[Asterisk-Users] Call forward & SER as SIP router

[Victor Alvarez]

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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