30 minute cutoff

[zcnkac]

Here are my asterisk logs for an outgoing and incoming call. I am using central01 on asterisk 1.4.21.2.

[root@pbx /etc/asterisk]# chans
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message
192.168.x.x 1001 31c81cc17b9 00102/00000 0x4 (ulaw) No Tx: ACK
67.228.251.106 408XXXXXXX 84017861_11 00101/25111 0x4 (ulaw) No Rx: ACK


[root@pbx /etc/asterisk]# asterisk -rx sip show channel 84017861_11

* SIP Call
Curr. trans. direction: Incoming
Owner channel ID: SIP/408XXXXXXX-xxx
Joint Codec Capability: 4
Format: 0x4 (ulaw)
MaxCallBR: 384 kbps
Caller-ID: 408XXXXXXX
Need Destroy: 0
Last Message: Rx: ACK
Promiscuous Redir: No
DTMF Mode: rfc2833
SIP Options: timer
Session-Timer: Active
S-Timer Interval: 1800
S-Timer Refresher: uac
S-Timer Expirys: 0
S-Timer Sched Id: 363759
S-Timer Peer Sts: Active
S-Timer Cached Min-SE: 90
S-Timer Cached SE: 1800
S-Timer Cached Ref: uac
S-Timer Cached Mode: Accept

[root@pbx /etc/asterisk]# asterisk -rx "sip show channels"
Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message
67.228.251.106 1408XXXX 5e5ba777527 00103/00000 0x4 (ulaw) No Tx: ACK
192.168.x.x 1001 NDI5N2I0OTE 00101/00002 0x4 (ulaw) No Rx: ACK


[root@pbx /etc/asterisk]# asterisk -rx "sip show channel 5e5ba777527"

* SIP Call
Curr. trans. direction: Outgoing
Call-ID: 5e5ba777527e8db43f7aa42836e8bc48@cen...pwel come.com
Owner channel ID: SIP/voipo-out-xxx
Joint Codec Capability: 4
Format: 0x4 (ulaw)
SIP User agent: FreeSWITCH-mod_sofia/1.0.trunk-11100
Username: 1408XXXXXXX
Peername: voipo-out
Original uri: sip:mod_sofia@174.132.131.133:5060
Need Destroy: 0
Last Message: Tx: ACK
Promiscuous Redir: No
DTMF Mode: rfc2833
SIP Options: (none)
Session-Timer: Inactive

[Montano]

Local call does not count. This is voip...
Is it sip to sip or sip to pots?

Do we really know for sure anymore ? People calling me assume it's a POTS number.......

I did not dial a sip address, so do we assume it went to POTS ?

[VOIPoTim]

No one is still seeing the problem and is just trying to better understand the technicalities of it, right?

[zcnkac]

Cool ! RFC 4028 timers are no longer required. I disabled session-timers on my asterisk box and the call status showed session timers to be inactive. I also made an incoming call that lasted more than 30 mins.

Better yet, session timers can now be enabled both on outgoing and incoming. It's optional, so adapters like PAP that don't support timers won't break but for others like grandstream/asterisk 1.6, it can be used. Sweet ! It's possible timers on outgoing calls is a new thing on central01. For asterisk users who want to try this out, add a session-timers=originate to your voipo peer settings.

[dswartz]

Yeah, I had to tweak asterisk 1.6 that way when I first converted, to avoid having my calls chopped off

[scott2020]

Cool ! RFC 4028 timers are no longer required. I disabled session-timers on my asterisk box and the call status showed session timers to be inactive. I also made an incoming call that lasted more than 30 mins.

Very interesting. Is this just a fluke or did Tim or someone from VOIPo come out and say they definitely have taken the timers out of the mix? In the past I would get hit or miss 30 minute cutoffs with my Asterisk box. Now I am tempted to go back.

But, my GS 502 is working so well I don't really want to mess with it!

[voxabox]

BYOD'ers,
I'm happy to report that that outbound calls (to plain POT numbers) have been verified working for 30+ minute durations - thanks to the wife
ATA: SPA-1001, 3.1.19(SE), behind router with STUN, NAT keep-alive, ports: 5062,16384-16482 forwarded

02/13/09, inbound calls have been working for more than 30+ mins.
Last time I checked to call log, my wife answered "her call" and talked to her friend for 80+ mins

Cheers,
-v

[scott2020]

Cool ! RFC 4028 timers are no longer required. I disabled session-timers on my asterisk box and the call status showed session timers to be inactive. I also made an incoming call that lasted more than 30 mins.

I'd still like the official word from VOIPo on whether or not the session timers are in fact gone, or is it a coincidence that some BYOD devices are working past 30 minutes. When I was running Asterisk 1.4 I would sometimes have calls go longer than 30 minutes and sometimes not.

Scott

[bubbanc]

Did anyone get an official response on this?

[scott2020]

Not that I have heard, but with the Linksys PAP2's they suffered from the same session timer problem. In order to get those to work, from what I understand, VOIPo had to turn off the session timers. That is how I understand it but I don't know for sure and didn't get an official answer yet.
Scott