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No call audio after 7 calls

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FranklinAvaya

Systems Engineer
Dec 21, 2020
4
US
I've been setup with a Cisco Meraki at home from my networking team and they've setup the DHCP scope settings to match my network at work.

The phone connects to my CLAN properly and can make inbound and outbound calls, but ONLY 7 times. Pressing the speakerphone 7 times to get dial tone, then pressing it an 8th time, no dial tone. I logoff the extension and then log back in, and the phone functions for another 7 calls/dial tones. Inbound and outbound add together. If I call my cell on the 8th call, I can hear the audio coming from my desk phone on the cell, but not from the cell to the deskphone.

I also have Avaya One-X communicator running pointed to the same CLAN, and it functions properly, no 7 call limit.

I'm connecting to a CM4 system with a NAT'd address.

Any ideas?

Thanks!
 
I’d start with a list trace station to monitor the call setup and see if something weird comes by. Once the call is active, do a status station to see the audio flow and try to disable direct audio on the phone or nr to see if that helps. Perhaps display denial events too? Firmware of the phone maybe?

Freelance Certified Avaya Aura Engineer

 
Thanks for the response!

Here is the station trace:
StationTrace_ebonhs.png

The details above the red line are the 7th time that the phone has dial tone, and below the line is when there is no dial tone on the 8th attempt.

I disabled direct IP-IP Audio, and that didn't make a difference. I also compared a status station when working and not working, and don't see any difference, except for the other end port number which changed every call. I went through denial events and don't find any related to this. I've tested on a J169 and a 9630, both act the same.
 
Are you the only one having this issue? And what if you wait a while, so do not hang up when there is no dialtone. I am not sure yet where to look. Is it possible to register to another clan?

Freelance Certified Avaya Aura Engineer

 
I have 2 J169's and a softphone attached to the Meraki. I'm the only one in the organization that has them, as we are testing to hopefully get it working so others can use it. I've kept it on the hook for over 30 seconds, and it never passes dial tone. I also let the phone sit all night and then tried again in the morning and still couldn't get a dial tone until I logged off and back on again.

As far as registering to another CLAN, it took me like 3 weeks to get the networking and firewall team to get the backend routes and what not setup. It would take the same to have them set up another NAT address for my other CLAN.
 
Well, I did more testing this afternoon and it consistently stops working after 15 dial tones... So 7 for a physical phone and 15 for a softphone. Even when I use my extension on the softphone or the physical phone, it follows this logic.
 
Is there positively no H323 ALG or firewall/security appliances between the telephones and the PROCR (or maybe even your DSPs)?

Any commonality with the audio missing, like the port numbers in the trace?

 
What is the first port the phone uses after you logoff and login and ask for dialtone? And then after asking dialtone again, the port is 2 higher, right? So logoff and login resets the ports mechanism? Do these phones have their own network region? With VPN, you can try to change ‘near end establishes tcp signaling socket’ on the IP network-region. But I think the problem might be the Meraki.

Freelance Certified Avaya Aura Engineer

 
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