asterisk users - Aug 2016 - chan_pjsip ignoring endpoint device state (qualify) on dial

On 2016-08-09 10:06, Faheem Muhammad wrote:
> trip time and Call Setup time of SIP Requests.
> 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.
No, that won't work.

First ? 't1min', 'timert1' and 'timerb' are chan_sip,
not pjsip options.

Second ? the 'timer_t1' and 'timer_b' settings available in
'pjsip.conf'
are global and not per-endpoint. I cannot change T1 for trunks, as they 
might not be fast enough to respond and I cannot set it for phones only.

It seems I need to bring back the chan_sip behaviour ? 'do not bother 
with INVITE to Unreachable devices'.


Jacek
> On Tue, Aug 9, 2016 at 12:03 PM, Jacek Konieczny <jajcus at jajcus.net
> <mailto:jajcus at jajcus.net>> wrote:
>
>     Hi,
>
>     We have been migrating our PBX system from Asterisk 1.8 and chan_sip to
>     Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have
>     stumbled on a behaviour difference I don't like.
>
>     With chan_pjsip when a phone went unexpectedly offline (Ethernet cable
>     disconnected) Asterisk would detect this quickly (through the
'qualify'
>     pings), mark the phone as 'Unavailable' and fail immediately
with
>     'CHANUNAVAIL' when dialling this phone.
>
>     With Asterisk 13 and chan_pjsip qualify still works for determining
>     current phone availability (endpoint shown as 'Unavailable'
shortly
>     after disconnecting the cable), but the phone is being dialled like
>     nothing is wrong ? Asterisk sends the INVITE and waits for the
response,
>     until SIP timeout (a bit more than 30s total). That is much longer time
>     until 'CHANUNAVAIL' than I expect. It is also longer than the
dial
>     timeout in some cases, so I would get 'NOANSWER' instead of
>     'CHANUNAVAIL' which breaks my dialplan logic.
>
>     Is that that the expected behaviour, a bug or a configuration problem?
>     Am I supposed to check for device availability in my dialplan?
>
>     Greets,
>     Jacek
>
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Jacek Konieczny wrote:
> On 2016-08-09 10:06, Faheem Muhammad wrote:
>> trip time and Call Setup time of SIP Requests.
>> 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.
>
> No, that won't work.
>
> First ? 't1min', 'timert1' and 'timerb' are
chan_sip, not pjsip options.
>
> Second ? the 'timer_t1' and 'timer_b' settings available in
'pjsip.conf'
> are global and not per-endpoint. I cannot change T1 for trunks, as they
> might not be fast enough to respond and I cannot set it for phones only.
>
> It seems I need to bring back the chan_sip behaviour ? 'do not bother
> with INVITE to Unreachable devices'.
I'd suggest filing an issue on the issue tracker[1] for this. It's 
reasonable behavior.

[1] https://issues.asterisk.org/jira

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

On 2016-08-10 11:53, Joshua Colp wrote:
> Jacek Konieczny wrote:
>> On 2016-08-09 10:06, Faheem Muhammad wrote:
>>> trip time and Call Setup time of SIP Requests.
>>> 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.
>>
>> No, that won't work.
>>
>> First ? 't1min', 'timert1' and 'timerb' are
chan_sip, not pjsip options.
>>
>> Second ? the 'timer_t1' and 'timer_b' settings
available in 'pjsip.conf'
>> are global and not per-endpoint. I cannot change T1 for trunks, as they
>> might not be fast enough to respond and I cannot set it for phones
only.
>>
>> It seems I need to bring back the chan_sip behaviour ? 'do not
bother
>> with INVITE to Unreachable devices'.
>
> I'd suggest filing an issue on the issue tracker[1] for this. It's
> reasonable behavior.
Done:
https://issues.asterisk.org/jira/browse/ASTERISK-26281

I just wanted to make sure I am not missing something, first.

Jacek