[Asterisk-Users] Xfering a call

[Andrei (MPI)]

Rich Adamson wrote:

>>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.
>  
>
I disagree about hangup. I would recommend to leave Hangup there. As you 
never know what could happen to an app (Voicemail in this case) -
 it may be spelled wrong or just not configured right.

Andrei (MPI)