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

[M D]

Hi

Sorry, I should have mentioned that we're only running SIP. Our calls
to the PSTN are routed through a VoIP carrier and all of our clients
are SIP.

Which version of Asterisk are you using? Is this killing your box? If
it is, have you established why? CPU being killed, memory starvation,
something else?

It is only happening on forwarded calls, though. I'll have to try your
workaround.

Thanks,

Mark

On 10/08/06, Kevin Savoy <ksavoy at novo1.com> wrote:
> 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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