Odd disconnect?

[dswartz]

Made a call to a nearby town around 9:30PM EST. Almost exactly 30 minutes into the call, it was dropped and I heard fast busy. Redial went right through. Has anyone else had this happen?

[voxabox]

Made a call to a nearby town around 9:30PM EST. Almost exactly 30 minutes into the call, it was dropped and I heard fast busy. Redial went right through. Has anyone else had this happen?

by any chance you're using your own ATA, if you do see this thread
especially my post
BTW, I asked for a VOIPo approved ATA and it has been working OK

[dswartz]

Hmm, interesting. I could swear I've had calls longer than 30 minutes before. Also, NAT is not an issue for me (I re-read that thread), and if you scan back in it, you'll see I mentioned that my gateway runs asterisk, so NAT is not involved.

[voxabox]

Hmm, interesting. I could swear I've had calls longer than 30 minutes before. Also, NAT is not an issue for me (I re-read that thread), and if you scan back in it, you'll see I mentioned that my gateway runs asterisk, so NAT is not involved.

for me, NAT was not an issue, it seemed to be the fact that the SIP server did not see my ATA (SPA1001) as a non RFC 4028 compliant one; furthermore, the SIP server did not refresh the session timer (send the re-INVITE) as called for by the RFC

you could verify this by sniffing the SIP convo

BTW, is your ATA a linksys/sipura one?

the grandstream HT286 and its variants are the only few ATAs out there that support RFC 4028

[dswartz]

Uh, I don't have an ATA (I'm running asterisk, remember?)

[voxabox]

Uh, I don't have an ATA (I'm running asterisk, remember?)

oops, sorry, I did not have enough cafein

I still have my doubt that it is a NAT problem

anyways, do sip trace to see if asterisk supports session timer and who's responsible for the refresh

if you want, you can post the trace here (after some editing to protect the identity of the innocent)

[dswartz]

I doubt it too, since NAT isn't involved for me. I will be making a 30+ minute call tomorrow, so I will fire up a sniffer first.

Update: I tried calling my cellphone from the voipo number and after 30:16, it disconnected. Looking at ethereal trace now...

Update2: I don't see the update request either, and can confirm a BYE sent from the other end around 1800 seconds into the call. Bad news: asterisk 1.4 (what I am running) does NOT support RFC-4028. Possibly good news: asterisk 1.6 does (and that is much more stable now, as well as being (mostly) supported by freepbx 2.5. I will give a try to 1.6 and report back.

[Brian]

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

[voxabox]

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

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

[burris]

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

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