Hi,
i need to save into a local variable the user's input dialed during
the cmd Authenticate(). Is there a way to do it?
thx
rich
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Danny, Doug
thx for the replies. According to the documentation, there is no
change for Authenticate() in version 1.6.x.x. So it seems i have to
use Read().
rich
On Tue, Jun 29, 2010 at 3:26 PM, Doug Lytle supp...@drdos.info wrote:
Coco Richard wrote:
Hi,
i need to save into a local
Hi,
there are several possibilities do to it
REGISTER Username/Extensions Enumeration
INVITE Username/Extensions Enumeration
OPTION Username/Extensions Enumeration
for more information:
http://www.hackingvoip.com/presentations/sample_chapter3_hacking_voip.pdf
rich...
On Thu, Nov 19, 2009 at
Hi,
asterisk version is 1.4.13
rich...
On Tue, Nov 10, 2009 at 7:01 AM, Tilghman Lesher tles...@digium.com wrote:
On Monday 09 November 2009 15:38:54 Coco Richard wrote:
i'm not sure to understand. Asterisk does support SIP INFO, so why
doesn't Asterisk add the INFO Method in the 200OK
I took a look in chan_sip.c an for 1.4.13 ALLOWED_METHODS doesn't add
INFO. So I will upgrade to 1.6...
thank you for the replies...
rich...
On Tue, Nov 10, 2009 at 9:21 AM, Coco Richard
richard.kingc...@gmail.com wrote:
Hi,
asterisk version is 1.4.13
rich...
On Tue, Nov 10, 2009 at 7
Hi all,
In the INVITE from my SIP provider to Asterisk i can see that the
Allow Header includes an INFO Method, but Asterisk replies a 200 OK
with an Allow Header without INFO Method. But in the RFC3261 (20.5)
you can read:
All methods, including ACK and CANCEL, understood by the UA MUST be
of that
method in order for the other UA to be willing to send messages with
that request method to it.
Coco Richard wrote:
Hi all,
In the INVITE from my SIP provider to Asterisk i can see that the
Allow Header includes an INFO Method, but Asterisk replies a 200 OK
with an Allow Header without INFO
Hi all,
our asterisk is connected to a sip proxy through a sip trunk. Let's say we
have following dial plan (only an example)
[from_sip_proxy]
exten = 36122512,1,Answer()
exten = 36122512,2,VoiceMailMain()
exten = 3612252,1,Answer()
exten = 3612252,2,MeetMe(313,MI)
exten = 3612252,3,HangUp()
Hi all,
How can i use different VLANs for signaling and audio, e.g vlan 100 for sip
and vlan 200 for rtp? Where can i find documentations for this?
Comments and suggestions are welcomed (a sample config too :-)))
thx in advance
rich
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