Asterisk With OpenBTS And Mobile Phone

Report
Question
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.
Asterisk Users 3.2 years ago 3 Answers

Answers ( 3 )

  1. Ioan Indreias
    +1
    July 13, 2012 at 09:06 am
    Reply

    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

  2. Ellen Apolinar
    +1
    July 18, 2012 at 05:38 am
    Reply

    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

  3. Ellen Apolinar
    +1
    July 23, 2012 at 11:24 am
    Reply

    --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:


    Hey Ioan,

    thanks for your answer.


    My new situation:

    *CLI> originate SIP/123456789101112 applica= tion MusicOnHold


    =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


    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 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

 Prev question

Next question