[asterisk-dev] branch/1.2 r 9404 SIP/Voicemail is broken..

[brett at websmyths.com]

More info...

On 2/11/2006, "brett at websmyths.com" <brett at websmyths.com> wrote:
>WOW - I just totally lost SIP.....
>It will not start unless I noload => chan_sip.so
>Something with  voicemail not being loaded....
>One phone does watch two extensions - I will change that and check again.
>
>Brett

OK - with noload => chan_sip.so - it loads and everything is good
except - well - there is no SIP...

renamed sip_notify.conf and got rid of double extension MWI (actually
one IAX2 and one SIP mailbox check)

Doing load chan_sip.so I get:

 Loaded /usr/lib/asterisk/modules/chan_sip.so => (Session Initiation
Protocol (SIP))
  == Parsing '/etc/asterisk/sip.conf': Found
    -- SIP Seeding peer from astdb: 'spa2000' at
spa2000 at 192.168.10.95:5060 for
300
  == SIP Listening on 0.0.0.0:5060
  == Using TOS bits 0
  == Parsing '/etc/asterisk/sip_notify.conf': Found
  == Registered channel type 'SIP' (Session Initiation Protocol (SIP))
  == Registered application 'SIPDtmfMode'
  == Registered application 'SIPAddHeader'
  == Registered application 'SIPGetHeader'
  == Registered custom function SIP_HEADER
  == Registered custom function SIPPEER
  == Registered custom function SIPCHANINFO
  == Registered custom function CHECKSIPDOMAIN
  == Manager registered action SIPpeers
  == Manager registered action SIPshowpeer

And the asterisk freezes - just the CLI - no asterisk -rx works but
the system does continue to process ZAP and IAX2.

No newer SVN branch updates - guess I will try to go back to r-9156.

Brett