<br><div class="gmail_quote"><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Hi,<br><br>Can you tell me what advantage on INVITE has against REFER?<br>
<br>REFER is a kind of blind transfer and INVITE in sipXbridge is 3PCC?<br><br>Thanks in advance.<br><font color="#888888"><br>Youngjin</font><div><div></div><div class="h5"><br><br><div class="gmail_quote">On Sun, Feb 27, 2011 at 6:40 AM, Grzegorz Stanislawski <span dir="ltr"><<a href="mailto:stangrze@netitel.pl" target="_blank">stangrze@netitel.pl</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Hi.<br>
We have proverb for this, i don't know english version but it goes like this:<br>
"When it isn't known what it's all about, it's about money"<br>
<br>
Your ITSP had troubles with proper handling second transfer for billing purposes so decided to disable it.<br>
Proxy doesnt participate in call transfer, but ITSP it must charge users properly: Alice should pay just for call to Bob, Bob for "his" call to Charlie and so on.<br>
If You are using sipX You should use sipXbridge, it replaces REFER with INVITE and bridges calls.<br>
<br>
Grzegorz Stanislawski<br>
<br>
<br>
<br>
W dniu 2011-02-25 11:40, niklas rehnberg pisze:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi,<br>
Thank for the quick response.<br>
The issue occur only when the Alice is a PSTN client.<br>
My ITSP says that they only supporting one call transfer (very strange).<br>
They can not explain why etc...<br>
PSTN client: Alice<br>
MGW/MGC(ITSP): Cisco/SER<br>
Our sip server: SIPX<br>
<br>
BR Niklas<br>
2011/2/25 Iñaki Baz Castillo <<a href="mailto:ibc@aliax.net" target="_blank">ibc@aliax.net</a> <mailto:<a href="mailto:ibc@aliax.net" target="_blank">ibc@aliax.net</a>>><br>
<br>
2011/2/25 niklas rehnberg <<a href="mailto:niklas.rehnberg@gmail.com" target="_blank">niklas.rehnberg@gmail.com</a><br>
<mailto:<a href="mailto:niklas.rehnberg@gmail.com" target="_blank">niklas.rehnberg@gmail.com</a>>>:<br>
> Hi,<br>
> Have following issue:<br>
><br>
> Alice calling Bob.<br>
> Bob make call transfer to Charlie (works fine)<br>
> Charlie transfer Alice to David. (the call break)<br>
><br>
> Why is not possible to transfer the call more than one time?<br>
> Is it any parameters?<br>
><br>
> My ITSP use SER together with Cisco MGW.<br>
<br>
Niklas, nothing in SIP protocol neither in SER/Kamailio makes your<br>
scenario to fail. It must be a problem in your custom setup. Try<br>
identifying the problem capturing SIP traces.<br>
Also take into account that a proxy doesn't participate at all in the<br>
process of a "call transfer". It's just a transparent mechanism for a<br>
proxy.<br>
<br>
--<br>
Iñaki Baz Castillo<br>
<<a href="mailto:ibc@aliax.net" target="_blank">ibc@aliax.net</a> <mailto:<a href="mailto:ibc@aliax.net" target="_blank">ibc@aliax.net</a>>><br>
<br>
<br>
<br>
<br>
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<br>
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