Hi
I want to impelement a simple scenario in Elastix with Extentions.
Suppose we have 2 extention: 100 and 101
we want to have this planing in our company: when some body called 100 extention, after 15 sec, if not answer, the caller is diverted to mobile phone of extention.
How can i do it?
Thanks.
with regards.Mojtaba
[quote=“mespio”]Hi
I want to impelement a simple scenario in Elastix with Extentions.
Suppose we have 2 extention: 100 and 101
we want to have this planing in our company: when some body called 100 extention, after 15 sec, if not answer, the caller is diverted to mobile phone of extention.
How can i do it?
Thanks.
with regards.Mojtaba[/quote]
see document [url]http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial[/url]
exten => 100,1,Dial(SIP/100,15,tTr)
exten => 100,1,Dial(SIP/98765432198,15)woren,
Almost the right question, this time, although you missed the reference to Elsastix!
The answer is wrong, for raw Asterisk, as the second line should be priority 2. Also, tTr is not necessarily right for this user, and you would expect the same options on line 2. In particular, specifying “T” makes the system vulnerable to toll fraud.
Regarding tbe “T”, the contexts might not actually allow an external caller to make an external call, but it still should not have been used blindly.
I also missed the fact that it would be a rather unusual configuration where SIP/98765432198 was valid, although it is not impossible; normally there would be a sip.conf section name in the middle. SIP/100 is quite likely to be valid, although it violates best current practice - device names should not be guessable.
There was no requirement to avoid going to mobile voicemail, so cutting off the call to the mobile after 15 seconds is a bit excessive.
None of this will be useful if the OP is trapped behind a GUI.
With this current setup, wouldn’t two SIP lines be in use? The SIP line connecting the client to the PBX and then the SIP line connecting the PBX to cell phone.
SIP lines are a commercial concept, which doesn’t have very much physical significance.
If the SIP service provider allows it, and you don’t use conflicting features, you can use “directmedia=yes” to have the media stream bypass Asterisk. This is actually the default.
You could also try using the Transfer application, rather than the second Dial. There is a good chance that your SIP service provider will not support this, and it is almost certain that they will not push a transfer back to the PSTN, as that would stop them blling for the call. If doing this, it is stronly advised not to Answer the call, as post answer redirections in Asterisk are flakey, particularly when they go wrong.