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Summary:ASTERISK-02883: Asterisk does not register with SIP proxies unless a "sip reload" command is issued
Reporter:Ben Brown (benbrown)Labels:
Date Opened:2004-11-24 15:37:07.000-0600Date Closed:2008-01-15 15:15:08.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Subscriptions
Versions:Frequency of
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Environment:Attachments:
Description:I am unable to register with SIP proxies after loading Asterisk unless I issue a "sip reload" command.  I've tried numerous configurations etc. from lists and irc, but the fact that it isn't registered before a sip reload and is afterwards makes me think it maybe a bug with due to my configuration.

****** ADDITIONAL INFORMATION ******

Example: (note, i've added a few extra variables to try and help with debugging).

server*CLI> sip show registry
Host                           Username   Refr Exp Tmout State
sipgate.co.uk:5060             530xxx7     120   -1   -1    0
sipgate.co.uk:5060             143xxx8     120   -1   -1    0
sipgate.de:5060                533xxx5     120   -1   -1    0
server*CLI> sip reload
Reloading SIP
 == Parsing '/etc/asterisk/sip.conf': Found
server*CLI> sip show registry
Host                           Username   Refr Exp Tmout State
sipgate.co.uk:5060             530xxx7     105   44   -1    3
sipgate.co.uk:5060             143xxx8     105   45   -1    3
sipgate.de:5060                533xxx5     105   48   -1    3

This problem works with any SIP proxy not just sipgate.
Comments:By: Brian West (bkw918) 2004-11-24 15:42:14.000-0600

Ok this IS NOT A BUG... and your output doesn't match up with latest cvs head or stable.

Host                            Username       Refresh State
iptel.org:5060                  bkwest             585 Registered
proxy01.sipphone.com:5060       17473861189       3585 Registered
fwd.pulver.com:5060             51991             3585 Registered

This is what proper output should look like on a sip show registry
By: Ben Brown (benbrown) 2004-11-24 15:44:34.000-0600

I guess just to be clear (my note obviously wasn't!) the output of sip show registry looked like:

server*CLI> sip show registry
Host                            Username       Refresh State
sipgate.co.uk:5060              530xxx7            105 Unregistered
sipgate.co.uk:5060              143xxx8            105 Unregistered
sipgate.de:5060                 533xxx5            105 Unregistered
server*CLI> sip reload
Reloading SIP
== Parsing '/etc/asterisk/sip.conf': Found
server*CLI> sip show registry
Host                            Username       Refresh State
sipgate.co.uk:5060              530xxx7            105 Registered
sipgate.co.uk:5060              143xxx8            105 Registered
sipgate.de:5060                 533xxx5            105 Registered

edited on: 11-24-04 15:52
By: Brian West (bkw918) 2004-11-24 16:53:37.000-0600

I can't reproduce this issue.. could it be a dns issue maybe.. are srvlookups on or off?
By: Ben Brown (benbrown) 2004-11-25 13:44:45.000-0600

I'm digging around and it seems to be falling over in chan_sip.c in the create_addr function for some reason.  It gets to this point:

 if ((p->addr.sin_addr.s_addr || p->defaddr.sin_addr.s_addr) &&
         (!p->maxms || ((p->lastms > 0)  && (p->lastms <= p->maxms)))) {

But:

addr.sin_addr.s_addr: 156175065
defaddr.sin_addr.s_addr: 0
maxms: 2000
lastms: 0

after a sip reload:

addr.sin_addr.s_addr: 156175065
defaddr.sin_addr.s_addr: 0
maxms: 2000
lastms: 42

So because it has a lastms of zero it isn't sending a register.  If I fudge this line to p->lastms >= 0 it all works ok.

edited on: 11-25-04 14:32

edited on: 11-25-04 15:01
By: Mark Spencer (markster) 2004-11-25 17:11:37.000-0600

Fixed in CVS
By: Russell Bryant (russell) 2004-11-25 20:55:02.000-0600

fixed in 1.0 - will be in 1.0.3
By: Digium Subversion (svnbot) 2008-01-15 15:15:07.000-0600

Repository: asterisk
Revision: 4347

U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r4347 | markster | 2008-01-15 15:15:07 -0600 (Tue, 15 Jan 2008) | 2 lines

Try call if call is 0 or more (bug ASTERISK-2883)

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http://svn.digium.com/view/asterisk?view=rev&revision=4347
By: Digium Subversion (svnbot) 2008-01-15 15:15:08.000-0600

Repository: asterisk
Revision: 4348

U   branches/v1-0/channels/chan_sip.c

------------------------------------------------------------------------
r4348 | russell | 2008-01-15 15:15:08 -0600 (Tue, 15 Jan 2008) | 2 lines

Try call if call is 0 or more (bug ASTERISK-2883)

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http://svn.digium.com/view/asterisk?view=rev&revision=4348