search for: _00xx

Displaying 4 results from an estimated 4 matches for "_00xx".

Did you mean: 100xx
2009 Sep 29
1
Fax and dial-up connection issues
...p=3 pickupgroup=3 mailbox=7875 channel => 125 /etc/asterisk/extensions.conf: [fax] ignorepat => 0 include => local ;Ligacoes locais exten => _0XXXXXXXX,1,SetTransferCapability(3K1AUDIO) exten => _0XXXXXXXX,n,Dial(DAHDI/g1/${EXTEN:1},60) ;Ligacoes DDD - telefones fixos exten => _00XX[2-6]XXXXXXX,1,SetTransferCapability(3K1AUDIO) exten => _00XX[2-6]XXXXXXX,n,Dial(DAHDI/g1/021${EXTEN:2},60) # dahdi_test: svoip01:~# dahdi_test -vv Opened pseudo dahdi interface, measuring accuracy... 8192 samples in 8199.664 system clock sample intervals (100.094%) 8192 samples in 8198.728 sy...
2005 Jan 17
1
ASTCC single stage + no access number + auth usingsip username and password
...to have all SIP phones to work on prepaid basis > and without need to dial any access number, instead I would > like to use the phone as normal dialing only the destination > number, for example 00464090510. I use the AccountCode for authentication. This is how, for example: exten => _00XX.,1,DeadAGI(astcc.agi,${ACCOUNTCODE},011${EXTEN:2}) > Once the call is finished I would like to have the balance > shown in the display by sending a sip message to the phone(if > possible otherwise not important). This would require adding code to the AGI, if it's even possible. --...
2011 Apr 05
1
Number Conversion
Hi all, Please, could somebody point me out what is going wrong in this line below? exten => _00XX.,1,Dial(DAHDI/G0/021${EXTEN:4},45,rT) As I know, such line must convert any number dialed to 021, therefore, as we can see, it's kept the number dialed! -- Executing [00151236445600 at a2billing:1] Dial("SIP/2000-00000000", "DAHDI/G0/0151236445600,45,rT}") in new stack...
2003 Dec 16
28
codec negotiation
Hi list, I'm with a little problem on codec negotiation between a cisco827 and asterisk. My sip.conf is like that: [general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:<asterisk-ip>