SIP Ext GW problem
Posted: 06 Jun 2018 07:42
HI guys
I need some assistance with SIP External gateway.
I have 2 VOIP providers on the PABX. Different SIP gateway setups. One works and the other one doesn't.
The problematic one authenticates on username and password, and only allows numbers within their number range in correct format.
There are speech problems.
If there is speech the call cuts after 20 seconds. When there is no speech the call cuts after 5 seconds.
On both traces I can see I am not getting a 200 OK from them, indicating the call is established. They can not tell me why. And I suppose that is why the call cuts and eventually then gives congestion.
From IP phone Every second or third call has speech. When connecting on Analog phone there is speech every time but call also cuts.
On incoming call there is one way speech.
My IP domain is on G7.29. I have supplied them with all RTP ports but they say they are allowing all RTP traffic at the moment but will set up QOS for the RTP ports on the router.
The VOIP provider has the attitude of their equipment works and I need to fix my problem.
Every time I make a few changes on my sip Gateway to test and run traces they block my IP address for 30 minutes.
Can someone please assist?
Please note there are 6 sip gateways on this pabx so on the trace there is registration 200 ok message.
Martin
I need some assistance with SIP External gateway.
I have 2 VOIP providers on the PABX. Different SIP gateway setups. One works and the other one doesn't.
The problematic one authenticates on username and password, and only allows numbers within their number range in correct format.
There are speech problems.
If there is speech the call cuts after 20 seconds. When there is no speech the call cuts after 5 seconds.
On both traces I can see I am not getting a 200 OK from them, indicating the call is established. They can not tell me why. And I suppose that is why the call cuts and eventually then gives congestion.
From IP phone Every second or third call has speech. When connecting on Analog phone there is speech every time but call also cuts.
On incoming call there is one way speech.
My IP domain is on G7.29. I have supplied them with all RTP ports but they say they are allowing all RTP traffic at the moment but will set up QOS for the RTP ports on the router.
The VOIP provider has the attitude of their equipment works and I need to fix my problem.
Every time I make a few changes on my sip Gateway to test and run traces they block my IP address for 30 minutes.
Can someone please assist?
Please note there are 6 sip gateways on this pabx so on the trace there is registration 200 ok message.
Martin