Sep 25 15:47:23 WARNING[1087986608]: chan_sip.c:598 __sip_xmit: sip_xmit of 0x81487dc (len 755) to 210.50.7.213 returned -1: Invalid argument
when i send a sip call to my cisco 3660. see my earlier post today.
Matt Darnell wrote:
Aloha,
I have a Cisco Gateway the is functioning as my SIP Gateway to the PSTN.
I have an Asterisk box, RC2 with a for port FXS card providing dialtone for a Norstar Key System.
I have it working so when you press a line key on the Norstar you get dial tone from the Asterisk box. The user has to dial '9' then they can dial there number which is sent to the Cisco GW via SIP and the call is completed.
I can not seem to get rid of the need to dial a lead digit. I don't need any other digits - i.e. voicemail, park - we aren't using any * 'features' just as a SIP<->FXS gateway.
Is it posible so I can create templates to collect the number and send the call to the Cisco when the template is completed
911 411 611 1[2-9]XX-XXX-XXX [2-9]XX-XXXX .....
The users are not likeing to have to dial '9'
Looking forward to updateing to 1.0.0
Matt
_______________________________________________
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
--
David Winter
Senior Network Engineer
Planet-Telecom, Inc.
Tampa FL
(813)901-5182 Office
(813)864-3162 Direct
(813)817-4204 Mobile
(813)881-9762 Fax
------------------------------------------
AIM: mobofool
ICQ: 3563403
MSN: [EMAIL PROTECTED]
Y!: vt_fool
_______________________________________________ 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
