[asterisk-users] Dialing out and recording

[Henrik Westerberg]

Yes I should really upgrade, just have to make sure that asterisk-java
will work properly with 1.8

/H








Den 2013-01-02 22:25 skrev Danny Nicholas <danny at debsinc.com>:

>1.6.2 is a "deader soldier" than 1.4.X.
>
>-----Original Message-----
>From: asterisk-users-bounces at lists.digium.com
>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Henrik
>Westerberg
>Sent: Wednesday, January 02, 2013 3:20 PM
>To: asterisk-users at lists.digium.com
>Subject: Re: [asterisk-users] Dialing out and recording
>
>#2 works for me on Asterisk 1.8.12 when setting the header like this:
>
>exten => _S,n,SipSetHeader("Diversion: " ${CALLERID(rdnis)})
>
>I haven't been able to make it work on 1.6 yet though, has anyone else?
>
>
>/Henrik
>
>
>>
>>
>>
>> 
>>
>>From: asterisk-users-bounces at lists.digium.com
>>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Don Kelly
>>Sent: Wednesday, January 02, 2013 9:32 AM
>>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>>Subject: Re: [asterisk-users] Dialing out and recording
>>
>> 
>>
>>I have the same requirement, but it's important that the caller ID
>>information from the original caller is presented to the destination
>>and we announce the call before the "transfer" is complete. The carrier
>>requires a diversion header if the ANI is not one of "our" DIDs. Does
>>someone have experience with this working?
>>
>>--
>>
>>Two suggestions for you, Don.  #1 if the Dial is "Private" the
>>"announcement" is taken care of. #2 I'm supposing that you could do a
>>"SIP Header" command before the Dial to resolve the diversion header
>>issue.
>>
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