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Registration Date: 01.01.1970
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verrückt unable to answer incoming call using ata extension but can answer using xlite Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Hi there!

I have three different Linksys / Sipura ATA's that I have configured as extensions. They register and I can dial out OK from them, through PBXes and any of three trunks.

I have a DID hosted with Ozsite in Toowoomba, Australia. I have setup the ozsite sip account associated with this DID as a trunk.

Calls to my DID can be answered OK from the extensions that are using Xlite, Xpro, Eyebeam, or any softphone.

But if I try to answer calls on any one of the ATA's when they ring then it is as if the handset was never picked up. My softphones and any other ATA just keep ringing.

So.... I can try to answer a call on the phone connected to an ATA when it is ringing. It stops ringing, but there is no audio. Meanwhile, my softphones and other ATA keep ringing. If I then try to answer the call on a softphone it answers, and the call goes through perfectly.

Thinking it was a problem with the ATA configuration led me to buying a new ATA SPA2102 today. With the factory default configuration, PBXes setup in line1, it has the same problem as my other three ATA's !!!!!

Is the problem with PBXes, Ozsite (the DID provider) or with the ATA configuration?

Hope you can help!

06.02.2008 12:05 endtech is offline Search for Posts by endtech Add endtech to your Buddy List
i-p
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Pfeil RE: unable to answer incoming call using ata extension but can answer using xlite Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Did you check the System Log of PBXes for these events?

06.02.2008 17:15 i-p-tel is offline Search for Posts by i-p-tel Add i-p-tel to your Buddy List
Dia
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Registration Date: 03.03.2006
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The way you describe the issue (without looking at a trace), it seems there a vocoder incompatibility between the SIP Proxy of Ozsite, and the supported vocoders of PBXes and the SPA devices.

Ensure the SIP Proxy of Ozsite includes in the SDP of the SIP Invite the G.711 vocoder as one of the offered vocoders.

06.02.2008 18:26 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
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fault appears to be with provider of did or subsequent upstream provider Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Hi, further to my original post

A fault appears to exist with the DID provider, previously Ozsite, now called Maxo.

If I configure any of my ATA's to use the DID provider directly then it still doesn't work properly.

While it doesn't exhibit exactly the same symptoms as when I use the provider through a PBXes trunk, the fact is that there is still clearly a fault.

The curious part of this fault is that the calls to the DID works perfectly when using a softphone, but generally don't work via a Sipura / Cisco / Linksys ATA.

thank you iptel and diafora for your comments.

From here on my standard troubleshooting step (obvious to many) will be to spend a few days testing any trunks or sip providers directly, well before trying to integrate them into PBXes, to be sure there are no intermittent faults.

:-)

22.03.2008 11:36 endtech is offline Search for Posts by endtech Add endtech to your Buddy List
Dia
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Troubleshooting tips Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

Endtech, thank you for your kind comments. A few points in your troubleshooting quest with DIDs and SIP trunks in general.

With the advent of the System Log in Premium accounts, troubleshooting intermittent issues with SIP trunks has become easier, since we now have access to a log over a period of time.

The equivalent setup to monitor any intermittent issue overtime using an ATA, would involve the use of either a Syslog server or an ethernet packet trace.

Regarding the setup of your SPA, in the SIP tab set the RTP Packet Size: to 0.020 since the default value of 0.030 is not quite compatible with every SIP proxy.

23.03.2008 19:26 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
 
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