Asterisk With OpenBTS And Mobile Phone

Ellen Apolinar July 12, 2012 Asterisk Users 3 Comments

Tags: asterisk, call, csv, nokia mobile phone, phone, sip, soft phones

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.

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3 Responses to “Asterisk With OpenBTS And Mobile Phone”

Ioan Indreias said:
Jul 13, 12 at 9:06 am
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
Ellen Apolinar said:
Jul 18, 12 at 5:38 am
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
Ellen Apolinar said:
Jul 23, 12 at 11:24 am
–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:

=3D"margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding=
-left:1ex" class=3D"gmail_quote">
asterisk -rvvvvv
originate SIP/IMSI123456789101112 appl=
ication MusicOnHold

=A0
Perhaps this will help yo=
u:

=A0*CLI> sip show peers

x 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class=
=3D"gmail_quote">
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

d rgb(204,204,204);padding-left:1ex" class=3D"gmail_quote">
<— SIP read from UDP: 5060" target=3D"_blank">192.168.0.102:5060 —>
SIP/2.0 200 OK
"margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-le=
ft:1ex" class=3D"gmail_quote"/>

– 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/ =3D"mailto:IMSI123456789101112@192.168.0.102" target=3D"_blank">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/ .0.102" target=3D"_blank">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

t:1px solid rgb(204,204,204);padding-left:1ex" class=3D"gmail_quote"> size=3D"1">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

margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-lef=
t:1ex" class=3D"gmail_quote">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 0×0 (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

yle=3D"margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padd=
ing-left:1ex" class=3D"gmail_quote">
=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/ ailto:IMSI123456789101112@192.168.0.102" target=3D"_blank">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

-left:1px solid rgb(204,204,204);padding-left:1ex" class=3D"gmail_quote"> ont size=3D"1">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 < .apolinar.td@googlemail.com" target=3D"_blank">ellen.apolinar.td@googlemail=
.com> wrote:

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

thanks for your answer.
g.

My new situation:

*CLI> originate SIP/123456789101112 applica=
tion MusicOnHold

ft:1px solid rgb(204,204,204);padding-left:1ex" class=3D"gmail_quote">

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

=A0=A0=
=A0 — Got SIP response 482 "Loop Detected" back from http://192.168.0.102:5060" target=3D"_blank">192.168.0.102:5060
[Jul=
18 10:38:27] WARNING[4615]: chan_sip.c:3873 __sip_autodestruct: Autodestru=
ct on dialog ' 8.0.102:5060" target=3D"_blank">446588d34c8b0e2d1920fec416ef0b5d@192.168.0.=
102:5060' with owner in place (Method: INVITE)

*CLI> sip show peers

gin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1=
ex" class=3D"gmail_quote">
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
=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

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
ckquote style=3D"margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204=
,204);padding-left:1ex" class=3D"gmail_quote">

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 0×0 (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 < =3D"_blank">indreias@gmail.com> wrote:

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

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) r>
>> 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
>> ables" target=3D"_blank">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 Callgroup =A0 =A0:
>> =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 : 5060" target=3D"_blank">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
>> 10:14:35",6,6,"ANSWERED","DOCUMENTATION",=
"1342088069.31",""
>
>
>
>
> 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 tal.com" target=3D"_blank">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 =A0 o" target=3D"_blank">http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> =A0 =A0 ers" target=3D"_blank">http://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 om" target=3D"_blank">http://www.api-digital.com –
New to Asterisk? Join us for a live introductory webinar every Thurs:
=A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 rget=3D"_blank">http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
=A0 =A0 target=3D"_blank">http://lists.digium.com/mailman/listinfo/asterisk-users

–002354790f40d6153304c581ae03