asterisk users - Aug 2005 - Call forward & SER as SIP router

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

On 8/9/05, Victor Alvarez <victor@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