M D
2006-Aug-10 06:50 UTC
[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@Company_110' (thanks to
SIP/123456-2241)
Jul 31 14:23:31 VERBOSE[32701] logger.c: -- Executing
Dial("Local/02075551212@Company_110-7282,2",
"SIP/02075551212@outbound.gateway:5070") in new stack
Jul 31 14:23:31 VERBOSE[32701] logger.c : -- Called
02075551212@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@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@Company_110-7282,2
Jul 31 14:23:31 VERBOSE[32696] logger.c: --
Local/02075551212@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@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@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
Kevin Savoy
2006-Aug-10 07:30 UTC
[asterisk-users] Fwd: Dropping incompatible frame killing Asterisk
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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of M D
Sent: Thursday, August 10, 2006 8:50 AM
To: asterisk-users@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@Company_110' (thanks to
SIP/123456-2241)
Jul 31 14:23:31 VERBOSE[32701] logger.c: -- Executing
Dial("Local/02075551212@Company_110-7282,2",
"SIP/02075551212@outbound.gateway:5070") in new stack
Jul 31 14:23:31 VERBOSE[32701] logger.c : -- Called
02075551212@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@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@Company_110-7282,2
Jul 31 14:23:31 VERBOSE[32696] logger.c: --
Local/02075551212@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@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@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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WideVOIP
2006-Aug-10 07:43 UTC
[asterisk-users] Fwd: Dropping incompatible frame killing Asterisk= same in 1.2.10
Hello We have the same issue asterisk[5303]: NOTICE[5303]: channel.c:1917 in ast_read: Dropping incompatible voice frame on Local/662@context_poste-be6d,2 of format alaw since our native format has changed to slin We are using asterisk 1.2.10 + zaptel 1.2.7 + libpri 1.2.3 on a redhat 9 P4 2 Go RAM (running as root) All is configured alaw + ulaw + slin in sip.conf [general + accounts] - iax.conf We have only alaw ulaw sln files for MOH and messages We are also suffering for no audio on Sip => Sip local calls It's ok after "reload" As anyone any suggestion Regards Thierry> -----Message d'origine----- > De : asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] De la part de M D > Envoy? : jeudi 10 ao?t 2006 15:50 > ? : asterisk-users@lists.digium.com > Objet : [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@Company_110' (thanks to > SIP/123456-2241) > Jul 31 14:23:31 VERBOSE[32701] logger.c: -- Executing > Dial("Local/02075551212@Company_110-7282,2", > "SIP/02075551212@outbound.gateway:5070") in new stack > Jul 31 14:23:31 VERBOSE[32701] logger.c : -- Called > 02075551212@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@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@Company_110-7282,2 > Jul 31 14:23:31 VERBOSE[32696] logger.c: -- > Local/02075551212@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@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@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 > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >