Re: [Openbts-discuss] Can't make phone calls

[P.K .S. <pks...@ya...>]

which version of openBTS are you using? 2.6 or 2.8.?



 


________________________________
 From: Sebastian Komorowski <ba...@o2...>
To: P.K .Srinivasan <pks...@ya...>; ope...@li... 
Sent: Sunday, 10 February 2013 3:41 PM
Subject: Re:  [Openbts-discuss] Can't make phone calls
 

The problem was solved. I must add to sip.conf:
 
[IMSI001019118319344]
callerid=1111
canreinvite=no
type=friend
allow=gsm
context=sip-external
host=dynamic
dtmfmode=info

[IMSI001012420672246]
callerid=2222
canreinvite=no
type=friend
allow=gsm
context=sip-external
host=dynamic
dtmfmode=info
 
 
and to extensions.conf:
 
[macro-dialGSM]
exten => s,1,Dial(SIP/${ARG1})
exten => s,2,Goto(s-${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,playback(ss-noservice)
exten => s-CANCEL,1,Hangup
[sip-external]
exten => 1111,1,Macro(dialGSM,IMSI001019118319344@127.0.0.1:5062)
exten => 2222,1,Macro(dialGSM,IMSI001012420672246@127.0.0.1:5062)
 
 
Now is question - how to send sms from phone to phone.
 
 
Regards
Sebastian
 
 
 



Dnia 9 lutego 2013 17:13 "P.K .Srinivasan" <pks...@ya...> napisał(a):



Try to send blank SMS to 411 to know whether your phone is successfully registered or not.
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>________________________________
>From: Sebastian Komorowski <ba...@o2...>
>To: =?UTF-8?Q?ope...@li...urceforge.?=
>Sent: Saturday, 9 February 2013 8:40 PM
>Subject: Re: [Openbts-discuss] Can't make phone calls
>
>
>When I call to 600 form 1111 or 2222 I hear some voice and on the asterisk console:
> 
>  -- Executing [600@default:1] playback("SIP/127.0.0.1-00000000", "demo-echotest")
>    -- <SIP/127.0.0.1-00000000> Playing 'demo-echotest.gsm' (language 'en')
>  == Spawn extension (default, 600, 1) exited non-zero on 'SIP/127.0.0.1-00000000'
>  == Using SIP RTP CoS mark 5
>    -- Executing [600@default:1] playback("SIP/127.0.0.1-00000001", "demo-echotest")
>    -- <SIP/127.0.0.1-00000001> Playing 'demo-echotest.gsm' (language 'en')
>    -- Executing [600@default:1] echo("SIP/127.0.0.1-00000001", "")
>  == Spawn extension (default, 600, 1) exited non-zero on 'SIP/127.0.0.1-00000001'
>  == Using SIP RTP CoS mark 5
>    -- Executing [600@default:1] playback("SIP/127.0.0.1-00000002", "demo-echotest")
>    -- <SIP/127.0.0.1-00000002> Playing 'demo-echotest.gsm' (language 'en')
>  == Spawn extension (default, 600, 1) exited non-zero on 'SIP/127.0.0.1-00000002'
> 
>I have this USRP kit from Olifantasia. I don't add anything do sip.conf.
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> 
>Sebastian
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>> try calling from 1111 to your echo number 2600 or 600 and see what happens.
>>also try calling from 2222 and see what happens.
>>
>>check u have added "context default" in the sip.conf entry for 2222
>>
>>
>>pk srinivasan
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>>
>>----------------------------------------------------------------------
>>
>>Message: 1
>>Date: Fri, 08 Feb 2013 20:57:26 +0100
>>From: Sebastian Komorowski <ba...@o2...>
>>Subject: [Openbts-discuss] Can't make phone calls
>>To: ope...@li...
>>Message-ID: <401...@o2...>
>>Content-Type: text/plain; charset="UTF-8"
>>
>>Hello,
>>
>>
>>I'm add entries for each phone in table sip_buddies and dialdata_table in /var/lib/asterisk/sqlite3dir/sqlite3.db. I give it number phone "1111" and "2222".
>>
>>This phones are connected to my OpenBTS - on phones LCD I see "001 01 OpenBTS", but I still can't make phone calls.
>>
>>If I type in console "sudo asterisk -vvvvvr" and I'm try to call on a screen I see:
>>[Feb  8 21:53:30] NOTICE[5508]: chan_sip.c:21614 handle_request_invite: Call from '' to extension '2222' rejected because extension not found in context 'default'.
>>
>>
>>You have some ideas what it's wrong?
>>
>>Sebastian
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