[asterisk-users] Garbled voicemail

2012-02-09 Thread Dan Ritter 
Our Asterisk system (1.8.8.1-1digium1~squeeze) has been very
stable and generally doing a good job -- except that one day,   
voicemail recordings started being garbled. 

It only manifests when the VM comes from our telco gateway  
service -- OnSIP/Junction -- and not from internal phones or
from an Asterisk box I have at home.

We have voicemail set to record to WAV, and real files are  
being generated -- but it sounds incredibly sped up, faster than
chipmunks. Completely unintelligible, even if you pull it into  
an audio editor and slow down playback. 

It is not perfectly consistent, but it happens in about 85% of  
voicemail recordings left from the outside world through OnSIP. 

We've had several years of trouble-free voicemail before this.  

Anyone seen anything similar? Advice? Wild speculation? 

-dsr-   


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[asterisk-users] MeetMe and dedicated conference room phone

2009-09-01 Thread Dan Ritter 

I've googled and not quite found what I need, so...

I have a conference room phone that I would like to make behave as
follows:

- when a call comes to that extension:
answer the call
put the call in a static MeetMe room with option 'w'
ring the phone by SIP
and when the phone picks up, put it in the same MeetMe
room as the marked call.

if subsequent calls come in, they are put in the same
room.

Is this difficult?

Part two, probably harder:

- when a call goes out from this SIP phone:
save whatever number was dialled
put the call in a static MeetMe room (same as above is
fine)
call the saved number
and connect that call to the room.

Any ideas?

-dsr-


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