP/IMSI123456789101112 appl=
    ication MusicOnHold

    =A0
    Perhaps this will help yo=
    u:

    =A0*CLI> sip show peers

    Name/username=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Host=
    =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Dyn Forcerport ACL Port=A0=A0=A0=A0 Status=
    =A0=A0=A0=A0
    6000/6000=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0 192.168.0.102=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0=A0=A0=A0=A0=A0=A0 D=A0=A0 N=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 5061=
    =A0=A0=A0=A0 Unmonitored

    6001/6001=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 192.168.0.102=
    =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0 D=A0=A0 N=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 5061=A0=A0=A0=A0 Unmon=
    itored

    IMSI123456789101112=A0=A0=A0=A0=A0=A0=A0 192.168.0.102=A0=
    =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0=A0=A0=A0 N=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 5060=A0=A0=A0=A0 OK (=
    1 ms)=A0

    36 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 33 offl=
    ine]

    *CLI> sip set debug peer IMSI12345678910=
    1112

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

    — Executing [6202@local:1] Macro("ALSA/default"=
    , "dialGSM,IMSI123456789101112") in new stack
    =A0=A0=A0 — Exe=
    cuting [s@macro-dialGSM:1] Dial("ALSA/default", "SIP/IMSI1234567=
    89101112@192.168.0.102") in new stack

    =A0 =3D=3D Using SIP RTP CoS mark 5
    =A0 =3D=3D Using SIP RTP CoS mark 5< = br>=A0=A0=A0 — Called SIP/IMSI123456789101112@192.168.0.102
    =
    =A0 =3D=3D Everyone is busy/congested at this time (1:0/1/0)

    =A0=A0=A0 — Executing [s@macro-dialGSM:2] Goto("ALSA/default", &=
    quot;s-CONGESTION,1") in new stack
    =A0=A0=A0 — Goto (macro-dialGSM=
    ,s-CONGESTION,1)
    =A0=A0=A0 — Auto fallthrough, channel 'ALSA/defaul=
    t' status is 'CONGESTION'

    =A0<< Hangup on console >>

    *CLI>=
    sip show users

    Username=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 S=
    ecret=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Accountcode=A0=A0=A0=A0=A0 Def.Context=
    =A0=A0=A0=A0=A0 ACL=A0 ForcerPort

    …=A0=A0=A0=A0
    6001=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0=A0=A0=A0 6001=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 DLPN_DialPlan1=A0=A0 No=A0=A0 Yes=A0=A0=
    =A0=A0=A0=A0
    6000=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0=A0=A0 6000=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 DLPN_DialPlan1=A0=A0 No=A0=A0 Yes=A0=A0=A0=
    =A0=A0=A0
    …=A0=A0=A0
    IMSI123456789101112=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0=A0=A0=A0=A0=A0=A0 sip_external=A0=A0=A0=A0 No=A0=A0 Yes

    *CLI> sip show channels

    Peer=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0=A0=A0 User/ANR=A0=A0=A0=A0=A0=A0=A0=A0 Call ID=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0 Format=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Hold=A0=A0=A0=A0 Last Message=
    =A0=A0=A0 Expiry=A0=A0=A0=A0 Peer=A0=A0=A0=A0=A0

    192.168.0.102=A0=A0=A0 (None)=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 3beb558b219a72c=
    =A0 0x0 (nothing)=A0=A0=A0 No=A0=A0=A0=A0=A0=A0 Rx: OPTIONS=A0=A0=A0=A0=A0=
    =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 <guest>=A0=A0
    1 active SIP dialog< = br>

    *CLI> console dial 6202

    =A0=A0=A0 — Executing [6202@local:1] Macro("ALSA/def=
    ault", "dialGSM,IMSI123456789101112") in new stack
    =A0=A0=
    =A0 — Executing [s@macro-dialGSM:1] Dial("ALSA/default", "S=
    IP/I=
    MSI123456789101112@192.168.0.102") in new stack

    =A0 =3D=3D Using SIP RTP CoS mark 5
    =A0=A0=A0 — Called SIP/IMSI123456789101=
    112@192.168.0.102
    =A0 =3D=3D Using SIP RTP CoS mark 5
    =A0 =3D=
    =3D Everyone is busy/congested at this time (1:0/1/0)

    =A0=A0=A0 — Executing [s@macro-dialGSM:2] Goto("ALSA/default", &=
    quot;s-CONGESTION,1") in new stack
    =A0=A0=A0 — Goto (macro-dialGSM=
    ,s-CONGESTION,1)
    =A0=A0=A0 — Auto fallthrough, channel 'ALSA/defaul=
    t' status is 'CONGESTION'

    =A0<< Hangup on console >>

    *CLI>=
    sip show registry

    Host=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 dnsmgr Username=A0=A0=
    =A0=A0=A0=A0 Refresh State=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Reg=
    .Time=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0

    0 SIP registrations.

    So I have no ide=
    a how to solve this and it would be appreciated if someone of this mailingl=
    ist is able to help me.

    Best regards and thank you for reading.

    Ellen

    On Wed, Jul 18, 201=
    2 at 12:37 PM, Ellen Apolinar <ellen.apolinar.td@googlemail=
    .com
    >
    wrote:

    x #ccc solid;padding-left:1ex">Hey Ioan,

    thanks for your answer.

    =A0 =3D=3D Using SIP RTP CoS mark 5

    =A0=A0=
    =A0 — 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: Autodestru=
    ct on dialog '446588d34c8b0e2d1920fec416ef0b5d@192.168.0.=
    102:5060' with owner in place (Method: INVITE)

    *CLI> sip show peers

    Name/username=A0=A0=A0=A0=
    =A0=A0=A0=A0=A0=A0=A0=A0=A0 Host=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Dyn Forcerp=
    ort ACL Port=A0=A0=A0=A0 Status=A0=A0=A0=A0

    123456789101112/6202=A0=A0=A0=A0=A0=A0 192.168.0.102=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 N=
    =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 5060=A0=A0=A0=A0 OK (1 ms)=A0

    6001/6001=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 192.168.0.102=
    =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0 D=A0=A0 N=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 5061=A0=A0=A0=A0 Unmon=
    itored

    *CLI> sip show channels

    Peer=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 User/ANR=A0=A0=A0=
    =A0=A0=A0=A0=A0 Call ID=A0=A0=A0=A0=A0=A0=A0=A0=A0 Format=A0=A0=A0=A0=A0=A0=
    =A0=A0=A0=A0 Hold=A0=A0=A0=A0 Last Message=A0=A0=A0 Expiry=A0=A0=A0=A0 Peer=
    =A0=A0=A0=A0=A0
    192.168.0.102=A0=A0=A0 (None)=A0=A0=A0=A0=A0=A0=A0=A0=
    =A0=A0 2dab9ef669bc9a4=A0 0x0 (nothing)=A0=A0=A0 No=A0=A0=A0=A0=A0=A0 Rx: O=
    PTIONS=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 <guest>=A0=A0

    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 s=
    o 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.
    E=
    llen

    On Fri, Jul 13, 2012 at 4:06 PM, Ioan In=
    dreias <indreias@gmail.com> wrote:

    On Thu, Jul 12, 2012 at 3:55 PM, E=
    llen Apolinar
    <e=
    llen.apolinar.td@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 OpenBS=
    C
    > is working with Asterisk successfully (OpenBSC is another project).
    > Perhaps you can help me because I think it is an issue with Asterisk.< = br>
    >
    >
    > sip.conf:
    >>
    >> ;SIP-Phones (Twinkle)
    >> [user1]
    >> callerid =3D 6000
    >> username =3D 6000
    >> secret =3D 6000
    >> canreinvite =3D no
    >> type =3D friend
    >> context =3D phones
    >> allow =3D all
    >> host =3D dynamic
    >> dtmfmode =3D info
    >>
    >> [user2]
    >> callerid =3D 6001
    >> username =3D 6001
    >> secret =3D 6001
    >> canreinvite =3D no
    >> type =3D friend
    >> context =3D phones
    >> allow =3D all
    >> host =3D dynamic
    >> dtmfmode =3D info
    >>
    >> ; Mobile phone
    >> [123456789101112]
    >> callerid =3D 6201
    >> username =3D 6201
    >> secret =3D 6201
    >> canreinvite =3D no
    >> type =3D friend
    >> context =3D sip_external
    >> ;context =3D open-bts
    >> disallow =3D all
    >> allow =3D gsm
    >> host =3D 192.168.0.102
    >> domain =3D 192.168.0.102
    >> dtmfmode =3D info
    >
    >
    > extensions.conf
    >>
    >> [internal]
    >> exten =3D> s,1,Verbose(1|Echo test application)
    >> exten =3D> s,n,Echo()
    >> exten =3D> s,n,Hangup()
    >> exten =3D> 6000,1,Verbose(1|Extension 6000)
    >> exten =3D> 6000,n,Dial(SIP/user1,30)
    >> exten =3D> 6000,n,Hangup()
    >> exten =3D> 6001,1,Verbose(1|Extension 6001)
    >> exten =3D> 6001,n,Dial(SIP/user2,30)
    >> exten =3D> 6001,n,Hangup()
    >>
    >> [phones]
    >> include =3D> internal
    >> include =3D> default
    >>
    >> [open-bts]
    >> exten =3D> 6002,1,Playback(demo-echotest)
    >> exten =3D> 6002,n,Echo
    >> exten =3D> 6002,n,Playback(demo-echodone)
    >> exten =3D> 6002,n,HangUp
    >>
    >> [sip_external]
    >> exten =3D> 6201,1,Macro(dialGSM,123456789101112)
    >>
    >> [macro-dialGSM]
    >> exten =3D> s,1,Dial(SIP/${ARG1},20)
    >> exten =3D> s,n,Goto(s-${DIALSTATUS},1)
    >> exten =3D> s-CANCEL,1,Hangup
    >> exten =3D> s-NOANSWER,1,Hangup
    >> exten =3D> s-BUSY,1,Busy(30)
    >> exten =3D> s-CONGESTION,1,Congestion (30)
    >> exten =3D> s-CHANUNAVAIL,1,Read(extension_digits,pbx-invalid)
    >> exten =3D> 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 (6=
    000)
    > I get following message in the CLI-window from Asterisk:
    >>
    >> =A0 =A0 =A0=3D=3D Using SIP RTP CoS mark 5
    >> =A0 =A0 =A0 =A0 — Executing [6201@DLPN_DialPlan1:1] Macro("S=
    IP/6000-00000013",
    >> "stdexten,6201,SIP/6201") in new stack
    >> =A0 =A0 =A0 =A0 — Executing [s@macro-stdexten:1] Set("SIP/60=
    00-00000013",
    >> "__DYNAMIC_FEATURES=3D") in new stack
    >> =A0 =A0 [Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:468 ast_yyer=
    ror:
    >> ast_yyerror(): =A0syntax error: syntax error, unexpected '=3D&=
    #39;, expecting $end;
    >> Input:
    >> =A0 =A0 =A0=3D 1
    >> =A0 =A0 =A0^
    >> =A0 =A0 [Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:472 ast_yyer=
    ror: If you
    >> have questions, please refer to
    >> https://wiki.asterisk.org/wiki/display/AST/Channel=
    +Variables
    >> =A0 =A0 =A0 =A0 — Executing [s@macro-stdexten:2] GotoIf("SIP=
    /6000-00000013",
    >> "?5:3") in new stack
    >> =A0 =A0 =A0 =A0 — Goto (macro-stdexten,s,3)
    >> =A0 =A0 =A0 =A0 — Executing [s@macro-stdexten:3] Dial("SIP/6=
    000-00000013",
    >> "SIP/6201,20,") in new stack
    >> =A0 =A0 [Jul 12 12:14:29] WARNING[7092]: app_dial.c:2274 dial_exec=
    _full:
    >> Unable to create channel of type 'SIP' (cause 20 – Unknown=
    )
    >> =A0 =A0 =A0 =3D=3D Everyone is busy/congested at this time (1:0/0/=
    1)
    >> =A0 =A0 =A0 =A0 — Executing [s@macro-stdexten:4] Goto("SIP/6=
    000-00000013",
    >> "s-CHANUNAVAIL,1") in new stack
    >> =A0 =A0 =A0 =A0 — Goto (macro-stdexten,s-CHANUNAVAIL,1)
    >> =A0 =A0 =A0 =A0 — Executing [s-CHANUNAVAIL@macro-stdexten:1]
    >> Goto("SIP/6000-00000013", "s-NOANSWER,1") in n=
    ew stack
    >> =A0 =A0 =A0 =A0 — Goto (macro-stdexten,s-NOANSWER,1)
    >> =A0 =A0 =A0 =A0 — Executing [s-NOANSWER@macro-stdexten:1]
    >> VoiceMail("SIP/6000-00000013", "6201,u") in ne=
    w stack
    >> =A0 =A0 =A0 =A0 — <SIP/6000-00000013> Playing 'vm-thepe=
    rson.gsm' (language 'en')
    >> =A0 =A0 =A0 =A0 — <SIP/6000-00000013> Playing 'digits/6=
    .gsm' (language 'en')
    >> =A0 =A0 =A0 =A0 — <SIP/6000-00000013> Playing 'digits/2=
    .gsm' (language 'en')
    >> =A0 =A0 =A0 =A0 — <SIP/6000-00000013> Playing 'digits/0=
    .gsm' (language 'en')
    >> =A0 =A0 =A0 =A0 — <SIP/6000-00000013> Playing 'digits/1=
    .gsm' (language 'en')
    >> =A0 =A0 =A0 =A0 — <SIP/6000-00000013> Playing 'vm-isuna=
    vail.gsm' (language 'en')
    >> =A0 =A0 =A0 =A0 — <SIP/6000-00000013> Playing 'vm-intro=
    .gsm' (language 'en')
    >> =A0 =A0 =A0 =3D=3D Spawn extension (macro-stdexten, s-NOANSWER, 1)=
    exited non-zero
    >> on 'SIP/6000-00000013' in macro 'stdexten'
    >> =A0 =A0 =A0 =3D=3D Spawn extension (DLPN_DialPlan1, 6201, 1) exite=
    d non-zero on
    >> 'SIP/6000-00000013'
    >
    >
    >
    > *CLI> sip show peers
    >>
    >> =A0 =A0 Name/username =A0 =A0 =A0 =A0 =A0 =A0 =A0Host =A0 =A0 =A0 =
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0Dyn
    >> Forcerport ACL Port =A0 =A0 Status
    >> =A0 =A0 123456789101112/6201 =A0 =A0 =A0 192.168.0.102
    >> N =A0 =A0 =A0 =A0 =A0 =A0 5060 =A0 =A0 Unmonitored
    >> =A0 =A0 6000/6000 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0192.168.0.102=
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0D
    >> N =A0 =A0 =A0 =A0 =A0 =A0 5061 =A0 =A0 Unmonitored
    >> =A0 =A0 6001/6001 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0192.168.0.102=
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0D
    >> N =A0 =A0 =A0 =A0 =A0 =A0 5061 =A0 =A0 Unmonitored
    >> =A0 =A0 (…)
    >> =A0 =A0 user1/6000 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 (Unspecified) =
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0D
    >> N =A0 =A0 =A0 =A0 =A0 =A0 0 =A0 =A0 =A0 =A0Unmonitored
    >> =A0 =A0 user2/6001 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 (Unspecified) =
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0D
    >> N =A0 =A0 =A0 =A0 =A0 =A0 0 =A0 =A0 =A0 =A0Unmonitored
    >
    >
    > *CLI> sip show peer 123456789101112
    >>
    >> =A0 =A0 =A0 * Name =A0 =A0 =A0 : 123456789101112
    >> =A0 =A0 =A0 Secret =A0 =A0 =A0 : <Set>
    >> =A0 =A0 =A0 MD5Secret =A0 =A0: <Not set>
    >> =A0 =A0 =A0 Remote Secret: <Not set>
    >> =A0 =A0 =A0 Context =A0 =A0 =A0: sip_external
    >> =A0 =A0 =A0 Subscr.Cont. : device-hints
    >> =A0 =A0 =A0 Language =A0 =A0 :
    >> =A0 =A0 =A0 AMA flags =A0 =A0: Unknown
    >> =A0 =A0 =A0 Transfer mode: open
    >> =A0 =A0 =A0 CallingPres =A0: Presentation Allowed, Not Screened
    >> =A0 =A0 =A0 Pickupgroup =A0:
    >> =A0 =A0 =A0 MOH Suggest =A0:
    >> =A0 =A0 =A0 Mailbox =A0 =A0 =A0:
    >> =A0 =A0 =A0 VM Extension : asterisk
    >> =A0 =A0 =A0 LastMsgsSent : 32767/65535
    >> =A0 =A0 =A0 Call limit =A0 : 0
    >> =A0 =A0 =A0 Max forwards : 0
    >> =A0 =A0 =A0 Dynamic =A0 =A0 =A0: No
    >> =A0 =A0 =A0 Callerid =A0 =A0 : "" <6201>
    >> =A0 =A0 =A0 MaxCallBR =A0 =A0: 384 kbps
    >> =A0 =A0 =A0 Expire =A0 =A0 =A0 : -1
    >> =A0 =A0 =A0 Insecure =A0 =A0 : no
    >> =A0 =A0 =A0 Force rport =A0: Yes
    >> =A0 =A0 =A0 ACL =A0 =A0 =A0 =A0 =A0: No
    >> =A0 =A0 =A0 DirectMedACL : No
    >> =A0 =A0 =A0 T.38 support : No
    >> =A0 =A0 =A0 T.38 EC mode : Unknown
    >> =A0 =A0 =A0 T.38 MaxDtgrm: -1
    >> =A0 =A0 =A0 DirectMedia =A0: No
    >> =A0 =A0 =A0 PromiscRedir : No
    >> =A0 =A0 =A0 User=3DPhone =A0 : No
    >> =A0 =A0 =A0 Video Support: No
    >> =A0 =A0 =A0 Text Support : No
    >> =A0 =A0 =A0 Ign SDP ver =A0: No
    >> =A0 =A0 =A0 Trust RPID =A0 : No
    >> =A0 =A0 =A0 Send RPID =A0 =A0: No
    >> =A0 =A0 =A0 Subscriptions: Yes
    >> =A0 =A0 =A0 Overlap dial : No
    >> =A0 =A0 =A0 DTMFmode =A0 =A0 : info
    >> =A0 =A0 =A0 Timer T1 =A0 =A0 : 500
    >> =A0 =A0 =A0 Timer B =A0 =A0 =A0: 32000
    >> =A0 =A0 =A0 ToHost =A0 =A0 =A0 : 192.168.0.102
    >> =A0 =A0 =A0 Addr->IP =A0 =A0 : 192.168.0.102:5060
    >> =A0 =A0 =A0 Defaddr->IP =A0: (null)
    >> =A0 =A0 =A0 Prim.Transp. : UDP
    >> =A0 =A0 =A0 Allowed.Trsp : UDP
    >> =A0 =A0 =A0 Def. Username: 6201
    >> =A0 =A0 =A0 SIP Options =A0: (none)
    >> =A0 =A0 =A0 Codecs =A0 =A0 =A0 : 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|si=
    ren14|testlaw|g719)
    >> =A0 =A0 =A0 Codec Order =A0: (none)
    >> =A0 =A0 =A0 Auto-Framing : =A0No
    >> =A0 =A0 =A0 Status =A0 =A0 =A0 : Unmonitored
    >> =A0 =A0 =A0 Useragent =A0 =A0:
    >> =A0 =A0 =A0 Reg. Contact :
    >> =A0 =A0 =A0 Qualify Freq : 60000 ms
    >> =A0 =A0 =A0 Sess-Timers =A0: Accept
    >> =A0 =A0 =A0 Sess-Refresh : uas
    >> =A0 =A0 =A0 Sess-Expires : 1800 secs
    >> =A0 =A0 =A0 Min-Sess =A0 =A0 : 90 secs
    >> =A0 =A0 =A0 RTP Engine =A0 : asterisk
    >> =A0 =A0 =A0 Parkinglot =A0 :
    >> =A0 =A0 =A0 Use Reason =A0 : No
    >> =A0 =A0 =A0 Encryption =A0 : No
    >
    >
    > Asterisk log file (path: /var/log/asterisk/cdr-csv/Master.csv):
    >>
    >> =A0 =A0 "","6000","6201","DLPN_=
    DialPlan1","""6000""
    >> <6000>","SIP/6000-00000013","",&quo=
    t;VoiceMail","6201,u","2012-07-12
    >> 10:14:29","2012-07-12 10:14:29","2012-07-12
    >
    >
    >
    >
    > If you need more informations write me and I will give you. It would b=
    e very
    > appreciated if some of you can help me or has an idea how I can fix th=
    is
    > erorr.
    >
    > Best regards and thanks for helping.
    > Ellen
    >

    > —
    > _____________________________________________________________________< = br>
    > — Bandwidth and Colocation Provided by http://www.api-digital.com
    > New to Asterisk? Join us for a live introductory webinar every Thurs:< = br>
    > =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0http://www.asterisk.org/hello
    >
    > asterisk-users mailing list
    > To UNSUBSCRIBE or update options visit:
    > =A0 =A0http://lists.digium.com/mailman/listinfo/asterisk-us=
    ers

    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&qu=
    ot;
    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:
    =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0http://www.asterisk.org/hello

    asterisk-users mailing list
    To UNSUBSCRIBE or update options visit:
    =A0 =A0http://lists.digium.com/mailman/listinfo/asterisk-users

    –002354790f40d6153304c581ae03