Re: [Asterisk-Users] Questionaire :

2004-08-09 Thread Aleph Communications
niko singh wrote:
Hi,
I have read quitea bit of the available resources and have this idea 
of asterisk. Would someone kindly answer these briefly
1-) Asterisk does not need a sound card...but if i am to record voice 
into an extension or dial from CLI ( basically use asterisk itself as 
a softphone ) then i need a sound card.  : Yes/No
Yes
a-) If yes creative soundblaster pci 128 is my best bet. Yes/No
Sorry, can't comment.
2-) Which is the best codec to use with different softphones
a-) SJphone :
b-)  Lipz4 :
c-)  Gnophone :
Sorry, I don't know.  I would recommend trying the different ones available.
3-)  Hardphones need to be plugged into the DSL router ( basically 
need an internet conn ) ... and need no digium cards.
a-) for recieving calls : Yes/No
They don't need digium cards. Depending on your network setup, you could 
probably get away without an internet connection for them as long as 
your asterisk box has connectivity.

b-) for making calls : Yes/no
See above answer
4-)  Sofphones run on machines different from the one running asterisk 
and can connect to asterisk , which routes the different phones 
according to contexts. Yes / no
yes
5-) Softphone can run on the same machine as asterisk and register the 
same way as if they were ona  different machine : yes /no
yes
6-) Softphones can behave as independent sip phones without asterisk.
Yes, at least if I understand you correctly.
Thanks
niko
-- SNIPPED--
Darren Wiebe
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[Asterisk-Users] Call File Routing

2004-08-09 Thread Aleph Communications
Here is a sample call file that I am using:
MaxRetries: 2
extension: 9997
Channel: IAX2/USERID:[EMAIL PROTECTED]/14037422000
CallerID: LAKEVIEW 4037422000
Anyway, this works fine.  The problem is that specifying the channel 
this way does not handle problems very well.  If hagenhomes is down, 
the call will not go through.  I would like to have the outgoing call 
routed as per one of my outgoing extensions, is this possible?  I have 
tried to get the server to register to itself but have been unable to 
succeed.  Any suggestions?

Darren Wiebe
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Re: [Asterisk-Users] Call File Routing

2004-08-09 Thread Aleph Communications
Thank you very much Jeremy.  This works perfectly.  I have been 
struggling for a long time to get this to work but it now works 
perfectly.  I have added a comment to the wiki: 
http://www.voip-info.org/wiki-Asterisk+auto-dial+out#comments

Thanks Again,
Darren Wiebe
[EMAIL PROTECTED]
Jeremy Hall wrote:
Aleph Communications  scribbled on Monday, August 09, 2004 10:50 AM:
 

Here is a sample call file that I am using:
MaxRetries: 2
extension: 9997
Channel: IAX2/USERID:[EMAIL PROTECTED]/14037422000
CallerID: LAKEVIEW 4037422000
Anyway, this works fine.  The problem is that specifying the channel
this way does not handle problems very well.  If hagenhomes is
down, the call will not go through.  I would like to have the
outgoing call routed as per one of my outgoing extensions, is this
possible?  I have tried to get the server to register to itself but
have been unable to succeed.  Any suggestions? 
   

I have not yet tried it, but earlier today I had a similar question and
Twisted in IRC gave me the following response:
twisted Channel: Local/[EMAIL PROTECTED]
twisted Application: Dial   
twisted and then the arguments go in the appropriate place
Hope that helps.
Jeremy
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