[SPAM:***** SpamScore] RE: [Asterisk-Users] Call Transfer using SIP clients

Frank Schoep frank at tintel.nl
Mon Jul 4 23:38:59 MST 2005


On Tuesday 05 July 2005 05:57, Tulika Pradhan wrote:
> call transfer works for me fine without any additions in features.conf
> by simply using Dial(SIP/${EXTEN},20,tT)
> and pressing #<number to be transfered to>
> this works both from caller as well as callee.
>
> tulika

Could you provide me with some more information so I can check where the 
differences in our setups are? It would really help to see how you 
implemented your extensions and SIP configuration. Could you describe the 
following regarding your Asterisk installation:

- Asterisk version
- The SIP clients you use
- Excerpt of "extensions.conf", which definitions and contexts do you include
- Excerpt of "features.conf", which lines (if any) are in there
- (Maybe) an excerpt of "sip.conf", how are the SIP peers configured

I hope you find the time to post these bits and pieces as it will make it 
easier for me to debug the situation. I've already tried numerous settings 
and combinations of options, but haven't had any luck yet. Thanks in advance 
for your precious time.

If anyone else has some ideas regarding my question, feel free to jump in - 
the more the merrier.

Sincerely,

Frank



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