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?
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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?
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 RFCQuote:
Originally Posted by dswartz
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
Uh, I don't have an ATA (I'm running asterisk, remember?)
oops, sorry, I did not have enough cafein:pQuote:
Originally Posted by dswartz
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)
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.
Got asterisk 1.6 working (mostly, still a few minor issues), and made a 1-hour call from my cellphone to the voipo line. Analysis of the wireshark trace doesn't show any Session-Expires (unless I just wasn't looking in the right place.) If it isn't right, I don't know why I can make a 1-hour call and couldn't before :(
Update: maybe wireshark just wasn't decoding the packets right? Dunno. I did find a reference for 1.6 that said to set the sip options to 'session-timeout=originate', which will flag my end (the UAC) as the refresher and will reply if the UAS sends the refresh. All seems well, but I have to concur, it seems the openser implementation voipo is using is not complying with the RFC...
it's good to hear that it's working ok for youQuote:
Originally Posted by dswartz
IMHO, it's openser config/handling of non-RFC4028 compliant UACs
Cheers,
-v
So have we determined that the PAP2's can't do this successfully?
I don't know. I do not have a PAP2 to tryQuote:
Originally Posted by Brian
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?Quote:
Originally Posted by voxabox
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.
I beg to differQuote:
Originally Posted by dswartz
in the User Manual for HandyTone-286 Rev 3.0, section 4.1
there's more info about session timer settings on p32-p33 of the same docCode:4.1 Key Features
• Support SIP Session Timer
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."
NP, I did that before except I posted before getting enough caffeineQuote:
Originally Posted by dswartz
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.