how to configure sangoma CPD on goautodial?

[imam_mahmudi]

GoAutodial
VERSION: 2.2.1-237
BUILD: 100510-2015

I have bought sangoma CPD with 30 licens, but i don't know how to configure on goautodial.
please can help me to configure on goautodial? with step by step configuration.

thank's

[williamconley]

did you buy it through The Vicidial Group?

[mflorell]

Sangoma is now referring all CPD sales on Vicidial directly to us, and we can offer some pretty steep discounts over the list pricing.

I don't recall selling any 30-channel CPD licenses lately.

[knotbeerdan]

can anyone explain why exactly we need these devices? I am in the beginning of developing a large solution and would like to know at what point I would be needing this type of device.

[mflorell]

Asterisk can't do call progress analysis, that's the major issue. Sangoma's product provides CPA starting when the call is placed using audio and signalling of the call, and is also able to do very accurate Answering Machine Detection.

[williamconley]

Another view: Asterisk relies on signalling from the provider for call progress. And there is no "standard" method of transmitting call progress (I cannot even testify that there is "A" method of accurately transmitting call progress to the same standard as PSTN). This is why AMD sucks (blunt enough?) and why carriers differ in their methods so widely (because SIP channels COULD transmit enough data to qualify, but there is no "agency" to enforce any form of "you must do it this way".)

So Sangoma's method sounds like they are analyzing the audio ... (just like a human would), which IS always present ... so it makes sense that would work where the other methods fail, unless you have a nice long discussion with your VOIP provider.

But remember: most VOIP providers are merely distributors, unless they have PSTN or E1/T1 on the other end of their wires, they have the same reporting issues that you do!

[cristian]

I cannot even testify that there is "A" method of accurately transmitting call progress to the same standard as PSTN.

Sip info carries all "telephony event" signaling including SS7, supplementing SIP dialog. It is a deployed standard and used by Cisco as a MGCP replacement.

http://www.ietf.org/rfc/rfc2976.txt

This is why AMD sucks (blunt enough?)...

Asterisk's AMD implementation does not use carrier signaling, save call connect (200 OK).

http://www.voip-info.org/wiki/index.php ... sk+cmd+AMD

[mflorell]

The biggest problem is that Asterisk has no method for analyzing pre-Answer audio, so AMD can only start working after Asterisk has realized that the carrier sent an Answer signal. This is the biggest limiting factor in having a complete call progress analysis system, and since Sangoma CPD operates outside of Asterisk and gets the audio before Asterisk, it can operate as a complete call progress analysis package.

[cristian]

I never thought about it that way.

[williamconley]

That was brilliant.

Oh: and

It is a deployed standard and used by Cisco as a MGCP replacement

was my point, actually. "Used by Cisco ..." is not "an agreed upon and enforced standard by all domestic VOIP" ... there is no regulation, and ONE company using a standard is not a standard, it's ONE company. each provider decides on their own what to pass. So there will not be a reliable solution anywhere in the near future.

Except Sangoma CPA/CPD, which is available NOW.

[cristian]

All most everything supports SIP Info, including asterisk.

SIP itself specifies proper setup and teardown signaling, that's what SIP is for. Sending a 200 OK upon a call coming up /is/ the SIP standard.

Asterisk AMD still only cares about a 200 OK. That's when asterisk calls it.

The truth is the carriers you are using are poor. The delay issue due to multiple SIP proxies is absurd.

Find better carriers. It will save you money and provide better quality. That's also available NOW.

[cristian]

I talked to my Sangoma rep and the Sangoma CPA does examine the early RTP. Awesome!