Bug#299912: asterisk: malformed ack reponse messages for 200 OK go to RTP messages

[Jamie Baddeley]

Package: asterisk
Version: 1:1.0.5-2
Severity: important



-- System Information:
Debian Release: 3.1
  APT prefers testing
  APT policy: (900, 'testing')
Architecture: i386 (i686)
Kernel: Linux 2.4.27-1-k7
Locale: LANG=C, LC_CTYPE=C (charmap=ANSI_X3.4-1968)

Versions of packages asterisk depends on:
ii  asterisk-config       1:1.0.5-2          config files for asterisk
ii  asterisk-sounds-main  1:1.0.5-2          sound files for asterisk
ii  libasound2            1.0.8-3            ALSA library
ii  libc6                 2.3.2.ds1-20       GNU C Library: Shared libraries an
ii  libedit2              2.9.cvs.20040827-1 BSD editline and history libraries
ii  libgsm1               1.0.10-13          Shared libraries for GSM speech co
ii  libncurses5           5.4-4              Shared libraries for terminal hand
ii  libpq3                7.4.7-2            PostgreSQL C client library
ii  libpri1               1.0.4-2            Primary Rate ISDN specification li
ii  libspeex1             1.0.rel.4-1        The Speex Speech Codec
ii  libssl0.9.7           0.9.7e-2           SSL shared libraries
ii  libtonezone1          1:1.0.4-3          tonezone library (runtime)
ii  unixodbc              2.2.4-11           ODBC tools libraries
ii  zlib1g                1:1.2.2-3          compression library - runtime

-- no debconf information

I've had problems with Asterisk 1.0.5-2. It seems that the ack messages to a
200 OK message (i.e ack'ing a 'go to channel') message are malformed. I have
two systems, one, which works, running asterisk 1.0.2 and 1.0.5 which is
faulty. The good system registers with SER as 4748880, the bad system
registers as 4980066. Aside from that, the configurations of the two are
identical. In both tests, I am trying to call 044989644.

As the following ack messages show, the key difference is in the Contact:
field. The good system shows the calling party, the bad system shows the
called number in the Contact: field.

I downgraded the 1.0.5 system to 1.0.2 (taking care to keep the configs
untouched) and it spluttered into life. This proves the other elements of
the system are OK (i.e phones/SER etc)

Practically speaking the fault makes itself apparent by outbound calls
clearing prematurely. On a Cisco AS5300 voice gateway, once the timer
expires after 19/20 seconds, the call is cleared. The AS5300 seems to ignore
the malformed ack, and repeats the 200 OK message - presumably because it
gets no understandable Ack, it repeats itself. Inbound calls are OK.

[Good]

U 202.53.189.50:5060 -> 202.7.4.40:5060
  ACK sip:044989644@202.7.4.40:5060 SIP/2.0..Max-Forwards: 10..Via:
SIP/2.0/UDP 202.53.189.50;branch=0..Via: SIP/2.0/UDP 202.53.189.20:5
  060;branch=z9hG4bK24d4aca4..From: "4748880"
<sip:4748880@202.53.189.20>;tag=as45321fc3..To:
<sip:044989644@vixen.fx.net.nz>;tag=B44866E0-1F8
..Contact: <sip:4748880@202.53.189.20>..Call-ID:
76f3be1e13c64bee51f667794998be51@202.53.189.20..CSeq: 103
ACK..User-Agent: Aste
  risk PBX..Content-Length: 0....

[bad]

U 202.53.189.50:5060 -> 202.7.4.40:5060
  ACK sip:044989644@202.7.4.40:5060 SIP/2.0..Max-Forwards: 10..Via:
SIP/2.0/UDP 202.53.189.50;branch=0..Via: SIP/2.0/UDP 161.29.255.254:
  5060;branch=z9hG4bK78460bfc..From: "4980066"
<sip:4980066@161.29.255.254>;tag=as149ab473..To:
<sip:044989644@vixen.fx.net.nz>;tag=B44A670C-BC3
..Contact: <sip:1044989644@161.29.255.254>..Call-ID:
221912984d7c51572d53649740013819@161.29.255.254..CSeq: 103
ACK..User-Agen
  t: Asterisk PBX..Content-Length: 0....

Cheers

jamie