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Author Topic: Routing through Freeswitch to record calls  (Read 5103 times)
v.2geofs
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Posts: 8


« on: December 17, 2012, 02:22:27 pm »

Hi.  We have the OBI110 in our office and are trying to configure it to port through freeswitch for recording.  I have run into a serious snag.  We've managed to use a gateway (vg1) to route LN1 to Freeswitch, then Freeswitch will setup recording and bridge the call back to SP2 where the inboundCallRoute is set to PH.  When I call our number it shows up in Freeswitch, and the phone rings.  But when I pick up the line it is silent for 2 seconds then I get a dialtone, and the inbound call is still ringing.  Not quite sure where I am missing on this but any help would be appreciated.  Ultimately what we would like to do is record all inbound and outbound calls to our office.

Thank you,

Geoff
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QBZappy
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« Reply #1 on: December 17, 2012, 03:31:38 pm »

v.2geofs,

Welcome.

I have a feeling that we can expect a "Freeswitch How To" very soon. There isn't much, if any documented use with Freeswitch.  Cheesy
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Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.
v.2geofs
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« Reply #2 on: December 18, 2012, 07:46:14 am »

I've discovered that.  Have you any idea why my bridged call is not picking up?
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QBZappy
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« Reply #3 on: December 18, 2012, 09:37:35 am »

v.2geofs,

I don't have any experience with Freeswitch. However that won't stop me from commenting.  Cheesy

Perhaps if you set the OBi as an extension off the pbx using one of the sip accts (not using VG), you might find it easier to set up. Routing can then be done using sip uri ex: SPX(100@freeswitch_ip:port).
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Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.
v.2geofs
Newbie
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Posts: 8


« Reply #4 on: December 18, 2012, 11:50:51 am »

What I have done is set SP1 and SP2 to be unique extensions on Freeswitch (1001 and 1002).  Then I configured VG1 to go to the Freeswitch IP as 1003.  I routed all LN1 calls to vg1($1), and all SP2 inbound calls to PH, and all Sp1 inbound calls to aa.

Then in freeswitch I bridged 1003 to 1002 and I bridged 1002 to 1001.  This allows me to record all inbound calls in freeswitch (destination_number=1002) and all outbound calls (destination_number=1001).  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.

Right now I am working on programming freeswitch to navigate the AA, unless the is another method to do that.  Going to try vg3 for that and see what happens.

Thanks for the suggestions.
Geoff
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QBZappy
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Posts: 2316



« Reply #5 on: December 18, 2012, 12:06:52 pm »

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.
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Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.
v.2geofs
Newbie
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Posts: 8


« Reply #6 on: December 21, 2012, 05:21:25 am »

Thanks.  I'll look into how to do that.  If I pass in the phone number, the Obi will dial it?
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v.2geofs
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Posts: 8


« Reply #7 on: December 21, 2012, 01:33:14 pm »

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
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QBZappy
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Posts: 2316



« Reply #8 on: December 21, 2012, 03:09:30 pm »

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

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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Owner of the 1st OBi110/100 units in service in Canada & South America. 1st OBi202 on my street. 1st OBi1032 in Montreal.
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