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[asterisk-users] one way audio after call transfer


 
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maillisting at gmail.com
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PostPosted: Wed Apr 30, 2008 11:42 am    Post subject: [asterisk-users] one way audio after call transfer Reply with quote

Hi all,

Recently, I experienced one way audio after call transfer.

incalling call (PSTN) A --> GXP2000 thro' zap --blind transfer--> destination B
Finally A and B reach each others, but there is only one way audio.
Anyone get the same experience before? How to solve the problem?

Asterisk vesion:
Asterisk 1.4.15
zaptel 1.4.7
asteriks-addon 1.4.5
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duncan at e-simple.co.nz
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PostPosted: Wed Apr 30, 2008 3:49 pm    Post subject: [asterisk-users] one way audio after call transfer Reply with quote

I had a similar issue in 1.2 after transfer and we were using SIP only
but an upgrade cured it

We are now on 1.4.18 still without issues

Cheers Duncan
Rilawich Ango wrote:

Quote:
Hi all,

Recently, I experienced one way audio after call transfer.

incalling call (PSTN) A --> GXP2000 thro' zap --blind transfer--> destination B
Finally A and B reach each others, but there is only one way audio.
Anyone get the same experience before? How to solve the problem?

Asterisk vesion:
Asterisk 1.4.15
zaptel 1.4.7
asteriks-addon 1.4.5

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maillisting at gmail.com
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PostPosted: Thu May 01, 2008 4:18 am    Post subject: [asterisk-users] one way audio after call transfer Reply with quote

Do you mean the problem is solved using asterisk 1.4.18? Are you
using the setting as mine?

Below is my setting. One way audio is a result after A & B connected.

PSTN (A)--1200P--> Asterisk --> GXP2000 --blind transfer --> Extension B

You can see that involve many parties in the blind transfer operation.
I am not sure the problem is related to 1200P, Asterisk or GXP2000.
That's why I seeking the solution from any person who touch the same
problem before.

asterisk version:
asterisk 1.4.15
zaptel 1.4.7
asterisk addons 1.4.5

On Thu, May 1, 2008 at 4:49 AM, Duncan Turnbull <duncan at e-simple.co.nz> wrote:
Quote:
I had a similar issue in 1.2 after transfer and we were using SIP only
but an upgrade cured it

We are now on 1.4.18 still without issues

Cheers Duncan
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