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NCP1000 / SIP Trunk problem

NCP1000 / SIP Trunk problem

NCP1000 / SIP Trunk problem

(OP)
Hi,

I have a problem with a SIP trunk in my customers.
I provide both IPBX maintenance (NCP 1000) and SIP trunk through an Audiocodes SBC.
Before migrate the customer, i test it on my lab. It's working.
But after migrate the customers's IPBX, it's working fin, except each call is cut after few 10th seconds (sometimes 10, 20 or 30sec).
I see in the logs, during a call :
- IPBX send 200 OK
- Server send ACK
- IPBX send 200 OK
--> Then the call is cut from the IPBX. It seems a keepalive timeout from the IPBX.
I search several times in my labs, compare the conf, firmware... but i cannot find were to resolve it.
(i don't have the cut in lab, i can have a call during 10 min).

Do you have an idea for me ?

IPCMPR firmware is 7.1104 in customer and lab. I think i can upgrade in 8.0.1.23 (i can see this version in PBX maintenance console), is there anyway to upgrade wihtout moving in customer ?

Thanks.

Edit : i just see that i forgot to disable SIP ALG on my Cisco router (881), it's done but i cannot test it today.

RE: NCP1000 / SIP Trunk problem

Could be the session timer needs to be set to enable active If the audio session doesn’t get a response to options request the call could drop

This depends on if the sip trunk provider is expecting this option to be enabled

Could be alg but that normally affects setup or one way or no audio

RE: NCP1000 / SIP Trunk problem

(OP)
"Session Timer Ability" is (already) Enable (Passive).
On the customer IPBX and on the lab one :

And the refresh method is re-INVITE or UPDATE, so i think the 200 OK (INVITE) from the IPBX is not from this parameter.

For the ALG, i think that router block the 2nd 200 OK from the IPBX, provider doesn't receive it and the IPBX doesn't receive packet from the provider and cut the call.
That's not a solution to deactivate it, but maybe it can help me.
I'll try tomorrow i think.

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