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Joined: Aug 2005
Posts: 106
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Joined: Aug 2005
Posts: 106 |
Connecting a VICIDial server to a Vertical Wave (VICI dialer as recommended by Kumba). Works fine unless the predictive dialer makes the call because the Wave 2500 returns Call Accepted immediately on hearing the 9. This triggers the VICI to think that it's been answered and to hand it off to the agent waiting.
I don't think it's a VICI problem, it's a Vertical problem. However, I'm just not sure how to make it not send that Call Accepted until the call is actually connected.
Any thoughts from a SIP perspective?
Toshiba Sales & Installation Tech
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Joined: Jun 2007
Posts: 2,106
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Joined: Jun 2007
Posts: 2,106 |
The correct method to handle it on a SIP perspective would be to send a "100 Trying", and optionally a "180 Ringing" until the call is actually answered on the PRI/PSTN side of the WAVE.
The problem is that as soon as Asterisk sends an invite to the Wave, the Wave immediately turns around and responds with a "200 OK". In SIP lingo, that's essentially saying that the call is OK and in progress/answered/connected/etc.
This is one of those issues that falls under the interpretation of the SIP protocol. From the Wave engineer's standpoint, I'm sure they respond with a 200 OK because the PBX is accepting the call, as opposed to waiting to respond with a 200 OK when the call is connected to it's endpoint.
For those less versed in SIP, you can equate it to a PBX that is providing a PRI that returns every call as being answered even though the remote end is still being connected/ringing. Not a problem for humanoids but an issue for any type of machine that uses these codes for automation purposes (Dialers, FAX Servers, channel banks, etc).
The only other option if the Wave can't provide different SIP methods is to reverse the order of things and have the Wave connect via SIP/PRI to ViciDial, and then take the PRI directly into ViciDial. We've done this for quite a few clients with great success.
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Joined: Jun 2007
Posts: 2,106
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Joined: Jun 2007
Posts: 2,106 |
Vertical came back that the issue was a DTMF payload type of '96' that they interpreted as just voice traffic versus a DTMF payload type of '101' which they interpreted as Video.
I ended up writing a patch for Asterisk to force it to default to DTMF payload type 96 and recompiled it. Hopefully this correct the issue with the Wave answering all sip calls instead of passing progress back to the endpoint.
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