[Asterisk-Users] Redirect problem/bug/feature

C F shmaltz at gmail.com
Tue Mar 28 13:35:48 MST 2006


This is just like you if you would be doing this:
Dial(Local/944 at default,22,t)
Which if Local/944 has then:
Dial(Local/904 at defualt)
and exten => 904,1 has:
Dial (Sip/904,,90,t)
Then asterisk will dial sip/904 for 22 seconds.

and whatever follows the first dial (Dial(Local/944 at default) will be
executed, in your case Voicemail for 944.

To confirm this try chaning the time for 944 to 90 secnonds, and the
time for 904 to 22. You will then see that asterisk will give you the
voicemail of 904.

The proper way to set this up is to test for the ${DNID} variable,
which should tell you in the above 944, that way you will know exactly
what VM to give the user.




On 3/28/06, Joe <jrothstein at comcentrixs.com> wrote:
> I have a major problem with SIP redirects, or maybe just do not understand
> how they are supposed to work. We are using Cisco 7940s and 7960s with SIP
> version 6.3. Asterisk 1.2.5.
>
> A call come in to extension 944 over the IAX trunk. Extension 944 has
> forward all to extension 904 set on the phone. According to the dialplan.
> extension 904 should ring for 90 seconds, then ring another extension, and
> then finally go to voicemail. But for some reason, Asterisk still honors the
> ring time of extension 944 which is 22 seconds, and then goes to the
> voicemail of extension 944.
>
> Here is the call flow:
>
> AST-Sanset*CLI>
>     -- Accepting AUTHENTICATED call from 195.27.242.8:
>        > requested format = g729,
>        > requested prefs = (),
>        > actual format = g729,
>        > host prefs = (g729),
>        > priority = mine
>     -- Executing Goto("IAX2/comcentrixs-uplink-6", "sanset|944|1") in new
> stack
>     -- Goto (sanset,944,1)
>     -- Executing Macro("IAX2/comcentrixs-uplink-6", "stdexten") in new stack
>     -- Executing Set("IAX2/comcentrixs-uplink-6", "LANGUAGE()=de") in new
> stack
>     -- Executing SetMusicOnHold("IAX2/comcentrixs-uplink-6", "default") in
> new stack
>     -- Executing Dial("IAX2/comcentrixs-uplink-6", "SIP/944|22|t") in new
> stack
>     -- Called 944
>     -- Got SIP response 302 "Moved Temporarily" back from 192.168.0.32
>     -- Now forwarding IAX2/comcentrixs-uplink-6 to
> 'SIP/904 at 192.168.0.2:5060' (thanks to SIP/944-657d)
>     -- Now forwarding IAX2/comcentrixs-uplink-6 to 'Local/904 at sanset'
> (thanks to SIP/192.168.0.2:5060-7b9b)
>     -- Executing Set("Local/904 at sanset-1df8,2", "LANGUAGE()=de") in new
> stack
>     -- Executing SetMusicOnHold("Local/904 at sanset-1df8,2", "default") in new
> stack
>     -- Executing Dial("Local/904 at sanset-1df8,2", "SIP/904|90|t") in new
> stack
>     -- Called 904
>     -- SIP/904-3622 is ringing
>     -- Local/904 at sanset-1df8,1 is ringing
>     -- Nobody picked up in 22000 ms
>     -- Executing VoiceMail("IAX2/comcentrixs-uplink-6", "u944 at sanset") in
> new stack
>     -- Playing 'vm-theperson' (language 'de')
>   == Spawn extension (sanset, 904, 3) exited non-zero on
> 'Local/904 at sanset-1df8,2'
> Mar 28 21:55:06 WARNING[4187]: channel.c:787 channel_find_locked: Avoided
> initial deadlock for '0x819ea38', 10 retries!
>     -- Playing 'digits/9' (language 'de')
>     -- Playing 'digits/4' (language 'de')
>     -- Playing 'digits/4' (language 'de')
>     -- Playing 'vm-isunavail' (language 'de')
>   == Spawn extension (macro-stdexten, s, 4) exited non-zero on
> 'IAX2/comcentrixs-uplink-6' in macro 'stdexten'
>   == Spawn extension (macro-stdexten, s, 4) exited non-zero on
> 'IAX2/comcentrixs-uplink-6'
>     -- Hungup 'IAX2/comcentrixs-uplink-6'
>
> Is this the way this is supposed to work? Seems like if the call if
> forwarded to 904, then the timers, voicemail, etc. of 904 should be used
> instead of that of 944.
>
> Also, can anyone explain this:
>
> Mar 28 21:55:06 WARNING[4187]: channel.c:787 channel_find_locked: Avoided
> initial deadlock for '0x819ea38', 10 retries!
>
> Regards to all,
> Joe
>
>
>
>
>
> _______________________________________________
> --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
>



More information about the asterisk-users mailing list