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Migrate acount from PAP2 to my new PBXnSIP server

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Hi

 

I am installing a new server and I would be describing each of the activities that I doing, and so you can help me if I have any problem, if everything looks good, the procedure is documented for future reference.

 

Greetings

 

First

I setup the the TRUNK with PAP2 acount and now I can make y recive calls, but when I make a call (VoIP Provider -> PBXnSIP -> IpPhone)

I do not have ringbacktone.

 

Any idea, what i can do?

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I am attaching the log of the tracert call to analize

 

Can you try changing the "Ringback" trunk setting (at the bottom of the trunk edit page) and see that changes the behavior that you are looking for? May be the VoIP provider is ignoring the '183 Media'.

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Can you try changing the "Ringback" trunk setting (at the bottom of the trunk edit page) and see that changes the behavior that you are looking for? May be the VoIP provider is ignoring the '183 Media'.

 

DING DING DING :wacko:

 

Its works, and weird, because, when i call from a number of the same PSTN, the call have the ringback, but calling from other PSTN to my number they don;t have.. but it is fixed, turning off the 183 media option..

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Next.. in this process

 

The call only lasts 11 minutes with 11 seconds

I check the value of "Maximum call duration" and it is 7200

 

Who disconnects the call - the service provider or the PBX? (Does the PBX send the BYE request or the "200 Ok" response?)

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Here the log..

 

 

416 685.530770 205.240.200.62 192.168.1.3 SIP Status: 200 OK (1 bindings)

417 685.644603 205.240.200.62 192.168.1.3 SIP/SDP Request: INVITE sip:5045403107@192.168.1.3:5060, with session description418

418 686.470395 192.168.1.3 205.240.200.62 SIP Request: REGISTER sip:g1sbc.cablecolor.hn

419 686.514361 205.240.200.62 192.168.1.3 SIP Status: 200 OK (1 bindings)

420 689.644800 205.240.200.62 192.168.1.3 SIP/SDP Request: INVITE sip:5045403107@192.168.1.3:5060, with session description

421 693.644257 205.240.200.62 192.168.1.3 SIP Request: BYE sip:5045403107@192.168.1.3:5060

422 693.647387 192.168.1.3 205.240.200.62 SIP Status: 200 Ok

423 697.440785 192.168.1.3 205.240.200.62 SIP Request: REGISTER sip:g1sbc.cablecolor.hn

424 697.554207 205.240.200.62 192.168.1.3 SIP Status: 200 OK (1 bindings)

425 700.410423 192.168.1.3 205.240.200.62 SIP Request: REGISTER sip:g1sbc.cablecolor.hn

Llamamada_Cortada.zip

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421 693.644257 205.240.200.62 192.168.1.3 SIP Request: BYE sip:5045403107@192.168.1.3:5060

422 693.647387 192.168.1.3 205.240.200.62 SIP Status: 200 Ok

 

Well that looks like your service provider disconnects the call... It is really strange after 11 minutes. Maybe they have a 11-minute demo key from their softswitch vendor ...

 

No it seems that the re-INVITE fails. Looking into it... Can you send us a private mail with your account information so we can give it a try from here?

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Well that looks like your service provider disconnects the call... It is really strange after 11 minutes. Maybe they have a 11-minute demo key from their softswitch vendor ...

 

No it seems that the re-INVITE fails. Looking into it... Can you send us a private mail with your account information so we can give it a try from here?

 

 

I send to you the account information

post-362-1226091739_thumb.png

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I send to you the account information

 

 

I try making a call with the PAP2, with the same configuration IP, account etc..

And the call never disconect until to hang up the phone.

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Next.. in the process

 

I having a response 487 Request canceled ONLY IN some destination.

 

Just verify whether you have multiple registrations on those destination. If one of the UA answers, then others will get CANCEL and this triggers a '487 Request Terminated' message.

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before the PBX age the provider use a 4 PAP2 with 2 phone line each PAP2.

Now I setup all this lines as trunks (SIP registration) in the PBXnSIP.

 

You think the problem is to have several SIP registrations from the PBXnSIP to SoftSwitch?

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