[asterisk-users] chan_pjsip ignoring endpoint device state (qualify) on dial
jajcus at jajcus.net
Wed Aug 10 02:18:23 CDT 2016
On 2016-08-09 10:06, Faheem Muhammad wrote:
> This might be a bug or configuration issue, but you need to understand
> the SIP Session Timers. With Session Timers you can control the round
> trip time and Call Setup time of SIP Requests.
I don't think you really mean SIP Session Timers
(https://tools.ietf.org/html/rfc4028) these do not affect RTT or call
setup, but provide kind of 'keepalive' and session expiration for
> In case of GSM Network with high delay you need to set the T1 timer a
> higher value like 1000ms (500 ms default). Similarly you can reduce the
> Call setup time by configuring 'T2' upto you choice as per you telephony
> network. Configure t1min, timert1 and timerb according to your network.
> Also set session-type=uas.
Yes, tweaking the T1 and T2 timers may work for me. I'll try that,
though the old 'qualify' magic with chan_sip was quite convenient. I
wonder why it doesn't work with chan_pjsip.
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