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Hi,
I'm trying to transfer an incoming call
from the PSTN to another PSTN number through a SER - Asterisk system. SER
doing only routing..
pstn call-> SER -> asterisk (call
forward) -> SER -> pstn
Logic for SER: If something comes from the
pstn, send it to asterisk. If something comes from asterisk, send it to the
pstn.
Every time I am getting a "Got SIP response
481 "Invalid CSeq Number back from" SER. And the call terminates.
Canreinvite makes a small difference here, If I have canreinvite=yes, I am able
to talk only in one direction and for a few seconds. With
canreinvite=no, CSeq error appears in the very moment you pick up the
phone. So every time the phone rings but It is not possible to
talk.
At this point, I must confess I am lost.
First, I don't know if this loop is possible (pstn call-> SER ->
asterisk -> SER -> pstn), I tried it with two SER machines (pstn
call-> SER1 -> asterisk -> SER2 -> pstn) getting the same
result, CSeq comes from SER1. If it is possible, I don't know the issues with
this configuration. The forwarding works fine internally, I mean, extension 22
calling 25 which is forwarded to my mobile phone. Problem comes when It is a
pstn number calling 25. The connection pstn->SER->asterisk-> UA is also
perfect. I never had any problem transferring calls through asterisk, it seems
that, for some reason, things get worse when SER is an intermediary in the
communication.
Could anybody help me here, please?
Victor.
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