Odd disconnect?

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Originally Posted by Brian

So have we determined that the PAP2's can't do this successfully?

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I don't know. I do not have a PAP2 to try
however, I can only speculate that none of the sipura/linksys ATAs work as I cannot see anywhere in their documentation that RFC 4028 is supported

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Originally Posted by voxabox

I don't know. I do not have a PAP2 to try
however, I can only speculate that none of the sipura/linksys ATAs work as I cannot see anywhere in their documentation that RFC 4028 is supported

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Are you talking only about asterisk in this conversation when you refer to what the PAP2 will do?

No. To clarify: the UAC (client, ATA, whatever) apparently needs to support RFC4028. The GS doesn't. To the best of our knowledge, the PAP2 doesn't. Asterisk prior to release 1.6 doesn't.

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Originally Posted by dswartz

No. To clarify: the UAC (client, ATA, whatever) apparently needs to support RFC4028. The GS doesn't. To the best of our knowledge, the PAP2 doesn't. Asterisk prior to release 1.6 doesn't.

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I beg to differ
in the User Manual for HandyTone-286 Rev 3.0, section 4.1

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4.1 Key Features
• Support SIP Session Timer

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there's more info about session timer settings on p32-p33 of the same doc

Sigh. That should teach me to post late before I'm heading to bed. I meant to say the GS *does*. Apologies... e.g. I should have said:

"No. To clarify: the UAC (client, ATA, whatever) apparently needs to support RFC4028. The GS does. To the best of our knowledge, the PAP2 doesn't. Asterisk prior to release 1.6 doesn't."

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Originally Posted by dswartz

Sigh. That should teach me to post late before I'm heading to bed. I meant to say the GS *does*. Apologies...

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NP, I did that before except I posted before getting enough caffeine

I had the same problem, using Asterisk 1.4 also. WAF hit bottom after that 30 minute drop. She likes to talk on the phone I tell you what. Any hope for the Asterisk 1.4 people in the crowd?

Scott

Not that I've heard of. On the other hand, it was not a big deal to convert to asterisk 1.6. Ran into a freepbx bug (now fixed) and an asterisk bug (also fixed).

I'm going to try out Asterisk 1.6 sometime when I get a chance. What was interesting is I was on a call with a friend of mine for 56 minutes last night and didn't get booted. But, he has Axvoice, so I wonder since I wasn't terminating to the PSTN if that had something to do with it? hmmm.