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

[Mark Spencer]

> 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.

Need more info here.  I can't think of anything Asterisk could tell the
Cisco to stop it from registering.

> 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.

A sip debug dump here might be helpful.  Is FWD sending is multiple
responses (requiring ACKS) or are we sending unsolicited ACKS?  Are the
ACKs associated with a particular kind of message?

> 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.

Is it possible that the remote SIP channel does not support RFC2833?

> 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.

The ATA186 does not seem to properly support reinvites and causes a
one-way (or in the case of two of them, no-way) audio problem if reinvites
are permitted.

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

Is the remote server trying to do a refer?  Again, a sip debug would be
helpful here.  Refer support was broken until yesterday.

> 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.

General is a section, not a context.  In any case, update CVS and look at
the new sip.conf.sample, which includes an example of how to setup a
different contact than just "s".

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

Boo hoo :)  I'll take a patch if someone wants to submit it, but I don't
see much of a reason not to just use a "lowest common demoninator"
approach.  As of Saturday the times are (on a global scale) configurable.

> 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
> 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

There are a number of issues related to REGISTER that probably need to be
worked out.  Since I didn't write the SIP REGISTER code, it's taking me a
little time to understand how it works.  If you find me on IRC, though,
I'd be happy to work with you on fixing your outgoing REGISTER issues.

Mark