30 minute cutoff

[voxabox]

PAP2T, sip.voipwelcome, local call.

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

[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 ?

[Montano]

I must disagree. All the calls I got cut off on, were initiated by me.

[zcnkac]

Outgoing calls cutting off might have been because of something else. I have since switched to central01. Before that, outgoing calls would drop sometimes because asterisk could not "re-authenticate". It might also be because I was using codeblue in fromdomain.

I can definitely say that RFC 4028 session-timers are being enabled only for incoming calls.

[dswartz]

Outgoing calls cutting off might have been because of something else. I have since switched to central01. Before that, outgoing calls would drop sometimes because asterisk could not "re-authenticate". It might also be because I was using codeblue in fromdomain.

I can definitely say that RFC 4028 session-timers are being enabled only for incoming calls.

Hmmm, that's weird. I could swear I had a number of calls dropped at the 30 minute mark when I was calling other people (that was on asterisk 1.4 before I moved to 1.6).

[voxabox]

Guys, read the rfc again
Sst is to prevent zombies

[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

[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