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

[Bin Zhang]

SER can not receive PSTN call directorily.....

On 8/9/05, Victor Alvarez <victor at sentidocomun.es> wrote:
> 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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-- 
Bin Zhang