[asterisk-users] Fwd: Dropping incompatible frame killing Asterisk

[Kevin Savoy]

This is an issue I'm having as well. Here is what I've discovered. 

Call comes in on a T1 line. Call is sent to a SIP phone (say 4000) based on
the extensions.conf setup. User of phone 4000 has set a forward in the phone
to an external number, 1-555-555-5555. There is nothing telling Asterisk to
Dial(Zap/g1) so the call does not get converted back to slin to send along
the T1 lines out of the building. Since SIP can't be sent the frame is
incompatible and is dropped. I know this probably isn't as technical as it
should be but in essence it is what is happening. I've had to do a
workaround and set up an extension that dials the number that the phone was
to be forwarded too. I set up extension 500. The user forwards the phone to
500. extensions.conf says Dial(Zap/g1/15555555555).

Band-aid solution. I've seen on the bug reports it is a known issue but not
resolved yet. Last update was July 5th.


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of M D
Sent: Thursday, August 10, 2006 8:50 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Fwd: Dropping incompatible frame killing Asterisk

Hi there

We're running Asterisk 1.2.1 (I know, it's old; we have an upgrade
planned but can't do it just yet) on Debian testing. Every now and
Asterisk and the box are dying -- no SSH login, no calls, nothing. The
last lines logged are:

Jul 31 14:23:31 VERBOSE[32696] logger.c:     -- Executing
Dial("SIP/5060-0843a7f0", "SIP/123456|30")
Jul 31 14:23:31 VERBOSE[32696] logger.c:     -- Called 123456
Jul 31 14:23:31 VERBOSE[14085] logger.c:     -- Got SIP response 302
"Moved Temporarily" back from 85.189.x.x
Jul 31 14:23:31 VERBOSE[32696] logger.c:     -- Now forwarding
SIP/5060-0843a7f0 to 'Local/02075551212 at Company_110' (thanks to
SIP/123456-2241)
Jul 31 14:23:31 VERBOSE[32701] logger.c:     -- Executing
Dial("Local/02075551212 at Company_110-7282,2",
"SIP/02075551212 at outbound.gateway:5070") in new stack
Jul 31 14:23:31 VERBOSE[32701] logger.c :     -- Called
02075551212 at outbound.gateway:5070
Jul 31 14:23:31 VERBOSE[32701] logger.c:     --
SIP/outbound.gateway:5070-550a is ringing
Jul 31 14:23:31 VERBOSE[32696] logger.c:     --
Local/02075551212 at Company_110-7282,1 is ringing
Jul 31 14:23:31 VERBOSE[32701] logger.c:     --
SIP/outbound.gateway:5070-550a is making progress passing it to
Local/02075551212 at Company_110-7282,2
Jul 31 14:23:31 VERBOSE[32696] logger.c:     --
Local/02075551212 at Company _110-7282,1 is making progress passing it to
SIP/5060-0843a7f0
Jul 31 14:23:31 NOTICE[32701] channel.c: Dropping incompatible voice
frame on Local/02075551212 at Company_110-7282,2 of format slin since our
native format has changed to alaw
Jul 31 14:23:31 NOTICE[32701] channel.c: Dropping incompatible voice
frame on Local/02075551212 at Company_110-7282,2 of format slin since our
native format has changed to alaw

The last lines are repeated until the server dies.

The phone appears to be a SNOM and should be using only g.711 alaw or ulaw.

I inherited this box with Asterisk running as root so I've changed it
to a non-privileged user but assuming the server is dynig through
resource starvation I doubt it'll help.

So, any ideas what this traffic is? What can we do to stop it? Clearly
I need to upgrade Asterisk but a cursory glance at the changelog
doesn't suggest a bug was reported with these symptoms which would
have been fixed in a later release.

Cheers,

Mark
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