[Asterisk-Users] My SIP example settings (and bugs)

[John Todd]

Firstly:  Thanks to all for the quick fixes on some of the SIP issues 
that have come up here in the last few weeks.  For the most part, SIP 
seems to work in and out of Asterisk, which is where I needed to be 
with this demonstration system I'm pulling together.

I've put together a minimal list of settings files required for 
getting Asterisk to run with some SIP channels.  Some of you have 
asked for example files to get FWD or iconnecthere working, and those 
are included, plus a lot of other very simple uses for Asterisk. 
This is very beginner-level stuff, but I try to fully comment what I 
do at each stage of a config file, so it may be useful to people 
starting out who are trying to avoid the traps that I fell into.  :)

Please find the examples on http://www.loligo.com/asterisk/   - I 
will submit revisions as I make them, and maybe someday I'll even put 
up some HTML pages with descriptions.

I've also found some problems, specifically SIP problems, that I've 
stumbled across while experimenting with different SIP 
servers/proxies and communications through them.  Some have been 
brought up on the list or in IRC, and some have not.  Anyone want to 
take a stab at them?

JT



-----

1) ATA-186 phones fail to stay registered.  Something within Asterisk 
is causing ATA-186 phones to stop sending REGISTER requests after ~2 
hours.   Experiments with 30 through 240 second timeouts on the 
ciscos have similar results.  Phone registry times out, calls fail.

2) Multiple ACK messages to certain SIP servers (FWD notably) - 
doesn't break anything, but why does it send ~8 ACKs to a successful 
registry?  Lots of fluff traffic.

3) DTMF relay through ATA-186 phones on SIP calls.  I'm uncertain if 
this is an ATA-186 issue or not; some in-depth prodding seems to show 
that it's an Asterisk problem, or lack of a feature.  DTMF reaches 
Asterisk, codes are shown on the console (in-band RFC2833) but are 
not played out the remote SIP channel; only slight garbled noise is 
heard.  Analog replay works fine (ATA -> Asterisk -> X100P)   Perhaps 
an origination problem with RFC2833 in-band signalling within 
Asterisk.  I've tried changing to in-band signalling on the ATA-186 
(AudioMode: 0x00050005) without success as well.

4) Calls via certain SIP servers fail if the calling party is an 
ATA-186 on both sides, seems to be an Asterisk issue.  I can 
reproduce.

5) Calls made back to oneself from a remote SIP server crashes Asterisk.

6) SIP register=  commands are only in the general context, 
preventing directive actions on inbound SIP calls.  This is a major 
issue, since a large number of PBX functions rely upon what number 
the caller was dialing.

7) Timers for register=  commands should be selectable on a per-service basis.

8) Asterisk crashes during remote REGISTER processes which have odd timing.

.
.
.
NOTICE[5126]: File chan_sip.c, Line 1763 (sip_reg_timeout): 
Registration timed out, trying again
NOTICE[5126]: File chan_sip.c, Line 1763 (sip_reg_timeout): 
Registration timed out, trying again
NOTICE[5126]: File chan_sip.c, Line 1763 (sip_reg_timeout): 
Registration timed out, trying again
NOTICE[5126]: File chan_sip.c, Line 2728 (handle_response): 
Registration successful
NOTICE[5126]: File chan_sip.c, Line 2729 (handle_response): 
Cancelling timeout 2232
NOTICE[5126]: File sched.c, Line 247 (ast_sched_del): Attempted to 
delete non-existant schedule entry 2217!
!! Forcing immediate crash a-la abort !!
Segmentation fault


9) REGISTER attempts to Vocal server (v1.4) fail with "404 Not Found" 
errors, despite correct entries (which work with ATA-186).  Perhaps 
notable is that even though a "404" error has been returned, and 
confirmation of failure has been printed on console, this is shown in 
a continuous loop on the console afterwards:

NOTICE[5126]: File chan_sip.c, Line 1763 (sip_reg_timeout): 
Registration timed out, trying again
WARNING[5126]: File chan_sip.c, Line 283 (__sip_xmit): sip_xmit of 
0x42c56480 (len 306) to 0.0.0.0 returned -1: Invalid argument