thr3ads.net - asterisk users - [asterisk-users] Probelm with incoming calls to my DID-Please help me [Sep 2006]

If this information is useful, please help other people find it:
Share via:

Crazy Boy

2006-Sep-01 04:37 UTC

[asterisk-users] Probelm with incoming calls to my DID-Please help me

Hi friends,
 
Thank you to all for your response and cooperation to me. I have a doubt.

We have registered with Teliax and got DID number. We are making calls to USA
successfully using your service. But,  We are unable to receive  incoming calls
to our DID. Here I am sending my config files and error message on Asterisk
console.
  
  Contents in IAX.CONF file:
  
  disallow=all 
  allow = ulaw 
  
  [general]
  register => teliaxusername:teliaxpassword@voip-co1.teliax.com
                      
  [teliax]
  context=telincoming
  type=friend
  host=voip-co1.teliax.com
  auth=md5
  secret=teliaxpassword
  disallow=all
  allow=ulaw
  allow=alaw
  allow=gsm
  
 Contents in Sip.conf file:
 
 [105]
 type=friend
 username=105
 secret=ravi
 callerid="RaviKanth"
 host=dynamic
 context=leader
 canreinvite=no
 nat=yes
 dtmfmode=rfc2833
 allow=all
 mailbox=605@vmail
 
 [107]
 type=friend
 username=107
 secret=suresh
 callerid="Suresh"
 host=dynamic
 context=administration
 canreinvite=no
 nat=yes
 dtmfmode=rfc2833
 allow=all
 mailbox=607@vmail
 
 Contents in Extensions.conf file:
  
  [telincoming]
  exten => 303xxxxxxx, 1, Answer()
  exten => 303xxxxxxx, n, Wait,2
  exten => 303xxxxxxx, n, Goto(incoming,s,1)
 include => internal
  include => incoming
 
  [incoming]
  exten => s,1,Wait(3)
  exten => s,n,Answer
  exten => s,n,SetMusicOnHold(default)
  exten => s,n,Set(TIMEOUT(digit)=5)
  exten => s,n,Set(TIMEOUT(response)=10)
  exten => s,n,Background(/tmp/virg2)
  exten => s,n,Goto(s,1)
  exten => s,n,Hangup()
  include => internal
  
 [internal]
  exten => 105,1,SetMusicOnHold(default)
  exten => 105,2,Dial(SIP/105,7,t,m,T)
  exten => 1605,1,VoiceMailMain(605@vmail)
  exten => 105,3,VoiceMail(605@vmail)
  exten => 105,4,Hangup
  
  exten => 107,1,SetMusicOnHold(default)
  exten => 107,2,Dial(SIP/107,7,t,m,T)
  exten => 1607,1,VoiceMailMain(607@vmail)
  exten => 107,3,VoiceMail(607@vmail)
  exten => 107,4,Hangup
  
  [uscall]
  exten => _1XXXXXXXXXX,1,DIAL(IAX2/teliaxusername@teliax/${EXTEN},30,tr)  
  
  [manager]
 include => uscall
   include => internal
 
   The error message on Asterisk console:
 
  *CLI>     -- Executing Dial("SIP/105-007951e0",
"IAX2/teliaxusername@teliax/1303xxxxxxx|30|tr") in new stack
       -- Called teliaxusername@teliax/1303xxxxxxx
       -- Call accepted by 207.174.202.2 (format ulaw)
       -- Format for call is ulaw
       -- IAX2/teliax-1 is ringing
       -- IAX2/teliax-1 is making progress passing it to SIP/105-007951e0
       -- IAX2/teliax-1 is ringing
       -- IAX2/teliax-1 is busy
       -- Hungup 'IAX2/teliax-1'
     == Everyone is busy/congested at this time (1:1/0/0)
     == Auto fallthrough, channel 'SIP/105-007951e0' status is
'BUSY'
  
 
 What is the problem? Can you please tell me the solution. Looking forward to
your response. Thank you.
 
 Regards,
 Chandra.
 
 				
---------------------------------
Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small
Business.
 		
---------------------------------
How low will we go? Check out Yahoo! Messenger?s low  PC-to-Phone call rates.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060901/736f1e75/attachment.htm

Marco Mouta

2006-Sep-01 08:57 UTC

head link

[asterisk-users] Probelm with incoming calls to my DID-Please help me

Hi,

Please read bellow:

On 9/1/06, Crazy Boy <crazymoonboy@yahoo.com>
wrote:>
> Hi friends,
>
> Thank you to all for your response and cooperation to me. I have a doubt.
>
> We have registered with Teliax and got DID number. We are making calls to
> USA successfully using your service. But,  We are unable to receive
> incoming calls to our DID. Here I am sending my config files and error
> message on Asterisk console.
>
>  Contents in IAX.CONF file:
>
> disallow=all
> allow = ulaw
>
> [general]
> register => teliaxusername:teliaxpassword@voip-co1.teliax.com
>
>
> [teliax]
> context=telincoming
> type=friend
> host=voip-co1.teliax.com
> auth=md5
> secret=teliaxpassword
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
>
> Contents in Sip.conf file:
>
> [105]
> type=friend
> username=105
> secret=ravi
> callerid="RaviKanth"
> host=dynamic
> context=leader
> canreinvite=no
> nat=yes
> dtmfmode=rfc2833
> allow=all
> mailbox=605@vmail
>
> [107]
> type=friend
> username=107
> secret=suresh
> callerid="Suresh"
> host=dynamic
> context=administration
> canreinvite=no
> nat=yes
> dtmfmode=rfc2833
> allow=all
> mailbox=607@vmail
>
> Contents in Extensions.conf file:
>
> [telincoming]
> exten => 303xxxxxxx, 1, Answer()
> exten => 303xxxxxxx, n, Wait,2
> exten => 303xxxxxxx, n, Goto(incoming,s,1)
>
You need to inser "_" before a pattern so asterisk can try to match
it:
exten => _303xxxxxxx, 1, Answer()
exten => _303xxxxxxx, n, Wait,2
exten => _303xxxxxxx, n, Goto(incoming,s,1)

Should solve your problem!

Also only as debug you can try _X.

Pls tell me if it solved your problem.


include => internal> include => incoming
>
> [incoming]
> exten => s,1,Wait(3)
> exten => s,n,Answer
> exten => s,n,SetMusicOnHold(default)
> exten => s,n,Set(TIMEOUT(digit)=5)
> exten => s,n,Set(TIMEOUT(response)=10)
> exten => s,n,Background(/tmp/virg2)
> exten => s,n,Goto(s,1)
> exten => s,n,Hangup()
> include => internal
>
> [internal]
> exten => 105,1,SetMusicOnHold(default)
> exten => 105,2,Dial(SIP/105,7,t,m,T)
> exten => 1605,1,VoiceMailMain(605@vmail)
> exten => 105,3,VoiceMail(605@vmail)
> exten => 105,4,Hangup
>
> exten => 107,1,SetMusicOnHold(default)
> exten => 107,2,Dial(SIP/107,7,t,m,T)
> exten => 1607,1,VoiceMailMain(607@vmail)
> exten => 107,3,VoiceMail(607@vmail)
> exten => 107,4,Hangup
>
> [uscall]
> exten => _1XXXXXXXXXX,1,DIAL(IAX2/teliaxusername@teliax/${EXTEN},30,tr)
>
> [manager]
> include => uscall
> include => internal
>
> The error message on Asterisk console:
>
> *CLI>     -- Executing Dial("SIP/105-007951e0", "
> IAX2/teliaxusername@teliax/1303xxxxxxx|30|tr") in new stack
>     -- Called teliaxusername@teliax/1303xxxxxxx
>     -- Call accepted by 207.174.202.2 (format ulaw)
>     -- Format for call is ulaw
>     -- IAX2/teliax-1 is ringing
>     -- IAX2/teliax-1 is making progress passing it to SIP/105-007951e0
>     -- IAX2/teliax-1 is ringing
>     -- IAX2/teliax-1 is busy
>     -- Hungup 'IAX2/teliax-1'
>   == Everyone is busy/congested at this time (1:1/0/0)
>   == Auto fallthrough, channel 'SIP/105-007951e0' status is
'BUSY'
>
>
> What is the problem? Can you please tell me the solution. Looking forward
> to your response. Thank you.
>
> Regards,
> Chandra.
>
> ------------------------------
> Get your own web address for just $1.99/1st yr. We'll help. Yahoo!
Small
>
Business<http://us.rd.yahoo.com/evt=41244/*http:/smallbusiness.yahoo.com/>
> .
>
> ------------------------------
> How low will we go? Check out Yahoo! Messenger's low PC-to-Phone call
> rates.
>
<http://us.rd.yahoo.com/mail_us/taglines/postman8/*http:/us.rd.yahoo.com/evt=39663/*http:/voice.yahoo.com>
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>

-- 
Com os melhores cumprimentos,

Marco Mouta
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060901/4b9f44ef/attachment.htm

asterisk users - Sep 2006 - Probelm with incoming calls to my DID-Please help me

[asterisk-users] Probelm with incoming calls to my DID-Please help me

[asterisk-users] Probelm with incoming calls to my DID-Please help me