[Asterisk-Users] Xfering a call

[Noah Miller]

> Well that didn't work....I now get this error
>
>
> Jan 12 16:56:21 NOTICE[4989]: app_dial.c:746 dial_exec: Unable to 
> create
> channel of type 'SIP'
>   == Everyone is busy/congested at this time
>     -- Executing VoiceMail("IAX2/iaxfwd at 65.39.205.121:4569/5", "b") in 
> new
> stackJan 12 16:56:21 WARNING[4989]: app_voicemail.c:1539 
> leave_voicemail: No
> entry in voicemail config file for ''
>     -- Timeout on IAX2/iaxfwd at 65.39.205.121:4569/5
>   == CDR updated on IAX2/iaxfwd at 65.39.205.121:4569/5
>     -- Executing Goto("IAX2/iaxfwd at 65.39.205.121:4569/5", "#|1") in new
> stack
>     -- Goto (home,#,1)
>     -- Executing Playback("IAX2/iaxfwd at 65.39.205.121:4569/5", 
> "sai-thanks")
> in new stack
> Jan 12 16:56:31 WARNING[4989]: file.c:475 ast_openstream: File 
> sai-thanks
> does not exist in any format
> Jan 12 16:56:31 WARNING[4989]: file.c:779 ast_streamfile: Unable to 
> open
> sai-thanks (format ulaw): No such file or directory
> Jan 12 16:56:31 WARNING[4989]: app_playback.c:83 playback_exec:
> ast_streamfile failed on IAX2/iaxfwd at 65.39.205.121:4569/5 for 
> sai-thanks
>     -- Executing Hangup("IAX2/iaxfwd at 65.39.205.121:4569/5", "") in new 
> stack
>   == Spawn extension (home, #, 2) exited non-zero on
> 'IAX2/iaxfwd at 65.39.205.121:4569/5'
>     -- Hungup 'IAX2/iaxfwd at 65.39.205.121:4569/5'
>
> This user does have an entry in the voicemail.conf file......
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rich 
> Adamson
> Sent: Wednesday, January 12, 2005 3:46 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Xfering a call
>
>> I'm having an issue when I transfer a call to another SIP extension it
> sees
>> that the sip phone is not there and goes to voicemail but in my case 
>> it
>> transfers to the main voicemail instead of the users voicemail.
>>
>> Here is what my SIP extensions look like in the extension.conf file
>>
>> exten => 3957,1,Dial(${Theresa},20,Tt)
>> exten => 3957,2,VoicemailMain2(u${TheresaVM})
>> exten => 3957,3,Hangup
>> exten => 3957,102,VoicemailMain2(b${TheresaVM})
>> exten => 3957,103,Hangup
>
> Change the above from VoicemailMain2 to Voicemail and it will work
> as expected.
>
> The 3,Hangup isn't required... remove it. 103 isn't actually needed
> either.

Well this error message is tricky:

>     -- Executing VoiceMail("IAX2/iaxfwd at 65.39.205.121:4569/5", "b") in 
> new
> stackJan 12 16:56:21 WARNING[4989]: app_voicemail.c:1539 
> leave_voicemail: No
> entry in voicemail config file for ''

Is "IAX2/iaxfwd at 65.39.205.121:4569/5" what the TheresaVM variable 
dereferences to?  If so, I don't think this is possible.  I think 
voicemail has to be stored locally.  I don't think you can use the 
Voicemail() application with an IAX forward (Is it?).  With the SIP/IAX 
and voicemail box setup this should work.  It's hard to tell exactly, 
since your extension numbers and voicemail box numbers are stored in 
variables, but if you configure Theresa for SIP extension 3957 and 
voicemail box 3957, her section in extensions.conf would look like 
this:

exten => 3957,1,Dial(SIP/3957,20,Tt)
exten => 3957,2,Voicemail(u3957)
exten => 3957,102,Voicemail(b3957)
exten => 3957,103,Hangup

If this does not work, check to make sure of the voicemail box number 
and the sip number.  If they are both correct, try recompiling asterisk 
with the latest CVS version.