Routing through Freeswitch to record calls
QBZappy:
Quote from: v.2geofs on December 18, 2012, 11:50:51 am
The downside so far is that I can only dial out using the AA as I have not yet figured out how to forward the number to be called through freeswitch and acquire them on the OBI side. Once I can do that then I will change my SP1 call route to LN1.
Set the freeswitch extension to unconditionally forward to the OBi extension.
v.2geofs:
Thanks. I'll look into how to do that. If I pass in the phone number, the Obi will dial it?
v.2geofs:
I am attempting to route my calls through freeswitch directly to the OBI extension (1001), but it keeps giving me a 503 error: No service available. I have set SP1 to route directly to LN1 so in theory, when FS passes the call through I should get at least a dialtone, but I am getting nothing. Any ideas?
Thanks,
Geoff
QBZappy:
Re: Possible to use OBi110 to as an FXO port on an Asterisk server?
http://www.obitalk.com/forum/index.php?topic=57.msg103#msg103
Quote from: OBi-Guru on January 23, 2011, 10:12:36 am
5. Set SP1 - InboundCallRoute = LI
a. This rule tells OBi to send all incoming calls (from Asterisk in this case) to the PSTN PORT
b. The number-to-dial will be taken from the incoming INVITE’s request-URI (assuming this is what Asterisk do when talking to a gateway)
c. Note: Under LINE Port – you can fine tune DialDelay, DialDigitOnTime, DialDigitOffTime, DTMFPlaybacklevel parameters, if needed. The default should just work with most PSTN services
d. NOTE: OBi SP1 does not challenge inbound INVITE. However you can setup a list of trusted IP addresses in the X_AccessList paramete (under ITSP Profile – SIP) to limit who can send SIP messages to the OBi SP1. Usually the gateway (OBi) and asterisk machines are in the same subnet; normally not a big issue
Set SP1 - InboundCallRoute = LI <- This is the secret sauce particular to the OBi. To understand this will enlighten you. (Felt like putting a little Zen into this)
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