Delayed Incoming Calls

I have following setup…

[img]http://i54.tinypic.com/rhtkc0.jpg[/img]

Asterisk Server, SPA8000 and SPA504G are all in local LAN series ip.

I see a delay when an incoming call arrives at the reception. And sometimes the call quality deteriorates.
Is this is something related to the…

  1. The LAN network
  2. TE121 card mounted in Asterisk Server
  3. ADTRAN gateway settings

Does anyone has any idea?

Thanks!

This system looks complex and the diagram is over-simplified. I can’t tell which traffic is T1, which SIP, and which, maybe, something else again. The Adtran appears to have SIP, T1 and FXO interfaces. (The TE121 is a low end Digium Primary Rate (T1(/E1?) inteface.)

You need to run timestamped traces to determine where the delay is.

You need to identify the version of Asterisk in use.

Poor quality tends to be the result of either network overload, or, VM artifacts.

PS Next time, don’t let line art like this get anywhere near JPG. JPG is just not suited to line art. Any reasonable line art package will be able to output PNG, or at least TIFF, which can be converted to PNG.

Hey David,

Yes, the ADTRAN has SIP configuration, a PRI T1 port from ADTRAN is connected to Asterick Server at TE121. These are configured as DAHDi channels. I have connected asterisk server, spa8000s and spa504g to one seperate network in order to make it distinct collision and broadcast domain to save excess routes which I think may contribute to delays. I am using Asterisk 1.6.2.17.2.

How do I trace this delay? How do I run the timestamp trace?

Edit logger.conf, and make sure that all relevant categories are going to a suitable file (I’d normally use the “full” file)

I would advise also turning on millisecond timing.

Set appropriate debug and verbose levels.

Then run the system.

You may have to turn on sip debug.

You may have to turn on dahdi debugging, although I’m not familiar with that.

Why do you have SIP going to Adtran then being converted to T1 then going to Asterisk? Cant you just pass the SIP right to Asterisk?