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Thread: 30 minute cutoff

  1. #31
    Join Date
    Dec 2008
    Location
    Bay Area
    Posts
    31

    Default Re: 30 minute cutoff

    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

  2. #32
    Join Date
    Feb 2007
    Location
    Central Cali :)
    Posts
    553

    Default Re: 30 minute cutoff

    Quote Originally Posted by voxabox View Post
    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 ?

  3. #33
    Join Date
    Feb 2007
    Location
    Irvine CA
    Posts
    1,542,128,043

    Default Re: 30 minute cutoff

    No one is still seeing the problem and is just trying to better understand the technicalities of it, right?
    Timothy Dick
    Founder/CEO
    VOIPo.com

    Interact with VOIPo: Twitter, Facebook

  4. #34
    Join Date
    Dec 2008
    Location
    Bay Area
    Posts
    31

    Default Re: 30 minute cutoff

    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.

  5. #35
    Join Date
    Mar 2007
    Posts
    478

    Default Re: 30 minute cutoff

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

  6. #36
    Join Date
    Sep 2008
    Location
    Southwest MO
    Posts
    219

    Default Re: 30 minute cutoff

    Quote Originally Posted by zcnkac View Post
    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!

  7. #37
    Join Date
    Feb 2007
    Posts
    270

    Default Re: 30 minute cutoff

    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
    Last edited by voxabox; 02-14-2009 at 08:04 AM. Reason: added info inbound calls
    my wish list:
    1. Intelligent Call Forwarding that detects the incoming call originated from the "forwarded" phone and rings the original destination instead
    2. Call History that makes use of CallerID/Custom CallerID+Location. Call History only shows Custom CallerId+Loc. No CNAM look up; Albeitly,it's a step in the right direction!
    3. Scheduled sim. ring with a twist (see wish #1)

  8. #38
    Join Date
    Sep 2008
    Location
    Southwest MO
    Posts
    219

    Default Re: 30 minute cutoff

    Quote Originally Posted by zcnkac View Post
    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

  9. #39

    Default Re: 30 minute cutoff

    Did anyone get an official response on this?

  10. #40
    Join Date
    Sep 2008
    Location
    Southwest MO
    Posts
    219

    Default Re: 30 minute cutoff

    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

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