[asterisk-dev] Blind Transfer is not working in incoming calls
Jeff Gehlbach
jeffg at jeffg.org
Mon Aug 25 10:49:38 CDT 2008
Gentlemen, this discussion belongs on the asterisk-users list. Please
move it there.
-jeff
On Aug 25, 2008, at 10:00 AM, Serge Berney wrote:
> I experiment the same problem… On some call, there’s no possibility
> to transfer the call (caller ear DTMF sounds) – AST 1.4.21.
>
> For Max :
> DialPlan must be like this :
>
> exten => 99, 1, Answer()
> exten => 99, n, Dial(SIP/sip_user1, 20, tT)
> exten => 99, 1, Hangup()
>
> Regards
>
>
> De : asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com
> ] De la part de Fernando Urzedo
> Envoyé : lundi, 25. août 2008 13:49
> À : Asterisk Developers Mailing List
> Objet : Re: [asterisk-dev] Blind Transfer is not working in incoming
> calls
>
> Hi Max,
>
> Please make sure you are setting the Dial command with parameters
> "T" and "t". I would say you are missing "t"...
>
> Regards!
>
> From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com
> ] On Behalf Of Max Alex
> Sent: sábado, 23 de agosto de 2008 03:53
> To: asterisk-dev at lists.digium.com
> Subject: [asterisk-dev] Blind Transfer is not working in incoming
> calls
> Hi Everybody,
> i have installed asterisk 1.4.19 on my box,
> I have setup agi script which is used while incoming and outgoing
> calls.
> It will find the users for incoming and calls to them which is
> registered in asterisk,
> I have a setup *# for blind transfer to call any outbound or inbound
> numbers.
> when i am calling any outbound call and the calls are connected with
> my sip peer, then i am pressing *# for blind transfer, it will ask
> me to enter the transfer number and it is working,
> But when an incoming call to my sip user and they are connected the
> *# is not worked even the transfer prompt is also played, and dtmf
> is also set properly.
>
> But i am not getting why the incoming call is not transfer to any
> other number?
> Please help for this issue!
>
>
> --
> Thanks,
> Max Alex
> Voip Developer
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