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

[Andre ANJOS <andre adois dyndns org>]

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Hi,

I'm sorry, I got confused with multiple executables in the system,
getting wrongly the impression ekiga-snapshot was crashing. With the
snapshot from yesterday I CANNOT reproduce the crash, only with 2.0.1.
So, I guess this must have been fixed! Thanks a lot for your time in
solving this issue. Looking forward to download 2.0.2!

André.

Damien Sandras wrote:
> Hi,
> 
> I can not reproduce the problem.
> 
> Can you mail me the output of ekiga -d 4 > output.txt 2>&1 ?
> (with Ekiga CVS going crazy and sending thousands of SUBSCRIBE's until
> it crashes)
> 
> Le mardi 09 mai 2006 à 10:44 +0200, Andre ANJOS a écrit :
> 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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