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" <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]>
To: [email protected]
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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]:5061



Who do I configure * to dial from one to another as an extension in a network?

Thanks

_________________________________________________________________
¿Estás pensando en cambiar de coche? Todas los modelos de serie y extras en MSN Motor. http://motor.msn.es/researchcentre/

_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

_________________________________________________________________
Móviles, DVD, cámaras digitales, coleccionismo... Con unas ofertas que ni te imaginas. http://www.msn.es/Subastas/

_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to