strange asterisk crash on calls forwarded to Local/ext

Greetings folks,

Could I ask a minute of your attention on the bug I've
posted at 
http://bugs.d
igium.com/view.php?id=7557 ?

I'm clueless what is the cause of it.

In short, if you set on any Sipura device (tested with
2100/3000/941) a 
"Cfwd all" (*72 service by default) to a queue
or moh extension number, 
asterisk core dumps every time you call that device
extension. If you 
put a direct extension (e.g. another sip device extension),
call gets 
through.

Anyone able to reproduce this?

Thanks for your time,
Vahan
_______________________________________________
Asterisk-BSD mailing list
Asterisk-BSDlists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-bsd

I'll hazard a guess.

When a Sipura device does a "forward" it sends a
SIP "temporarily moved" 
message back to the originator, with the URI that you are
forwarding to.  If 
this is another SIP device, this is pretty easy, as Asterisk
just re-invites.  
The problem is when it tries to re-invite to itself, it gets
a little 
confused.  At least that's what I took from the
documentation.  Based on your 
debug output, it looks like it is trying to re-invite to a
context called 
main.  Does this context exist, and are the appropriate
extensions in it?

-Tim

On July 20, 2006 11:25, Vahan Yerkanian wrote:
> Greetings folks,
>
> Could I ask a minute of your attention on the bug I've
posted at
> http://bugs.d
igium.com/view.php?id=7557 ?
>
> I'm clueless what is the cause of it.
>
> In short, if you set on any Sipura device (tested with
2100/3000/941) a
> "Cfwd all" (*72 service by default) to a
queue or moh extension number,
> asterisk core dumps every time you call that device
extension. If you
> put a direct extension (e.g. another sip device
extension), call gets
> through.
>
> Anyone able to reproduce this?
>
> Thanks for your time,
> Vahan
> _______________________________________________
> Asterisk-BSD mailing list
> Asterisk-BSDlists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-bsd

-- 
Tim St. Pierre

IP telephony specialist
sip://5101communicatefreely.net
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
timcommunicatefreely.net
_______________________________________________
Asterisk-BSD mailing list
Asterisk-BSDlists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-bsd

I'll hazard a guess.

When a Sipura device does a "forward" it sends a
SIP "temporarily moved" 
message back to the originator, with the URI that you are
forwarding to.  If 
this is another SIP device, this is pretty easy, as Asterisk
just re-invites.  
The problem is when it tries to re-invite to itself, it gets
a little 
confused.  At least that's what I took from the
documentation.  Based on your 
debug output, it looks like it is trying to re-invite to a
context called 
main.  Does this context exist, and are the appropriate
extensions in it?

-Tim

On July 20, 2006 11:25, Vahan Yerkanian wrote:
> Greetings folks,
>
> Could I ask a minute of your attention on the bug I've
posted at
> http://bugs.d
igium.com/view.php?id=7557 ?
>
> I'm clueless what is the cause of it.
>
> In short, if you set on any Sipura device (tested with
2100/3000/941) a
> "Cfwd all" (*72 service by default) to a
queue or moh extension number,
> asterisk core dumps every time you call that device
extension. If you
> put a direct extension (e.g. another sip device
extension), call gets
> through.
>
> Anyone able to reproduce this?
>
> Thanks for your time,
> Vahan
> _______________________________________________
> Asterisk-BSD mailing list
> Asterisk-BSDlists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-bsd

-- 
Tim St. Pierre

IP telephony specialist
sip://5101communicatefreely.net
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
timcommunicatefreely.net
_______________________________________________
Asterisk-BSD mailing list
Asterisk-BSDlists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-bsd

I'll hazard a guess.

When a Sipura device does a "forward" it sends a
SIP "temporarily moved" 
message back to the originator, with the URI that you are
forwarding to.  If 
this is another SIP device, this is pretty easy, as Asterisk
just re-invites.  
The problem is when it tries to re-invite to itself, it gets
a little 
confused.  At least that's what I took from the
documentation.  Based on your 
debug output, it looks like it is trying to re-invite to a
context called 
main.  Does this context exist, and are the appropriate
extensions in it?

-Tim

On July 20, 2006 11:25, Vahan Yerkanian wrote:
> Greetings folks,
>
> Could I ask a minute of your attention on the bug I've
posted at
> http://bugs.d
igium.com/view.php?id=7557 ?
>
> I'm clueless what is the cause of it.
>
> In short, if you set on any Sipura device (tested with
2100/3000/941) a
> "Cfwd all" (*72 service by default) to a
queue or moh extension number,
> asterisk core dumps every time you call that device
extension. If you
> put a direct extension (e.g. another sip device
extension), call gets
> through.
>
> Anyone able to reproduce this?
>
> Thanks for your time,
> Vahan
> _______________________________________________
> Asterisk-BSD mailing list
> Asterisk-BSDlists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-bsd

-- 
Tim St. Pierre

IP telephony specialist
sip://5101communicatefreely.net
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
timcommunicatefreely.net
_______________________________________________
Asterisk-BSD mailing list
Asterisk-BSDlists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-bsd

I'll hazard a guess.

When a Sipura device does a "forward" it sends a
SIP "temporarily moved" 
message back to the originator, with the URI that you are
forwarding to.  If 
this is another SIP device, this is pretty easy, as Asterisk
just re-invites.  
The problem is when it tries to re-invite to itself, it gets
a little 
confused.  At least that's what I took from the
documentation.  Based on your 
debug output, it looks like it is trying to re-invite to a
context called 
main.  Does this context exist, and are the appropriate
extensions in it?

-Tim

On July 20, 2006 11:25, Vahan Yerkanian wrote:
> Greetings folks,
>
> Could I ask a minute of your attention on the bug I've
posted at
> http://bugs.d
igium.com/view.php?id=7557 ?
>
> I'm clueless what is the cause of it.
>
> In short, if you set on any Sipura device (tested with
2100/3000/941) a
> "Cfwd all" (*72 service by default) to a
queue or moh extension number,
> asterisk core dumps every time you call that device
extension. If you
> put a direct extension (e.g. another sip device
extension), call gets
> through.
>
> Anyone able to reproduce this?
>
> Thanks for your time,
> Vahan
> _______________________________________________
> Asterisk-BSD mailing list
> Asterisk-BSDlists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-bsd

-- 
Tim St. Pierre

IP telephony specialist
sip://5101communicatefreely.net
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
timcommunicatefreely.net
_______________________________________________
Asterisk-BSD mailing list
Asterisk-BSDlists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-bsd

I'll hazard a guess.

When a Sipura device does a "forward" it sends a
SIP "temporarily moved" 
message back to the originator, with the URI that you are
forwarding to.  If 
this is another SIP device, this is pretty easy, as Asterisk
just re-invites.  
The problem is when it tries to re-invite to itself, it gets
a little 
confused.  At least that's what I took from the
documentation.  Based on your 
debug output, it looks like it is trying to re-invite to a
context called 
main.  Does this context exist, and are the appropriate
extensions in it?

-Tim

On July 20, 2006 11:25, Vahan Yerkanian wrote:
> Greetings folks,
>
> Could I ask a minute of your attention on the bug I've
posted at
> http://bugs.d
igium.com/view.php?id=7557 ?
>
> I'm clueless what is the cause of it.
>
> In short, if you set on any Sipura device (tested with
2100/3000/941) a
> "Cfwd all" (*72 service by default) to a
queue or moh extension number,
> asterisk core dumps every time you call that device
extension. If you
> put a direct extension (e.g. another sip device
extension), call gets
> through.
>
> Anyone able to reproduce this?
>
> Thanks for your time,
> Vahan
> _______________________________________________
> Asterisk-BSD mailing list
> Asterisk-BSDlists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-bsd

-- 
Tim St. Pierre

IP telephony specialist
sip://5101communicatefreely.net
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
timcommunicatefreely.net
_______________________________________________
Asterisk-BSD mailing list
Asterisk-BSDlists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-bsd

Tim St. Pierre wrote:
> I'll hazard a guess.
> 
> When a Sipura device does a "forward" it
sends a SIP "temporarily moved" 
> message back to the originator, with the URI that you
are forwarding to.  If 
> this is another SIP device, this is pretty easy, as
Asterisk just re-invites.  
> The problem is when it tries to re-invite to itself, it
gets a little 
> confused.  At least that's what I took from the
documentation.  Based on your 
> debug output, it looks like it is trying to re-invite
to a context called 
> main.  Does this context exist, and are the appropriate
extensions in it?
> 
> -Tim

Aye, the content does exist, and actually what happens is
asterisk 
forwards to a Local/111Main , that's the moment asterisk
crashes:

     -- Executing Macro("SIP/10040-4c43",
"call|10027") in new stack
     -- Executing Set("SIP/10040-4c43",
"ext=10027") in new stack
     -- Executing Dial("SIP/10040-4c43",
"SIP/10027|20|o") in new stack
     -- Called 10027
     -- Got SIP response 302 "Moved Temporarily"
back from 10.20.30.40
     -- Now forwarding SIP/10040-4c43 to 'Local/111Main'
(thanks to 
SIP/10027-4f37)
sip*CLI>
Disconnected from Asterisk server
#

However, If I set forwarding to go to another phone
registered to 
asterisk, say 10011, everything works:


     -- Executing Macro("SIP/10040-be47",
"call|10027") in new stack
     -- Executing Set("SIP/10040-be47",
"ext=10027") in new stack
     -- Executing Dial("SIP/10040-be47",
"SIP/10027|20|o") in new stack
     -- Called 10027
     -- Got SIP response 302 "Moved Temporarily"
back from 195.250.90.228
     -- Now forwarding SIP/10040-be47 to 'Local/10011Main'
(thanks to 
SIP/10027-91c8)
     -- Executing Macro("Local/10011Main-9305,2", "call|10011") in new 
stack
     -- Executing Set("Local/10011Main-9305,2", "ext=10011") in new
stack
     -- Executing Dial("Local/10011Main-9305,2", "SIP/10011|20|o") in 
new stack
     -- Called 10011
     -- SIP/10011-5cff is ringing
     -- Local/10011Main-9305,1 is ringing






_______________________________________________
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Asterisk-BSDlists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-bsd

Tim St. Pierre wrote:
> I'll hazard a guess.
> 
> When a Sipura device does a "forward" it
sends a SIP "temporarily moved" 
> message back to the originator, with the URI that you
are forwarding to.  If 
> this is another SIP device, this is pretty easy, as
Asterisk just re-invites.  
> The problem is when it tries to re-invite to itself, it
gets a little 
> confused.  At least that's what I took from the
documentation.  Based on your 
> debug output, it looks like it is trying to re-invite
to a context called 
> main.  Does this context exist, and are the appropriate
extensions in it?
> 
> -Tim

Aye, the content does exist, and actually what happens is
asterisk 
forwards to a Local/111Main , that's the moment asterisk
crashes:

     -- Executing Macro("SIP/10040-4c43",
"call|10027") in new stack
     -- Executing Set("SIP/10040-4c43",
"ext=10027") in new stack
     -- Executing Dial("SIP/10040-4c43",
"SIP/10027|20|o") in new stack
     -- Called 10027
     -- Got SIP response 302 "Moved Temporarily"
back from 10.20.30.40
     -- Now forwarding SIP/10040-4c43 to 'Local/111Main'
(thanks to 
SIP/10027-4f37)
sip*CLI>
Disconnected from Asterisk server
#

However, If I set forwarding to go to another phone
registered to 
asterisk, say 10011, everything works:


     -- Executing Macro("SIP/10040-be47",
"call|10027") in new stack
     -- Executing Set("SIP/10040-be47",
"ext=10027") in new stack
     -- Executing Dial("SIP/10040-be47",
"SIP/10027|20|o") in new stack
     -- Called 10027
     -- Got SIP response 302 "Moved Temporarily"
back from 195.250.90.228
     -- Now forwarding SIP/10040-be47 to 'Local/10011Main'
(thanks to 
SIP/10027-91c8)
     -- Executing Macro("Local/10011Main-9305,2", "call|10011") in new 
stack
     -- Executing Set("Local/10011Main-9305,2", "ext=10011") in new
stack
     -- Executing Dial("Local/10011Main-9305,2", "SIP/10011|20|o") in 
new stack
     -- Called 10011
     -- SIP/10011-5cff is ringing
     -- Local/10011Main-9305,1 is ringing






_______________________________________________
Asterisk-BSD mailing list
Asterisk-BSDlists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-bsd

Tim St. Pierre wrote:
> I'll hazard a guess.
> 
> When a Sipura device does a "forward" it
sends a SIP "temporarily moved" 
> message back to the originator, with the URI that you
are forwarding to.  If 
> this is another SIP device, this is pretty easy, as
Asterisk just re-invites.  
> The problem is when it tries to re-invite to itself, it
gets a little 
> confused.  At least that's what I took from the
documentation.  Based on your 
> debug output, it looks like it is trying to re-invite
to a context called 
> main.  Does this context exist, and are the appropriate
extensions in it?
> 
> -Tim

Aye, the content does exist, and actually what happens is
asterisk 
forwards to a Local/111Main , that's the moment asterisk
crashes:

     -- Executing Macro("SIP/10040-4c43",
"call|10027") in new stack
     -- Executing Set("SIP/10040-4c43",
"ext=10027") in new stack
     -- Executing Dial("SIP/10040-4c43",
"SIP/10027|20|o") in new stack
     -- Called 10027
     -- Got SIP response 302 "Moved Temporarily"
back from 10.20.30.40
     -- Now forwarding SIP/10040-4c43 to 'Local/111Main'
(thanks to 
SIP/10027-4f37)
sip*CLI>
Disconnected from Asterisk server
#

However, If I set forwarding to go to another phone
registered to 
asterisk, say 10011, everything works:


     -- Executing Macro("SIP/10040-be47",
"call|10027") in new stack
     -- Executing Set("SIP/10040-be47",
"ext=10027") in new stack
     -- Executing Dial("SIP/10040-be47",
"SIP/10027|20|o") in new stack
     -- Called 10027
     -- Got SIP response 302 "Moved Temporarily"
back from 195.250.90.228
     -- Now forwarding SIP/10040-be47 to 'Local/10011Main'
(thanks to 
SIP/10027-91c8)
     -- Executing Macro("Local/10011Main-9305,2", "call|10011") in new 
stack
     -- Executing Set("Local/10011Main-9305,2", "ext=10011") in new
stack
     -- Executing Dial("Local/10011Main-9305,2", "SIP/10011|20|o") in 
new stack
     -- Called 10011
     -- SIP/10011-5cff is ringing
     -- Local/10011Main-9305,1 is ringing






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On 07/21/06 00:28 Vahan Yerkanian said the following:
>     -- Got SIP response 302 "Moved
Temporarily" back from 10.20.30.40
>     -- Now forwarding SIP/10040-4c43 to 'Local/111Main'
(thanks to 
> SIP/10027-4f37)

that occurs around line 469 in apps/app_dial.c. you may want
to try adding 
ast_debug calls at each stage to see exactly where it
crashes, or perhaps 
try debugging the core file with gdb to see why it crashes.
it's most 
likely a pointer being doubly freed.

-- 
Regards,                           /\_/\   "All dogs
go to heaven."
dineshalphaque.com                (0 0)   http://www.openmalay
siablog.com/
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