Hello, I am new on the forum and I don’t know if my thread it is in the appropriate section of the forum. I have create a dialplan script in the extensions_custom.conf, where the script check some parameters in the database through some php’s and depending at the results from the php’s it allow the extension to make out-bound calls or not. My problem is when I have the script, the my sip phones cannot make transfer call or forward calls? Can you advice my how to debug my problem? Thanks for your attention and time. Sorry for my English.
extensions_custom.conf sounds like a FreePBX file name. We cannot support the FreePBX dialplan here.
Otherwise you will need to provide the script and a verbose level 5 trace of the call.
Note that transfers are also outbound calls. In the case of true SIP attended transfers, they are indistinguishable from an outbound call from the transferrer until the transfer is completed.
[quote=“david55”]
Note that transfers are also outbound calls. In the case of true SIP attended transfers, they are indistinguishable from an outbound call from the transferrer until the transfer is completed.[/quote]
First thanks for your response, this is very useful information. The call forward is the same case with the transfers ???
A forward will still result in an outbound call, but, one would expect it to be done with a 302 redirect, before answer. Normally Asterisk will use the user part of the new URI and effectively goto:
Dial(Local/user@???) on the redirected number. I think ??? is the context of the original call, but it might be that of the phone that did the redirect. Alternatively you can configure Asterisk to use the full URI and Dial that uri. I’m a bit hazy on all this, because not many people do 302 redirects from phones.