Re: [GnomeMeeting-list] Ekiga keeps crashing after repetitive subscribe messages



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Hi, here it is. My system is behind a firewall. The localnetwork is
192.168.1.*. Both ekiga and asterisk run on 192.168.1.20. Asterisk takes
care of the default port for initiation (i.e. UDP/5060). Ekiga runs on
port 10060, as a client to asterisk. Now, this configuration will
produce the problem here. If I set canreinvite=no at the '[home]'
section. Ekiga seems to not crash anymore. I use asterisk 1.2.7.1, the
lastest stable ekiga plus all required libraries.

sip.conf:
=========

; SIP configuration channels for asterisk
[general]
context=default         ; Default context for incoming calls
bindport=5060           ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0        ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes           ; Enables DNS SRV lookups on outbound calls
externhost=adois.dyndns.org ; Makes asterisk find out our real IP address
externrefresh=300       ; Refresh the address every 5 minutes
localnet=192.168.1.0/24 ; Our local network is everybody from 192.168.1.*
pedantic=yes            ; Be strict on the package content to clients
realm=sip.pbx.adois.org ; Our default realm

[home]
type=friend
; generated with `echo -n '<user>:<realm>:<pass>' | md5sum`
md5secret=<to-be-filled>
;secret=<to-be-filled>
qualify=no
nat=no
host=dynamic
canreinvite=yes
context=internal
port=10060
dtmfmode=rfc2833
; codecs supported (by ekiga)
disallow=all     ;Note order of disallow/allow is important.
allow=gsm

extensions.conf (taken from asterisk's example, w/o changes)
============================================================

; Dialplan configuration for asterisk
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]
CONSOLE=Console/dsp     ; Console interface for demo

[internal]
exten => s,1,Wait,1                     ; Wait a second, just for fun
exten => s,n,Answer                     ; Answer the line
exten => s,n,Set(TIMEOUT(digit)=5)      ; Set Digit Timeout to 5 seconds
exten => s,n,Set(TIMEOUT(response)=10)  ; Set Response Timeout to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory
message
exten => s,n(instruct),BackGround(demo-instruct)        ; Play some
instructions
exten => s,n,WaitExten          ; Wait for an extension to be dialed.

exten => 2,1,BackGround(demo-moreinfo)  ; Give some more information.
exten => 2,n,Goto(s,instruct)

exten => 3,1,Set(LANGUAGE()=fr)         ; Set language to french
exten => 3,n,Goto(s,restart)                    ; Start with the
congratulations

exten => 1000,1,Goto(default,s,1)
exten => 1234,1,Playback(transfer,skip)         ; "Please hold while..."
                                        ; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${CONSOLE})

exten => 1235,1,Voicemail(u1234)                ; Right to voicemail

exten => 1236,1,Dial(Console/dsp)               ; Ring forever
exten => 1236,n,Voicemail(u1234)                ; Unless busy

exten => #,1,Playback(demo-thanks)              ; "Thanks for trying the
demo"
exten => #,n,Hangup                     ; Hang them up.

exten => t,1,Goto(#,1)                  ; If they take too long, give up
exten => i,1,Playback(invalid)          ; "That's not valid, try again"

exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/guest misery digium com/s default)     ; Call
the Aster
isk demo
exten => 500,n,Playback(demo-nogo)      ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6)                ; Return to the start over message.


Damien Sandras wrote:
> Le mardi 09 mai 2006 à 09:30 +0200, André Anjos a écrit :
>> Okay, I just found the reason: If I set 'canreinvite=no' in asterisk, it
>> stops doing this iterative thing. Why doesn't Ekiga support this option?
>>
> 
> It does support canreinvite=yes. Canreinvite=yes only has an effect
> during calls, but obviously, SUBSCRIBE is not related to reinvite.
> 
> I do not understand the problem, I would like to reproduce it.
> 
> Can you email me your configuration of Asterisk? (sip.conf and
> extensions.conf) allowing to reproduce the problem?
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