[Asterisk-Users] Dial SIP extension

[Patricio Ku]

I have the following configuration:
; /etc/asterisk/extensions.conf.

[extensionessip]
;exten=>i,1,NoCDR()
;exten=>i,2,Hangup()
;exten=>s,1,Wait(2)
;exten=>s,2,DigitTimeout(6)
;exten=>s,3,ResponseTimeout(10)
;exten=>t,3,Hangup() ; t: transfer call to another extension
;exten=>_10XX.,1,Dial(SIP/${EXTEN},60,tr) ; r= tono falso

exten => 1001,1,Dial(SIP/1001,60)
exten => 1001,2,Hangup

exten => 1002,1,Dial(SIP/1002,60)
exten => 1002,2,Hangup

exten => 1003,1,Dial(SIP/1003,60)
exten => 1003,2,Hangup

;exten=>_009[13456789].,1,Dial(SIP/primus/${EXTEN},60,tr)
;exten=>_009[2].,1,Dial(SIP/primus/${EXTEN},60,tr)
;exten=>_00[12345678].,1,Dial(SIP/primus/${EXTEN},60,tr)
;exten=>_6[0123456789].,1,Dial(SIP/primus/${EXTEN},60,tr)
;exten=>_9[123456789].,1,Dial(SIP/primus/${EXTEN},60,tr)


But when I try to call from the sipura 1002 to 1003, 1003 does not ring.
Any one has an idea whats the problem?

thanks

>From: "Patricio Ku" <patricioamanku at hotmail.com>
>Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
><asterisk-users at lists.digium.com>
>To: asterisk-users at lists.digium.com
>Subject: [Asterisk-Users] Dial SIP extension
>Date: Mon, 11 Jul 2005 09:43:15 +0000
>
>I have 2 Sipuras with the following configuration:
>The first:
>SAS Enable: no, NAT Mapping Enable: No, Sip Port 5060, USER ID 1002, Auth 
>ID: 1002, Preferred Codec G711a, Prefered Codec Only: no, DTMF Tx Method, 
>INFO, Enable IP Dialing: no.
>The second the same with USER ID : 1003
>
>  * Name       : 1002
>  Secret       : <Not set>
>  MD5Secret    : <Not set>
>  Context      : outgoing
>  Language     :
>  AMA flags    : Unknown
>  CallingPres  : Presentation Allowed, Not Screened
>  Callgroup    : 1, 33
>  Pickupgroup  : 1, 33
>  Mailbox      :
>  LastMsgsSent : -1
>  Inc. limit   : 0
>  Outg. limit  : 0>
>  Dynamic      : Yes
>  Callerid     : "" <>
>  Expire       : 243334
>  Expiry       : 900
>  Insecure     : no
>  Nat          : Always
>  ACL          : No
>  CanReinvite  : No
>  PromiscRedir : No
>  User=Phone   : No
>  DTMFmode     : info
>  LastMsg      : 0
>  ToHost       :
>  Addr->IP     : x.x.x.x (dont ask) Port 32453
>  Defaddr->IP  : 0.0.0.0 Port 5060
>  Def. Username: 1002
>  Codecs       : 0x8 (alaw)
>  Codec Order  : (alaw)
>  Status       : OK (92 ms)
>  Useragent    : Sipura/SPA2000-2.0.10(e)
>  Reg. Contact : sip:1002 at 192.168.1.200:5061
>
>* Name       : 1003
>  Secret       : <Not set>
>  MD5Secret    : <Not set>
>  Context      : outgoing
>  Language     :
>  AMA flags    : Unknown
>  CallingPres  : Presentation Allowed, Not Screened
>  Callgroup    : 1, 33
>  Pickupgroup  : 1, 33
>  Mailbox      :
>  LastMsgsSent : -1
>  Inc. limit   : 0
>  Outg. limit  : 0
>  Dynamic      : Yes
>  Callerid     : "" <>
>  Expire       : 242099
>  Expiry       : 900
>  Insecure     : no
>  Nat          : Always
>  ACL          : No
>  CanReinvite  : No
>  PromiscRedir : No
>  User=Phone   : No
>  DTMFmode     : info
>  LastMsg      : 0
>  ToHost       :
>  Addr->IP     : x.x.x.x Port 27495
>  Defaddr->IP  : 0.0.0.0 Port 5060
>  Def. Username: 1003
>  Codecs       : 0x8 (alaw)
>  Codec Order  : (alaw)
>  Status       : OK (90 ms)
>  Useragent    : Sipura/SPA2000-2.0.10(e)
>  Reg. Contact : sip:1003 at 192.168.1.75:5061
>
>
>
>Who do I configure * to dial from one to another as an extension in a 
>network?
>
>Thanks
>
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