Alcatel Unleashed

Hi Wallace - what patch did you upgrade from/to. We're coming across a similar issue for one of our OXEs.

Thanks,
Neil

If I remember well the release was 9.1 i60516c and I upgraded to 9.1 i60537a and it is working fine with this release.

Regards
Wallace

Thanks Wallace - in the end we were advised the same by TAC.

omg! I'm having the same problem, OXE is in realease R11.0 k1.400.37, SBC is Audiocodes.
Oxe-SBC-Trunk , payload 97, G711. But still I'm having problems in incoming calls when the caller dial over the Automatic Attendant.
Any ideas?
In traces I see DTMF are repited, is it normal?

I attach you traces between SBC and OXE

No, check your network, there could be a delay

cavagnaro wrote:No, check your network, there could be a delay

Tnahk you.
Do you have some recommendations?
The customer is very "jealousy" with his network :S.

As any customer. Network guys won't lift a finger as they are so good in their job.
Ask for QoS, packet loss, jitter.
Usually double DTMF means also that not all parties involved are talking about the same setting. For example, your MG can be Inband but your SIP server can be Info or RFC2833... Or auto...só everything has to be well set up and like shinning floor...for sip to run smoothly there

Enviado de meu C6602 usando Tapatalk

Hi guys

I don´t know Audiocodes, but in the trace I can realise when systems establish the call, two INVITES are sent. I had a trouble with a customer network where there was a service called "thee way handshake". It was necessary to send three INVITE to establish a call. It doesn't seem in this case, but the trace is strange.
Considering the systems are in the same network, according to IP@, I think there is a delay as Cavagnaro wrote. Take a look at the difference of time between the first INVITE and the first ACK after 200 OK, more than 1 second. There should be miliseconds of difference in my opinion.
Look at the trace I shared, that's a call between an OXO and a SIP GSM gateway with a DTMF event. You can see the SIP signaling is very fast.

If you think there can be a problem with customer network, connect your systems directly out of the customer network and check the traces again.
If there is a problem with customer network, you need to prove him there is a problem.

Best regards.

No, there are timers, 1 second is not wow what a network, but is fine. You have to check those timers on your Audiocodes to understand why a second invite was sent.
I see you send DTMF via RTP thats why second invite. You need to understand the SIP callflow for each configuration. SIP has many variations and options will modify them. You need to learn it well.

Enviado de meu C6602 usando Tapatalk

Finally, In my case, the issue was solved by upgrade to K1.520.40.a
It never was problem from audiocodes or SIP provider.
Even, we detect the same problem in another customer with a PRI discarding SIp issues.