Here is a SIP Trace of my Call Pickup<br><br><a href="http://pastebin.com/beARSgmj">http://pastebin.com/beARSgmj</a><br><br>If you do a search for <b>Replaces</b> you will see the INVITE that is sent in order to do a Call Pickup of the call coming in from the SIP Provider.<br>
<br><br><br><div class="gmail_quote">On Wed, Jun 6, 2012 at 9:26 PM, osiris123d <span dir="ltr"><<a href="mailto:duane.larson@gmail.com" target="_blank">duane.larson@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
In the past I used a VoIP SIP trunk provider that I guess used Asterisk<br>
servers as their gateways and I was able to perform the Call Pickup feature<br>
with OpenSIPS and the SIP Trunk Provider. Now I have a new provider and I<br>
need to use the B2B modules for Transfers. Currently I am not able to<br>
perform Call Pickup with calls coming from the PSTN to internal customers.<br>
I tried to use the b2b_bridge_request but when the PSTN Caller calls<br>
internal_user_A and internal_user_B tries to do a call pickup I see the<br>
following error in syslog<br>
<br>
ERROR:b2b_logic:b2bl_bridge_msg: Wrong state for entity ek= [B2B.106.390],<br>
tk=[536.0]<br>
<br>
When I do an "opensipsctl fifo b2b_list" all the states are zero.<br>
<br>
So does the state of the call have to be a "on call" state instead of a<br>
"calling" state? How can you bridge a call that is in process of ringing?<br>
<br>
--<br>
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<br>
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</blockquote></div><br><br clear="all"><br>-- <br>--<br>*--*--*--*--*--*<br>Duane<br>*--*--*--*--*--*<br>--<br>