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

Thu Aug 10 08:51:10 MST 2006

What about using a *72 application to forward the calls rather than
the phone itself.

on Thursday 08/10/2006 M D(md1979md at googlemail.com) wrote
 > 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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