SIP calls have long pause after last digit dialed

[JPDearing55]

Hope this is a good place to post this question as I am somewhat new to the ALU system. One of our stations (a 4028) is exhibiting unusual behavior when making an outbound call. User dials 9 (for outside call) and then dials 10 digits. After the last digit is dialed it can take up to 30 seconds until the display changes to "alerted" and you start to hear audible ringback. The default outbound path is SIP and inbound calls are just fine and nobody else is reporting this issue. We also have a T1-PRI. If I override and force the call out over the PRI, the operation is normal. That is, dial the digits and 1 or 2 seconds after the last digit is dialed, the display changes and you start to hear audible ringback. This happens on ALL outbound calls from this particular station. Bad 4028, maybe?

Our SIP traffic rides on a DS-3 so bandwidth congestion isn't the issue and it only affects one station.

Any ideas or thoughts??

Thanks folks!!

[cavagnaro]

Discriminator is set to 10 or 255 on external numbering plan?

[JPDearing55]

Wouldn't the External Numbering Plan affect ALL users? This issue is with just the one station. Is there something I can look at using the 4760 for that extension that would affect this?

Thanks!

John

[cavagnaro]

Only one? It is perfect copy of others? Entity as well? Try to capture SIP logs and compare a good vs bad call

[fe80]

Same unsolved problem here with public SIP trunk. Is there any timer for the end of dialing recognition = 30?

[cavagnaro]

So...logs? How is your configuration? ARS? Network Routing?
Post all details, not just the issue...we can't help without you describing your scenario

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[fe80]

The only thing you see in in debugging is, that the INVITE for the outgoing call gets generated 30s *after* dialing the last digit. Independent using ARS or not.

[cavagnaro]

Can't be. If not using ARS means you are using Network Routing which has a fixed lenght of digits.
So there is more here which you are not mentioning. Which SIP points are you using? Or all of them are Alcatel sets?
You need to describe your scenario.... Please? Now I have to be polite to help you...

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[fe80]

@cavagnaro thanks for your answer. What do you mean with SIP points? Here is only a public SIP trunk. The sets are 8028/4028 with UANOE. How can I export relevant config options you need?

[cavagnaro]

Ok, so lets see, how is your sip configuration done to use that SIP trunk? For the behavior I guess you are using an ARS table, right?
If so, post the discriminator settings of each rule of just the one that is being used. If you don't know which try to use the "lookars i" command


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