When we were on 4.2.1, we had it limited to g711. With 4.4, we are able to change it back to allow g729. Something changed in sipx to resolve this issue for us. This has been consistent in all our setups. The problem is the same when they were 4.2.1 and since going to 4.4.
The problem did not occur when the site was initially on auto attendant. They decided they wanted to have a human answer the phone, and that is when the problems started.
I didn't know if this pointed to a hardware issue at the phone, or something else. I'm waiting for them to report the next occurrence so I can try to isolate it in the logfiles.

On 5/17/2011 12:34 PM, Todd Hodgen wrote:

I would be surprised if you have just one issue here.   I think looking at logs will give you the best clues as to what might be going on.

 

Have you considered limiting Codecs?  Tried sending the calls to an auto attendant to see what results you get with that?

 

From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew Kitchin (public/usenet)
Sent: Tuesday, May 17, 2011 9:02 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Need some help, dropping calls, paging locking up

 

One more additional note. I forgot about this one setting. The initial call here is routed to a phantom extension that then uses scheduling and call forwarding to route to different handsets or hunt groups depending on the time of day. These problems did occur before the somewhat complicated scheduling was put in place. 10 is an extension, and 90 is a hunt group. 10 was a member of 90. I just removed that, because I remember some issues kind of similar where problems occurred when an extension was dialed back to back using automated methods like this.



On 5/17/2011 10:39 AM, Matthew Kitchin (public/usenet) wrote:

This is a fun one. I need some pointers on where to start.
I have about 15 sites that are all roughly identical in their setup, so I know the general design is ok.
This particular site was Sipx 4.2.1 and Polycom 450/550 firmware 3.2.4 and bootrom 4.3 up until last week.
They are now sipx 4.4 (latest) Polycom 450/550 firmware 3.2.5 and bootrom 3.2.5
Cisco 2960 POE 24 port switch
Sipxbridge, Verizon VoIP services, no IP address NATing
Verizon MPLS T1
Everything is QOSd properly. There are no errors on the router or the switch interfaces.
Unfortunately, the problems remain even after the Sipx and Polycom upgrade.
There are 16 Polycom 450s and 1 550. The 550 is the receptionist, and she answers the bulk of the calls. They were on an auto attendant, but decided they wanted a live person answering the calls. That is when my problems started.
The path of most calls is answer, park with blind transfer, page with paging group, user retrieves call from park on one of the 450s.
From everything we can tell, they are doing this properly. They are experiencing 2 major problems. I am not sure if one is causing the other or if they are related.

Problem 1, straight from the Polycom 550 user "About 7-8 times a day when the main line rings, I go to answer it and no one is there, but on the LED screen the first button says “answer”. You cannot pick up the call at all and the caller hangs up and calls back."

Problem 2, also from the 550 user "about 50% of the time the overhead can’t be heard, then when you try to use the overhead a second time it rings busy"
She is referring to the the paging group. Nobody hears anything. The paging call then sits in the active calls list within sipx indefinitely. Usually, I will have to restart the paging service to get everything going again.

This site seems like a simple setup to me. I have several others that are identical. I have tried everything I can think of, so I'm ready to start at square 1 again. Can anyone give me some tips?

Thanks as always,
Matthew

 


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