What I did was modify sip to update the status on the sip friends in
realtime. Then via FAGI dial them directly with the data found in real-time.
(ie dial (SIP/[EMAIL PROTECTED]:5060) Of course you need to check the status
in realtime data before you dial. This allows MANY Asterisk servers
-Users] Realtime SIP Registrations
can you elaborate on modify sip to update the status on the sip friends in
realtime
thanks
On 6/29/06, Doug G [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
What I did was modify sip to update the status on the sip friends in
realtime. Then via
I am having a problem with sip in asterisk 1.2.1 1.2.8 . I have an
account setup with a sip provider. The inbound call is coming from a
SIP proxy, the call is setup (I have audio) and then drops down after
15sec.
What I see in sip traces is that the sip proxy is sending 200 ok
asterisk is
Iam having a problem with sip in asterisk 1.2.1
1.2.8 . I have an account setup with a sip provider. The inbound call is
coming from a SIP proxy, the call is setup (I have audio) and then drops down
after 15sec.
What I see in sip traces is that the sip proxy is sending "200 ok" asterisk
ITSP seeking C programmer to work on Asterisk and SER.
[EMAIL PROTECTED]
Located in Northern NJ
Sorry if I should not post this here
Doug
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Signate runs asterisk on a SGI box.
Nothing special, do yourself a favor and just buy the SGI box yourself. In
fact I have 3 SGI boxes for sale. Ill rip off the Signate labels and
sell them to you.
I worked out an asterisk load
balance solution, so I dont need one all powerful PC. I
Agreed, the first book kind of looked the WIKI in print..
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver
Sent: Tuesday, January 10, 2006 3:44 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
We have done some work on this since my last post.We added some code
to update new fields in the realtime SIP database. Status, Qualify, and
Host Server. We then place the call directly to the phone the SIP
full contact (i.e. dial(sip/[EMAIL PROTECTED]:5060) Via a AGI
script. Our AGI
I think I have 4 options.
1, Modify chan_sip.c to update a new field in sipusers realtime table
with the status of the sip peer/user. Then use agi to dial sip calls.
Check the status field if OK then dial the fullcontact from the sip
table. If not goto voicemail or where ever else I want the
