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 Post subject: Save SIP info to MySQL
PostPosted: Tue Jun 19, 2007 7:36 pm 
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Joined: Tue Jun 19, 2007 1:33 pm
Posts: 52
Location: Troy, Michigan
Hello All,

Is there any way to save the generated SIP info to MySQL DB instead of generating on plain file "additional_a2billing_sip.conf" ? And let Asterisk read the SIP information from MySQL DB.

Cheers,
Nitesh


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 Post subject:
PostPosted: Tue Jun 19, 2007 9:24 pm 
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Joined: Fri Jun 01, 2007 5:12 am
Posts: 41
This is "RealTime"
use search on voipinfo.org site

i totaly agree to change the billing from "file-style" rails to SQL-style is best idea.


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 Post subject:
PostPosted: Wed Jun 20, 2007 2:14 pm 
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Joined: Tue Jun 19, 2007 1:33 pm
Posts: 52
Location: Troy, Michigan
Thanks Areski,

I manage to configure Asterisk RealTime and now it loads the SIP users/peers from MySQL DB. The table I am using is of A2Billing DB "cc_sip_buddies".

Now the only problem I am facing is incoming calls are failing... The ATA which is assigned this DID number is behind NAT and according to Olle's explanations he said "there's no support for NAT keep-alives (qualify=) or voicemail indications for these peers." http://www.voip-info.org/wiki/view/Asterisk+RealTime

So does this mean that I can not have any ATA behind NAT with this kind of setup? Below is the error message.

Here is my "extconfig.conf": -
[settings]
sipusers => mysql,mya2billing,cc_sip_buddies
sippeers => mysql,mya2billing,cc_sip_buddies

Here is my "res_mysql.conf": -
[general]
dbhost = 127.0.0.1
dbname = mya2billing
dbuser = billinguser
dbpass = 000eFm500F9E36
dbport = 3306
dbsock = /tmp/mysql.sock

When I do "sip show peers" on the *CLI I can see my SIP user: -
hyperion*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
2486543210/248654321 69.148.36.78 D N 38813 OK (20 ms)
1 sip peers [1 online , 0 offline]

Error message while receiving the call: -
-- AGI Script Executing Application: (DIAL) Options: (SIP/2486543210|60|HL(3600000:61000:30000))
-- Limit Data for this call:
-- - timelimit = 3600000
-- - play_warning = 61000
-- - play_to_caller= yes
-- - play_to_callee= no
-- - warning_freq = 30000
-- - start_sound = UNDEF
-- - warning_sound = timeleft
-- - end_sound = UNDEF
Jun 20 09:49:58 NOTICE[24952]: app_dial.c:1069 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
a2billing.php|1|did: file:Class.A2Billing.php - line:634 - [CARD STATUS UPDATE : UPDATE cc_card SET inuse=inuse-1 WHERE username='9353477140']
-- AGI Script a2billing.php completed, returning 0

Any advice...

Cheers,
Nitesh


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