Hi,
I already contacted what I inputed on my softphone but we
both can't hear each other. I used X-lite and the other is a hardware
SIP phone. What could be the problem?
Thanks,
Ryan
At 03:03 PM 12/16/05, you wrote:
yes
$AGI->exec('Dial', "SIP/[EMAIL PROTECTED]");
Diyanat
From: Ryan Pagquil <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussion<asterisk-users@lists.digium.com>
To: Asterisk Users Mailing List - Non-Commercial
Discussion<asterisk-users@lists.digium.com>, asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] SIP Trunk please help
Date: Fri, 16 Dec 2005 13:56:09 +0800
MIME-Version: 1.0
X-OriginalArrivalTime: 16 Dec 2005 05:58:00.0170 (UTC)
FILETIME=[AB7B14A0:01C60205]
Hi,
Thanks for the reply... Actually I'm using AGI to do it
instead of defining it on extensions.conf... Would it be the same
in extensions.conf? Should I write $AGI->exec('Dial',
'SIP/[EMAIL PROTECTED]'); to dial it from AGI script (perl), is this correct?
Thank you very much,
Ryan
At 01:45 PM 12/16/05, Diyanat Ali wrote:
in the sip.conf have the following enteries
; for regsitering with ser
register:seruser:[EMAIL PROTECTED]:5060;(put ser machine ip:port)
;add a user for the ser machine
[seruser]
type=friend
host=0.0.0.0 ;(put ser machine ip here)
nat=no ;(change as needed )
canreinvite=yes ;(change as needed)
insecure=very ;(change as needed)
disallow=all
allow=ulaw
allow=gsm
context=sip
dtmfmode=rfc2833
in extensions.conf under contect [sip]
[sip]
;replace extension and the priority to macth your dial plan
exten => _X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) ;(seruser is defined
in sip.conf)
Diyanat
From: Ryan Pagquil <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussion<asterisk-users@lists.digium.com>
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP Trunk please help
Date: Fri, 16 Dec 2005 10:31:24 +0800
MIME-Version: 1.0
Hi,
I've been setting up asterisk for prepaid use. I'm
testing to call a SER registered user from the Asterisk just to
simulate the prepaid calls. Now, I can already contact Asterisk
and it prompts me to input my call card number and after that I
dial in the number I want to call (a SER registered device). My
question is how can I implement on sip.conf to use my SER as the
trunk line? So that calls will be forwarded to it. Do I also
need to register asterisk on SER?How?
Please help!
Thanks,
Ryan
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