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Asterisk able to receive in comming calls but unable to place out going

Posted on 2008-10-03
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Last Modified: 2011-10-19
I have three Cisco 7940 phones connecting to a Server I setup running Asterisk now. I am able to see the phones connect through SIP, I am able to call each phone via there extension. But I am not able to dial out.
I receive this error
 NOTICE[2261]: chan_sip.c:13885 handle_request_invite: Call from '' to extension '93997574' rejected because extension not found.
Phone*CLI>

I am unsure what could be wrong. Any help would be greatly appreciated
 
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Question by:flipb18b
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5 Comments
 
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Accepted Solution

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Mysidia earned 500 total points
ID: 22642710
When you see "extension not found"   it most likely means that there is no exten entry in extensions.conf that matches '93997574'
from the 'context' that the SIP peer is in.

Please check  /etc/asterisk/sip.conf     find the extension's sip peer
check what the outbound context is.

Then use "show dialplan"  to see how outbound calls to that numbers are handled.
Or post the "show dialplan"  command and output used here.


Instructions below on how to use show dialplan:


at a shell command line, try typing this...  
#rasterisk

Then you should be in the  asterisk CLI:
CLI> sip show peer    <TYPE THE EXTENSION NUMBER HERE>

You should see a result like this
  * Name       : 1234
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : from-internal
! ^^^^^^^^^^^^^^^^^^^^^ Context for outbound dialing
! ^^^^^^^^now you type  'show dialplan (EXTENSION)@(CONTEXT)'
! ^^^^^^^^ for example:   'show dialplan 1234@from-internal'
  Subscr.Cont. : <Not set>
  Language     :
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Mailbox      : 1234@device
  VM Extension : *97
  LastMsgsSent : 0/1
  Call limit   : 50
  Dynamic      : Yes
  Callerid     : "device" <1234>
  MaxCallBR    : 384 kbps
  Expire       : 1430
  Insecure     : no
  Nat          : No
  ACL          : No
  T38 pt UDPTL : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : Yes
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       :
  Addr->IP     : aaa.bbb.ccc.ddd Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 1234
  SIP Options  : (none)
  Codecs       : 0xc (ulaw|alaw)
  Codec Order  : (ulaw:20,alaw:20)
  Auto-Framing:  No
  Status       : Unmonitored
  Useragent    : Cisco-CP7960G/8.0
  Reg. Contact : sip:1234@aaa.bbb.ccc.ddd:5060;transport=udp


This is an example of what show dialplan output looks like

ast1*CLI> show dialplan 93997574@from-internal
[ Included context 'outrt-006-9Default' created by 'pbx_config' ]
  '_9.' =>          1. Macro(user-callerid|SKIPTTL|)              [pbx_config]
                    2. Set(_NODEST=)                              [pbx_config]
                    3. Macro(record-enable|${AMPUSER}|OUT|)       [pbx_config]
                    4. Macro(dialout-trunk|5|${EXTEN:1}||)        [pbx_config]
                    5. Macro(outisbusy|)                          [pbx_config]
ast1*CLI>
[ Included context 'bad-number' created by 'pbx_config' ]
  '_X.' =>          1. ResetCDR()                                 [pbx_config]
                    2. NoCDR()                                    [pbx_config]
                    3. Wait(1)                                    [pbx_config]
                    4. Playback(silence/1&cannot-complete-as-dialed&check-number-dial-again|noanswer) [pbx_config]
                    5. Wait(1)                                    [pbx_config]
                    6. Congestion(20)                             [pbx_config]
                    7. Hangup()                                   [pbx_config]

-= 2 extensions (12 priorities) in 2 contexts. =-
ast1*CLI> exit


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LVL 4

Expert Comment

by:denisdsr20
ID: 22649166
If you are using atserisk@home and may be freePbx or some stuff like this to configure your server, you should go to add extension page and add a sip extension for any of the number you'd like to call as local attached sip devices.

Hope this help

Regards.

Denis / SR20
AddSIPexten.bmp
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