[Freeswitch-users] Sofia invite issues

Peter Olsson peter.olsson at visionutveckling.se
Thu Jun 9 11:18:55 MSD 2011 

Please pastebin a complete debug log, with "sofia global siptrace on", to http://pastebin.freeswitch.org.

I'm pretty sure this is not a bug in FS, if it was, we would have lots of people complaining...

/Peter


-----Ursprungligt meddelande-----
Från: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] För Kurtis Heimerl
Skickat: den 8 juni 2011 23:32
Till: freeswitch-users at lists.freeswitch.org
Ämne: [Freeswitch-users] Sofia invite issues

Hello Freeswitch-users!

I'm currently writing a SIP middlebox that intercepts sip messages and
changes their username. The reasons for this are complicated and
somewhat outside of the scope of this discussion. However, I've come
upon a very strange issue: when making a phone-to-phone call across
freeswitch; it is connecting the wrong user. I'm fairly well convinced
this is a bug, but I thought I'd send the issue here and see if it's
anything I'm obviously doing wrong.

Basically, FS sees a SIP message from a registered UA (call it A)
inviting another registered user to a call (B). This is acked
correctly (a TRYING message). FS then responds by inviting B into a
call with some OTHER user C. C is not mentioned at any point in the
initial SIP messages (as verified by wireshark). Even more strangely,
if I remove user C from FS (by removing their config file) FS responds
to the invite by inviting B into a call with itself. In each case, the
appropriate from header in the invite should be the original caller A.
 It's worth noting that I am not monkeying with the RTP packets at
all, but my understanding is that SIP signalling shouldn't be affected
by that.

My general guess is that I'm messing up the naming somehow, and FS is
running an algorithm to guess at who the call originator is. However,
the naming must be roughly correct; Asterisk is able to handle this
call just fine. I've included a sip trace of the second situation (A
calls B, FS invites B to a call with itself)

For the record A:1300
B:1301
C:IMSI641104878332498

REGISTER MESSAGE (Which works fine)

REGISTER sip:192.168.1.144 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
To: 1300 <sip:1300 at 192.168.1.144>
Call-ID: 1032827938 at 192.168.1.144
CSeq: 91 REGISTER
Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
Max-Forwards: 70
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347
From: 1300 <sip:1300 at 192.168.1.144>;tag=ftoui
To: 1300 <sip:1300 at 192.168.1.144>;tag=jU64NXypQc57F
Call-ID: 1032827938 at 192.168.1.144
CSeq: 91 REGISTER
Contact: <sip:1300 at 192.168.1.144:5063>;expires=7200
Date: Wed, 08 Jun 2011 21:02:04 GMT
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0

SIP TRACE (A -> B, B -> B response)

INITIAL INVITE
INVITE sip:1301 at 192.168.1.144 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
To: <sip:1301 at 192.168.1.144>
Call-ID: 1817795092 at 192.168.1.144
CSeq: 485 INVITE
Contact: <sip:1300 at 192.168.1.144:5063>;expires=3600
Content-Type: application/sdp
Max-Forwards: 70
Content-Length: 143

INVITE ACK
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974
From: 1300 <sip:1300 at 192.168.1.144>;tag=bgdqx
To: <sip:1301 at 192.168.1.144>
Call-ID: 1817795092 at 192.168.1.144
CSeq: 485 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
Content-Length: 0

FS INVITE (Note the from address being wrong)
INVITE sip:1301 at 192.168.1.144:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.144;rport;branch=z9hG4bKeeQvcyZ70SDUg
Max-Forwards: 69
From: "1301" <sip:1301 at 192.168.1.144>;tag=rHyS0Z3B61arN
To: <sip:1301 at 192.168.1.144:5063>
Call-ID: 84df47f9-0cb7-122f-13b5-5cff350d9de5
CSeq: 13447852 INVITE
Contact: <sip:mod_sofia at 192.168.1.144:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
sla, include-session-description, presence.winfo, message-summary,
refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 315
X-FS-Support: update_display
Remote-Party-ID: "1301"
<sip:1301 at 192.168.1.144>;party=calling;screen=yes;privacy=off

USER CONFIGURATIONS

1300.xml
<include>
  <user id="1300" cidr="192.168.1.0/24,127.0.0.1/32">
    <params>
      <param name="accept-blind-reg" value="true"/>
      <param name="accept-blind-auth" value="true"/>
      <param name="vm-password" value="1000"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1300"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="1300"/>
      <variable name="effective_caller_id_number" value="1300"/>
      <variable name="outbound_caller_id_name" value="1300"/>
      <variable name="outbound_caller_id_number" value="1300"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

1301.xml
<include>
  <user id="1301" cidr="192.168.1.0/24,127.0.0.1/32">
    <params>
      <param name="accept-blind-reg" value="true"/>
      <param name="accept-blind-auth" value="true"/>
      <param name="vm-password" value="1000"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1301"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="1301"/>
      <variable name="effective_caller_id_number" value="1301"/>
      <variable name="outbound_caller_id_name" value="1301"/>
      <variable name="outbound_caller_id_number" value="1301"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

IMSI641104878332498.xml
<include>
  <user id="IMSI641104878332498" cidr="192.168.1.0/24,127.0.0.1/32"
number-alias="1302">
    <params>
      <param name="accept-blind-reg" value="true"/>
      <param name="accept-blind-auth" value="true"/>
      <param name="vm-password" value="1000"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="1302"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="IMSI641104878332498"/>
      <variable name="effective_caller_id_number" value="1302"/>
      <variable name="outbound_caller_id_name" value="IMSI641104878332498"/>
      <variable name="outbound_caller_id_number" value="1302"/>
      <variable name="callgroup" value="techsupport"/>
    </variables>
  </user>
</include>

DIALPLAN

    <extension name="Local Numbers">
      <condition field="destination_number" expression="^130[0-9]$">
	<!--<action application="bridge"
data="user/${user_data(${destination_number}@${domain_name} attr
id)}"/> -->
	<action application="bridge" data="user/${destination_number}"/>
      </condition>
    </extension>

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