[OpenSER-Users] Problem with "uac_replace_from()" and Linksys PAP: "vsf" parameter in Record-Route

Iñaki Baz Castillo ibc at in.ilimit.es
Thu Jan 3 11:21:50 CET 2008


Hi, I use "uac_replace_from" with:
  modparam("uac","from_restore_mode","auto")
  modparam("uac","rr_store_param","vsf")

So when a call arrives and must be sent to an Asterisk voicemail server I 
change the From:


Linksys PAP -> OpenSer

  INVITE sip:*500 at openser.ilimit.es SIP/2.0
  Via: SIP/2.0/UDP 192.168.1.109:5061;branch=z9hG4bK-91c6d79a
  From: PAP809 <sip:809 at openser.ilimit.es>;tag=bd9af3255e5ce6ao1
  To: <sip:*500 at openser.ilimit.es>
  Call-ID: 148834d2-a640054a at 192.168.1.109

OpenSer -> Asterisk

  INVITE sip:*500 at 80.94.0.111 SIP/2.0
 Record-Route: <sip:80.94.0.110;lr=on;ftag=bd9af3255e5ce6ao1;vsf=AAAAAAAAAB8AAAAAAAAAAAAAAAAAAAAAAEBvcGVuc2VyLmlsaW1pdC5lcw-->
  Via: SIP/2.0/UDP 80.94.0.110;branch=z9hG4bKb114.2e2e6ca5.0
  Via: SIP/2.0/UDP 192.168.1.109:5061;rport=5061;received=212.121.235.18;branch=z9hG4bK-91c6d79a
  From: PAP809 <sip:809_openser.ilimit.es at openser.ilimit.es>;tag=bd9af3255e5ce6ao1
  To: <sip:*500 at openser.ilimit.es>
  Call-ID: 148834d2-a640054a at 192.168.1.109

Asterisk -> OpenSer

  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 80.94.0.110;branch=z9hG4bKb114.2e2e6ca5.0;received=80.94.0.110
  Via: SIP/2.0/UDP 192.168.1.109:5061;rport=5061;received=212.121.235.18;branch=z9hG4bK-91c6d79a
  Record-Route: <sip:80.94.0.110;lr=on;ftag=bd9af3255e5ce6ao1;vsf=AAAAAAAAAB8AAAAAAAAAAAAAAAAAAAAAAEBvcGVuc2VyLmlsaW1pdC5lcw-->
  From: PAP809 <sip:809_openser.ilimit.es at openser.ilimit.es>;tag=bd9af3255e5ce6ao1
  To: <sip:*500 at openser.ilimit.es>;tag=as32a464eb
  Call-ID: 148834d2-a640054a at 192.168.1.109

OpenSer -> Linksys PAP

 SIP/2.0 200 OK
 Via: SIP/2.0/UDP 192.168.1.109:5061;rport=5061;received=212.121.235.18;branch=z9hG4bK-91c6d79a
 Record-Route: <sip:80.94.0.110;lr=on;ftag=bd9af3255e5ce6ao1;vsf=AAAAAAAAAB8AAAAAAAAAAAAAAAAAAAAAAEBvcGVuc2VyLmlsaW1pdC5lcw-->
 From: PAP809 <sip:809 at openser.ilimit.es>;tag=bd9af3255e5ce6ao1
 To: <sip:*500 at openser.ilimit.es>;tag=as32a464eb
 Call-ID: 148834d2-a640054a at 192.168.1.109


The problem is that this "200 OK" from OpenSer to Linksys PAP is not recognized by Linksys PAP
who doesn't reply with an ACK (just ignores the "200 OK"), so the "200 OK" is resent again 
and again by Asterisk.

This issue just occurs with the Linksys PAP:
  Product Name:  PAP2-NA
  Software Version:  2.0.12(LS)  Hardware Version:  0.03.4

There is no NAT problem and if I set:
  modparam("uac","from_restore_mode","none")
then the problem dissapeares, so the problem is the existence of "vsf" parameter in Record-Route.
If this parameter doesn't exist the problem doesn't occur.

Again I repeat that this issue doesn't occur with others phones. Any idea of why this occurs
with Linksys PAP? Thanks for any suggestion.

PD: Changing the parameter name does solve nothing.

PPD: What about if I don't restore the From in the replies? According to RFC3261 the only important is the
matching of From_tag, To_tag and Call-id, so, maybe I just should set:
  modparam("uac","from_restore_mode","none")
and forget this issue?


-- 
Iñaki Baz Castillo
ibc at in.ilimit.es




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