Hey mailinglist,<br><br>my problem still exists and I need a little bit help.<br><br>When I start Asterisk, I do the following:<br><blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote">
<font size="1">asterisk -rvvvvv<br>originate SIP/IMSI123456789101112 application MusicOnHold</font><br></blockquote> <br>Perhaps this will help you:<br><br> *CLI> sip show peers <br><blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote">
<font size="1">Name/username Host Dyn Forcerport ACL Port Status <br>6000/6000 192.168.0.102 D N 5061 Unmonitored <br>
6001/6001 192.168.0.102 D N 5061 Unmonitored <br>...<br>IMSI123456789101112 192.168.0.102 N 5060 OK (1 ms) <br>
36 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 33 offline]<br></font></blockquote><br>*CLI> sip set debug peer IMSI123456789101112 <br><blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote">
<font size="1"><--- SIP read from UDP:<a href="http://192.168.0.102:5060" target="_blank">192.168.0.102:5060</a> ---><br>SIP/2.0 200 OK<br></font></blockquote><br>*CLI> console dial 6202<br><blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote">
<font size="1">-- Executing [6202@local:1] Macro("ALSA/default", "dialGSM,IMSI123456789101112") in new stack<br> -- Executing [s@macro-dialGSM:1] Dial("ALSA/default", "SIP/<a href="mailto:IMSI123456789101112@192.168.0.102" target="_blank">IMSI123456789101112@192.168.0.102</a>") in new stack<br>
== Using SIP RTP CoS mark 5<br> == Using SIP RTP CoS mark 5<br><b> -- Called SIP/<a href="mailto:IMSI123456789101112@192.168.0.102" target="_blank">IMSI123456789101112@192.168.0.102</a></b><br><b> == Everyone is busy/congested at this time (1:0/1/0)</b><br>
-- Executing [s@macro-dialGSM:2] Goto("ALSA/default", "s-CONGESTION,1") in new stack<br> -- Goto (macro-dialGSM,s-CONGESTION,1)<br> -- Auto fallthrough, channel 'ALSA/default' status is 'CONGESTION'<br>
<< Hangup on console >> <br></font></blockquote><br>*CLI> sip show users<br><blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote"><font size="1">Username Secret Accountcode Def.Context ACL ForcerPort<br>
... <br>6001 6001 DLPN_DialPlan1 No Yes <br>6000 6000 DLPN_DialPlan1 No Yes <br>... <br>IMSI123456789101112 sip_external No Yes <br>
</font></blockquote><br>*CLI> sip show channels<br><blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote"><font size="1">Peer User/ANR Call ID Format Hold Last Message Expiry Peer <br>
192.168.0.102 (None) 3beb558b219a72c 0x0 (nothing) No Rx: OPTIONS <guest> <br>1 active SIP dialog<br></font></blockquote><br><br>*CLI> console dial 6202<br><blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote">
<font size="1"> -- Executing [6202@local:1] Macro("ALSA/default", "dialGSM,IMSI123456789101112") in new stack<br> -- Executing [s@macro-dialGSM:1] Dial("ALSA/default", "SIP/<a href="mailto:IMSI123456789101112@192.168.0.102" target="_blank">IMSI123456789101112@192.168.0.102</a>") in new stack<br>
== Using SIP RTP CoS mark 5<br> -- Called SIP/<a href="mailto:IMSI123456789101112@192.168.0.102" target="_blank">IMSI123456789101112@192.168.0.102</a><br> == Using SIP RTP CoS mark 5<br><b> == Everyone is busy/congested at this time (1:0/1/0)</b><br>
-- Executing [s@macro-dialGSM:2] Goto("ALSA/default", "s-CONGESTION,1") in new stack<br> -- Goto (macro-dialGSM,s-CONGESTION,1)<br> -- Auto fallthrough, channel 'ALSA/default' status is 'CONGESTION'<br>
<< Hangup on console >> <br></font></blockquote><br>*CLI> sip show registry <br><blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote"><font size="1">Host dnsmgr Username Refresh State Reg.Time <br>
<b>0 SIP registrations.</b><br></font></blockquote><br><br>So I have no idea how to solve this and it would be appreciated if someone of this mailinglist is able to help me. <br><br>Best regards and thank you for reading.<br>
<br>Ellen<br><br><br><br><br><div class="gmail_quote">On Wed, Jul 18, 2012 at 12:37 PM, Ellen Apolinar <span dir="ltr"><<a href="mailto:ellen.apolinar.td@googlemail.com" target="_blank">ellen.apolinar.td@googlemail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hey Ioan, <br><br>thanks for your answer.<br><br>It helped a little bit but I have no idea what exactly could work wrong.<br>
<br>My new situation:<br><br>*CLI> originate SIP/123456789101112 application MusicOnHold<br><blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote">
<font size="1"><div> == Using SIP RTP CoS mark 5<br></div> -- Got SIP response 482 "Loop Detected" back from <a href="http://192.168.0.102:5060" target="_blank">192.168.0.102:5060</a><br>[Jul 18 10:38:27] WARNING[4615]: chan_sip.c:3873 __sip_autodestruct: Autodestruct on dialog '<a href="http://446588d34c8b0e2d1920fec416ef0b5d@192.168.0.102:5060" target="_blank">446588d34c8b0e2d1920fec416ef0b5d@192.168.0.102:5060</a>' with owner in place (Method: INVITE)<br>
</font></blockquote><br>*CLI> sip show peers<br><blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote"><font size="1"><div>Name/username Host Dyn Forcerport ACL Port Status <br>
</div>
123456789101112/6202 192.168.0.102 N 5060 OK (1 ms) <br><div>6000/6000 192.168.0.102 D N 5061 Unmonitored <br>
6001/6001 192.168.0.102 D N 5061 Unmonitored <br></div></font></blockquote><br>*CLI> sip show channels<br><blockquote style="margin:0px 0px 0px 6.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex" class="gmail_quote">
<font size="1">Peer User/ANR Call ID Format Hold Last Message Expiry Peer <br>192.168.0.102 (None) 2dab9ef669bc9a4 0x0 (nothing) No Rx: OPTIONS <guest> <br>
1 active SIP dialog<br></font></blockquote><br>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. <br>
<br>asterisk -rx doesn't work. <br><br>What do you think is wrong with my extensions.conf?<br><br>Best regards.<span><font color="#888888"><br>Ellen</font></span><div><div><br>
<br><br><div class="gmail_quote">On Fri, Jul 13, 2012 at 4:06 PM, Ioan Indreias <span dir="ltr"><<a href="mailto:indreias@gmail.com" target="_blank">indreias@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>On Thu, Jul 12, 2012 at 3:55 PM, Ellen Apolinar<br>
<<a href="mailto:ellen.apolinar.td@googlemail.com" target="_blank">ellen.apolinar.td@googlemail.com</a>> wrote:<br>
> Hello mailinglist,<br>
><br>
> I want to connect Asterisk with OpenBTS and make a call with a mobile phone.<br>
><br>
> I use:<br>
> Ubuntu 11.10 + Kernel 3.0.22<br>
> GnuRadio 3.3.0<br>
> Asterisk 1.8.13<br>
> OpenBTS 2.8<br>
> Nokia Mobile Phone<br>
><br>
> OpenBTS works and I can send sms from the OpenBTS server to the<br>
> mobile phone. What I also need is a call between Asterisk and OpenBTS.<br>
><br>
> I have also two soft phones which works with Asterisk. And also OpenBSC<br>
> is working with Asterisk successfully (OpenBSC is another project).<br>
><br>
> Perhaps you can help me because I think it is an issue with Asterisk.<br>
><br>
><br>
> sip.conf:<br>
>><br>
>> ;SIP-Phones (Twinkle)<br>
>> [user1]<br>
>> callerid = 6000<br>
>> username = 6000<br>
>> secret = 6000<br>
>> canreinvite = no<br>
>> type = friend<br>
>> context = phones<br>
>> allow = all<br>
>> host = dynamic<br>
>> dtmfmode = info<br>
>><br>
>> [user2]<br>
>> callerid = 6001<br>
>> username = 6001<br>
>> secret = 6001<br>
>> canreinvite = no<br>
>> type = friend<br>
>> context = phones<br>
>> allow = all<br>
>> host = dynamic<br>
>> dtmfmode = info<br>
>><br>
>> ; Mobile phone<br>
>> [123456789101112]<br>
>> callerid = 6201<br>
>> username = 6201<br>
>> secret = 6201<br>
>> canreinvite = no<br>
>> type = friend<br>
>> context = sip_external<br>
>> ;context = open-bts<br>
>> disallow = all<br>
>> allow = gsm<br>
>> host = 192.168.0.102<br>
>> domain = 192.168.0.102<br>
>> dtmfmode = info<br>
><br>
><br>
> extensions.conf<br>
>><br>
>> [internal]<br>
>> exten => s,1,Verbose(1|Echo test application)<br>
>> exten => s,n,Echo()<br>
>> exten => s,n,Hangup()<br>
>> exten => 6000,1,Verbose(1|Extension 6000)<br>
>> exten => 6000,n,Dial(SIP/user1,30)<br>
>> exten => 6000,n,Hangup()<br>
>> exten => 6001,1,Verbose(1|Extension 6001)<br>
>> exten => 6001,n,Dial(SIP/user2,30)<br>
>> exten => 6001,n,Hangup()<br>
>><br>
>> [phones]<br>
>> include => internal<br>
>> include => default<br>
>><br>
>> [open-bts]<br>
>> exten => 6002,1,Playback(demo-echotest)<br>
>> exten => 6002,n,Echo<br>
>> exten => 6002,n,Playback(demo-echodone)<br>
>> exten => 6002,n,HangUp<br>
>><br>
>> [sip_external]<br>
>> exten => 6201,1,Macro(dialGSM,123456789101112)<br>
>><br>
>> [macro-dialGSM]<br>
>> exten => s,1,Dial(SIP/${ARG1},20)<br>
>> exten => s,n,Goto(s-${DIALSTATUS},1)<br>
>> exten => s-CANCEL,1,Hangup<br>
>> exten => s-NOANSWER,1,Hangup<br>
>> exten => s-BUSY,1,Busy(30)<br>
>> exten => s-CONGESTION,1,Congestion (30)<br>
>> exten => s-CHANUNAVAIL,1,Read(extension_digits,pbx-invalid)<br>
>> exten => s-CHANUNAVAIL,n,GoTo(open-bts,${extension_digits},1)<br>
><br>
> I have tried both contexts, [open-bts] and [sip_external] and both don't<br>
> work<br>
><br>
><br>
> If I want to call the mobile phone (6201) with a Twinkle soft phone (6000)<br>
> I get following message in the CLI-window from Asterisk:<br>
>><br>
>> == Using SIP RTP CoS mark 5<br>
>> -- Executing [6201@DLPN_DialPlan1:1] Macro("SIP/6000-00000013",<br>
>> "stdexten,6201,SIP/6201") in new stack<br>
>> -- Executing [s@macro-stdexten:1] Set("SIP/6000-00000013",<br>
>> "__DYNAMIC_FEATURES=") in new stack<br>
>> [Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:468 ast_yyerror:<br>
>> ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end;<br>
>> Input:<br>
>> = 1<br>
>> ^<br>
>> [Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:472 ast_yyerror: If you<br>
>> have questions, please refer to<br>
>> <a href="https://wiki.asterisk.org/wiki/display/AST/Channel+Variables" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Channel+Variables</a><br>
>> -- Executing [s@macro-stdexten:2] GotoIf("SIP/6000-00000013",<br>
>> "?5:3") in new stack<br>
>> -- Goto (macro-stdexten,s,3)<br>
>> -- Executing [s@macro-stdexten:3] Dial("SIP/6000-00000013",<br>
>> "SIP/6201,20,") in new stack<br>
>> [Jul 12 12:14:29] WARNING[7092]: app_dial.c:2274 dial_exec_full:<br>
>> Unable to create channel of type 'SIP' (cause 20 - Unknown)<br>
>> == Everyone is busy/congested at this time (1:0/0/1)<br>
>> -- Executing [s@macro-stdexten:4] Goto("SIP/6000-00000013",<br>
>> "s-CHANUNAVAIL,1") in new stack<br>
>> -- Goto (macro-stdexten,s-CHANUNAVAIL,1)<br>
>> -- Executing [s-CHANUNAVAIL@macro-stdexten:1]<br>
>> Goto("SIP/6000-00000013", "s-NOANSWER,1") in new stack<br>
>> -- Goto (macro-stdexten,s-NOANSWER,1)<br>
>> -- Executing [s-NOANSWER@macro-stdexten:1]<br>
>> VoiceMail("SIP/6000-00000013", "6201,u") in new stack<br>
>> -- <SIP/6000-00000013> Playing 'vm-theperson.gsm' (language 'en')<br>
>> -- <SIP/6000-00000013> Playing 'digits/6.gsm' (language 'en')<br>
>> -- <SIP/6000-00000013> Playing 'digits/2.gsm' (language 'en')<br>
>> -- <SIP/6000-00000013> Playing 'digits/0.gsm' (language 'en')<br>
>> -- <SIP/6000-00000013> Playing 'digits/1.gsm' (language 'en')<br>
>> -- <SIP/6000-00000013> Playing 'vm-isunavail.gsm' (language 'en')<br>
>> -- <SIP/6000-00000013> Playing 'vm-intro.gsm' (language 'en')<br>
>> == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero<br>
>> on 'SIP/6000-00000013' in macro 'stdexten'<br>
>> == Spawn extension (DLPN_DialPlan1, 6201, 1) exited non-zero on<br>
>> 'SIP/6000-00000013'<br>
><br>
><br>
><br>
> *CLI> sip show peers<br>
>><br>
>> Name/username Host Dyn<br>
>> Forcerport ACL Port Status<br>
>> 123456789101112/6201 192.168.0.102<br>
>> N 5060 Unmonitored<br>
>> 6000/6000 192.168.0.102 D<br>
>> N 5061 Unmonitored<br>
>> 6001/6001 192.168.0.102 D<br>
>> N 5061 Unmonitored<br>
>> (...)<br>
>> user1/6000 (Unspecified) D<br>
>> N 0 Unmonitored<br>
>> user2/6001 (Unspecified) D<br>
>> N 0 Unmonitored<br>
><br>
><br>
> *CLI> sip show peer 123456789101112<br>
>><br>
>> * Name : 123456789101112<br>
>> Secret : <Set><br>
>> MD5Secret : <Not set><br>
>> Remote Secret: <Not set><br>
>> Context : sip_external<br>
>> Subscr.Cont. : device-hints<br>
>> Language :<br>
>> AMA flags : Unknown<br>
>> Transfer mode: open<br>
>> CallingPres : Presentation Allowed, Not Screened<br>
>> Callgroup :<br>
>> Pickupgroup :<br>
>> MOH Suggest :<br>
>> Mailbox :<br>
>> VM Extension : asterisk<br>
>> LastMsgsSent : 32767/65535<br>
>> Call limit : 0<br>
>> Max forwards : 0<br>
>> Dynamic : No<br>
>> Callerid : "" <6201><br>
>> MaxCallBR : 384 kbps<br>
>> Expire : -1<br>
>> Insecure : no<br>
>> Force rport : Yes<br>
>> ACL : No<br>
>> DirectMedACL : No<br>
>> T.38 support : No<br>
>> T.38 EC mode : Unknown<br>
>> T.38 MaxDtgrm: -1<br>
>> DirectMedia : No<br>
>> PromiscRedir : No<br>
>> User=Phone : No<br>
>> Video Support: No<br>
>> Text Support : No<br>
>> Ign SDP ver : No<br>
>> Trust RPID : No<br>
>> Send RPID : No<br>
>> Subscriptions: Yes<br>
>> Overlap dial : No<br>
>> DTMFmode : info<br>
>> Timer T1 : 500<br>
>> Timer B : 32000<br>
>> ToHost : 192.168.0.102<br>
>> Addr->IP : <a href="http://192.168.0.102:5060" target="_blank">192.168.0.102:5060</a><br>
>> Defaddr->IP : (null)<br>
>> Prim.Transp. : UDP<br>
>> Allowed.Trsp : UDP<br>
>> Def. Username: 6201<br>
>> SIP Options : (none)<br>
>> Codecs : 0x80030c7fffff<br>
>> (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)<br>
>> Codec Order : (none)<br>
>> Auto-Framing : No<br>
>> Status : Unmonitored<br>
>> Useragent :<br>
>> Reg. Contact :<br>
>> Qualify Freq : 60000 ms<br>
>> Sess-Timers : Accept<br>
>> Sess-Refresh : uas<br>
>> Sess-Expires : 1800 secs<br>
>> Min-Sess : 90 secs<br>
>> RTP Engine : asterisk<br>
>> Parkinglot :<br>
>> Use Reason : No<br>
>> Encryption : No<br>
><br>
><br>
> Asterisk log file (path: /var/log/asterisk/cdr-csv/Master.csv):<br>
>><br>
>> "","6000","6201","DLPN_DialPlan1","""6000""<br>
>> <6000>","SIP/6000-00000013","","VoiceMail","6201,u","2012-07-12<br>
>> 10:14:29","2012-07-12 10:14:29","2012-07-12<br>
>> 10:14:35",6,6,"ANSWERED","DOCUMENTATION","1342088069.31",""<br>
><br>
><br>
><br>
><br>
> If you need more informations write me and I will give you. It would be very<br>
> appreciated if some of you can help me or has an idea how I can fix this<br>
> erorr.<br>
><br>
> Best regards and thanks for helping.<br>
> Ellen<br>
><br>
</div></div>> --<br>
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<br>
Your extensions.conf looks to be incomplete. Any way, dialling<br>
SIP/6201 failed as 6201 is not a valid SIP account (you probably like<br>
to dial SIP/123456789101112<br>
<br>
Please try the following command:<br>
asterisk -rx "originate SIP/123456789101112 application MusicOnHold"<br>
and check asterisk logs. It should dial to the mobile phone and<br>
connect to the MOH application.<br>
<br>
HTH,<br>
Ioan<br>
<br>
--<br>
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</blockquote></div><br>
</div></div></blockquote></div><br>