[Asterisk-Users] SIP Testing

[Gregg Lebovitz]

I would have someone explicity test against iconnect (I volunteer) since
I use this service quite a bit. I would like to see it finally work
properly. I have seen the problems with hangups reported on the list.

Iconnect uses a Cisco Gateway.

Gregg

On Mon, 2003-04-07 at 01:14, Adam Goryachev wrote:
> > Anyway I'm soliciting for ideas from the list.  I'd be happy to get some
> > feedback.
> 
> Perhaps a checklist could be provided to various (at least 2) people for
> each type of equipment.
> 
> ie for device = ata186
> Call from SIP device to SIP device
> Transfer from SIP device to SIP device
> Transfer from SIP device to Zap device
> Call from Zap to SIP
> Call from SIP to Zap
> Music on Hold
> Conferencing
> DTMF to asterisk
> DTMF to zap device
> etc...
> 
> Obviously each time a release is made and a subsequent bug found we have
> another item for the checklist.
> 
> Regards,
> Adam
> 
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