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davidnewt

join:2009-08-10
Conroe, TX

[Equipment] SPA3000 dialout time delay

I am trying to use my old SPA3000 to connect PSTN line and voip line. It works ok with the following dialplan:

L:10,S:3,(5xxS0|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|00xx.<:@gw1>|<011:00>xx.<:@gw1>|[2-9]11S0<:@gw0>|<*,:>xx.<:@gw0>|[x*][x*].)

But there is a problem: the dialout (e.g. 501) for voip takes 15 seconds to reach out the voip proxy server, even putting # after the number not working, while the gw0 (pstn) can go out without delay. It seems that S0 or # after dialed number is not effective for voip calls. Can anyone know what is the cause for the delay of voip calls while no problem for pstn calls? Anyway to solve this problem? Thanks a lot in advance.

Stewart

join:2005-07-13
kudos:25

1 edit

Refresh the Info page one second after dialing 50 and confirm that the Call 1 State is Dialing. Then immediately press 1 and refresh again; it should now show Connecting. If not, report whether pressing # will cause it to advance to Connecting.

Assuming the above is ok, use SIP Debug to see where the delay lies. One possibility is DNS lookup, e.g. Primary server times out and it queries the Secondary. If that's the trouble, set up parallel lookup and/or change the servers. STUN lookup could also cause this trouble -- turn STUN off if not needed, or use a different STUN server.

If the initial INVITE does get sent quickly after dialing the last digit, post the SIP timing; the trouble could be at the server end.


davidnewt

join:2009-08-10
Conroe, TX

said by Stewart:

STUN lookup could also cause this trouble -- turn STUN off if not needed to use a different STUN server.

Thanks a lot, Stewart. Following your instruction, I found out the issue is indeed from stun server. After I switch stun.xten.com to stun.ekiga.net, the call goes out immediately.