Re: [asterisk-users] Asterisk with OpenBTS and mobile phone
Hey mailinglist,
my problem still exists and I need a little bit help.
When I start Asterisk, I do the following:
> asterisk -rv
> originate SIP/IMSI123456789101112 application MusicOnHold
>
Perhaps this will help you:
*CLI> sip show peers
> Name/username HostDyn
> Forcerport ACL Port Status
> 6000/6000 192.168.0.102D
> N 5061 Unmonitored
> 6001/6001 192.168.0.102D
> N 5061 Unmonitored
> ...
> IMSI123456789101112192.168.0.102
> 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@local:1] Macro("ALSA/default",
> "dialGSM,IMSI123456789101112") in new stack
> -- Executing [s@macro-dialGSM:1] Dial("ALSA/default", "SIP/
> IMSI123456789101112@192.168.0.102") in new stack
> == Using SIP RTP CoS mark 5
> == Using SIP RTP CoS mark 5
> *-- Called SIP/IMSI123456789101112@192.168.0.102*
> * == Everyone is busy/congested at this time (1:0/1/0)*
> -- Executing [s@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 MessageExpiry Peer
> 192.168.0.102(None) 3beb558b219a72c 0x0 (nothing)
> No Rx: OPTIONS
> 1 active SIP dialog
>
*CLI> console dial 6202
> -- Executing [6202@local:1] Macro("ALSA/default",
> "dialGSM,IMSI123456789101112") in new stack
> -- Executing [s@macro-dialGSM:1] Dial("ALSA/default", "SIP/
> IMSI123456789101112@192.168.0.102") in new stack
> == Using SIP RTP CoS mark 5
> -- Called SIP/IMSI123456789101112@192.168.0.102
> == Using SIP RTP CoS mark 5
> * == Everyone is busy/congested at this time (1:0/1/0)*
> -- Executing [s@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
> Hostdnsmgr Username Refresh
> StateReg.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...@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@192.168.0.102:5060' with owner in place
>> (Method: INVITE)
>>
>
> *CLI> sip show peers
>
>> Name/username HostDyn
>> Forcerport ACL Port Status
>> 123456789101112/6202 192.168.0.102
>> N 5060 OK (1 ms)
>> 6000/6000 192.168.0.102D
>> N 5061 Unmonitored
>> 6001/6001 192.168.0.102D
>> N 5061 Unmonitored
>>
>
> *CLI> sip show channels
>
>> Peer User/ANR Call ID Format
>> Hold Last MessageExpiry Peer
>> 192.168.0.102(None) 2dab9ef669bc9a4 0x0 (nothing)
>> No Rx: OPTIONS
>> 1 active SI
Re: [asterisk-users] Asterisk with OpenBTS and mobile phone
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@192.168.0.102:5060' with owner in place
> (Method: INVITE)
>
*CLI> sip show peers
> Name/username HostDyn
> Forcerport ACL Port Status
> 123456789101112/6202 192.168.0.102
> N 5060 OK (1 ms)
> 6000/6000 192.168.0.102D
> N 5061 Unmonitored
> 6001/6001 192.168.0.102D
> N 5061 Unmonitored
>
*CLI> sip show channels
> Peer User/ANR Call ID Format
> Hold Last MessageExpiry Peer
> 192.168.0.102(None) 2dab9ef669bc9a4 0x0 (nothing)
> No Rx: OPTIONS
> 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 wrote:
> On Thu, Jul 12, 2012 at 3:55 PM, Ellen Apolinar
> 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)
[asterisk-users] Asterisk with OpenBTS and mobile phone
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@DLPN_DialPlan1:1] Macro("SIP/6000-0013",
> "stdexten,6201,SIP/6201") in new stack
> -- Executing [s@macro-stdexten:1] Set("SIP/6000-0013",
> "__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@macro-stdexten:2] GotoIf("SIP/6000-0013",
> "?5:3") in new stack
> -- Goto (macro-stdexten,s,3)
> -- Executing [s@macro-stdexten:3] Dial("SIP/6000-0013",
> "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@macro-stdexten:4] Goto("SIP/6000-0013",
> "s-CHANUNAVAIL,1") in new stack
> -- Goto (macro-stdexten,s-CHANUNAVAIL,1)
> -- Executing [s-CHANUNAVAIL@macro-stdexten:1]
> Goto("SIP/6000-0013", "s-NOANSWER,1") in new stack
> -- Goto (macro-stdexten,s-NOANSWER,1)
> -- Executing [s-NOANSWER@macro-stdexten:1]
> VoiceMail("SIP/6000-0013", "6201,u") in new stack
> -- Playing 'vm-theperson.gsm' (language 'en')
> -- Playing 'digits/6.gsm' (language 'en')
> -- Playing 'digits/2.gsm' (language 'en')
> -- Playing 'digits/0.gsm' (language 'en')
> -- Playing 'digits/1.gsm' (language 'en')
> -- Playing 'vm-isunavail.gsm' (language 'en')
> -- Playing 'vm-intro.gsm' (language 'en')
> == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero
> on 'SIP/6000-0013' in macro 'stdexten'
> == Spawn extension (DLPN_DialPlan1, 6201, 1) exited non-zero on
> 'SIP/6000-0013'
>
*CLI> sip show peers
> Name/username HostDyn
> Forcerport ACL Port Status
> *123456789101112/6201
> 192.168.0.102N 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 0Unmonitored
> user2/6001 (Unspecified)
> D N
[asterisk-users] Asterisk with LCR -> chan_lcr needed?
Hello, I short question: I want to connect Asterisk to OpenBSC with mISDN, mISDNuser and LCR. Do I need chan_lcr? I have: Asterisk 1.8 mISDN .v2 integrated in Kernel 3.0.22 mISDNuser lcr 1.7 HFC-E1 Evaluation board from cologne chip I tried to configure Asterisk with <./configure --prefix=/usr/src/lcr --with-gsm-bs> and it runs without errors, But make and make install didn't run fine. I have written my problems in this forum. It is german but you can see my error messages in the last post: http://www.ip-phone-forum.de/showthread.php?t=247932&p=1839572#post1839572 It would be very appreciate is someone can answer my question because the most HowTo's in the internet are old so I don't know if I can use them for my case. Best regards 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
