Where to set lag latency (qualify=)

[krzykat]

I'm working on a project to report extension changes in registration. To that end, I want to test for when device gets lagged (default = 2000ms). I know you used to be able to go to SIP and change qualify = yes to qualify = 120 for example, and then if it was over 120ms, it would register lagged. I still see this in sip_extensions_additional, but not sure where it is in FreePBX any longer. Additionally, where would this setting be for PJSIP ?

 

[krzykat]

Well, its where its supposed to be on SIP, but where is it for PJSIP ?

 

[KNERD]

How about in the extension -> Advanced -> Qualify Frequency ?

 

[dallas]

How about in the extension -> Advanced -> Qualify Frequency ?

That setting is in seconds and is to check the extension is still reachable.

 

[KNERD]

To me that is what he is describing what he want to check for.

 

[krzykat]

How about in the extension -> Advanced -> Qualify Frequency ?

No - this say's how often is the extension verified as being available. I'm asking about the latency. In SIP, it is defined as sip notify = Yes (or sip notify = 2000) is the same thing. You are correct that it is under advanced under extension for SIP, but I don't see that for PJSIP.

 

[dallas]

No - this say's how often is the extension verified as being available. I'm asking about the latency. In SIP, it is defined as sip notify = Yes (or sip notify = 2000) is the same thing. You are correct that it is under advanced under extension for SIP, but I don't see that for PJSIP.

Have you compared SIP and PJSIP extensions in sip_extensions_additional? My initial thought was that FreePBX just combined the SIP setting of Qualify and Qualify Frequency into the one field for PJSIP.

 

[KNERD]

Here is a suggestion. Make a chan_sip extension with the sip notifiy, and any other settings you want.

Inside of the Asterisk source, there is a script for converting chan_sip to PJSIP.

Go find the extension you created in the extensions.conf file of your PBX, copy that over to a new file, and run that PSJIP converter script through it to see what comes out.

 

[krzykat]

Have you compared SIP and PJSIP extensions in sip_extensions_additional? My initial thought was that FreePBX just combined the SIP setting of Qualify and Qualify Frequency into the one field for PJSIP.

sip_additional.conf doesn't have PJSIP extensions. It seems the comparable PJSIP file would be pjsip.endpoint.conf - but I don't see any references to qualify in it. There's a field called timers=yes ... is this possible it? No clue.

 

[dallas]

I don't see any references to qualify in it.

qualify_frequency is in pjsip.aor.conf

 

[krzykat]

qualify_frequency is in pjsip.aor.conf

There are 2 items here and I think there is some confusion:

1) Qualify Frequency - this is how often asterisk will check to see if the extension is still reachable or the AOR in PJSIP. This part is known and not what I'm looking for.

2) Qualify - in SIP this is often set to Yes - meaning YES - we will check if this is available. If you set this to a number, then this number is the latency amount that asterisk determines is acceptable as "reachable" - the default is 2000 ms. You can change this number to something such as 500 - becoming more strict one what is reachable or not.

Qualify frequency, is known and I can find that line in both, but the second item, qualify, I can't locate for PJSIP.


If I'm mistaken, I'm open.

 

[KNERD]

There are 2 items here and I think there is some confusion:

1) Qualify Frequency - this is how often asterisk will check to see if the extension is still reachable or the AOR in PJSIP. This part is known and not what I'm looking for.

2) Qualify - in SIP this is often set to Yes - meaning YES - we will check if this is available. If you set this to a number, then this number is the latency amount that asterisk determines is acceptable as "reachable" - the default is 2000 ms. You can change this number to something such as 500 - becoming more strict one what is reachable or not.

Qualify frequency, is known and I can find that line in both, but the second item, qualify, I can't locate for PJSIP.


If I'm mistaken, I'm open.

I think you are looking for

aor/authenticate_qualify= yes

If you do :

pjsip show qualify aor <extension #>

Will output something like :

AOR ‘7005’
Qualify frequency : 30 sec
Qualify timeout : 0 ms
Authenticate qualify : no

 

[krzykat]

That is the closest thing I've heard yet. Will investigate

Thanks

 

[dallas]

from https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample
so probably no cigar.

;authenticate_qualify=no        ; Authenticates a qualify request if needed
                                ; (default: "no")

 

[billsimon]

There is a well-written wiki that provides fine documentation on all the options: https://wiki.asterisk.org/wiki/disp...nfiguration_res_pjsip-contact_qualify_timeout

 

[krzykat]

There is a well-written wiki that provides fine documentation on all the options: https://wiki.asterisk.org/wiki/disp...nfiguration_res_pjsip-contact_qualify_timeout

Thank You - not sure why I didn't find this.