[Asterisk-Users] How to transfer a call??

Pertti Pikkarainen ppik at lanwan.fi
Fri Mar 14 06:15:14 MST 2003


I have it like this

exten => 9998,1,Dial,SIP/9998|30|t

30 is a timeout value
Check 'show application dial'


WipeOut ™ wrote:

>What is the correct syntax to use the 't' option??
>
>This is the current line in my extensions.conf
>exten => 9998,1,Dial,SIP/9998
>So would I change it to 
>exten => 9998,1,Dial,SIP/9998,t
>
>Thanks.
>
>----- Original Message -----
>From: Pertti Pikkarainen <ppik at lanwan.fi>
>Date: Fri, 14 Mar 2003 13:50:21 +0200 
>To: asterisk-users at lists.digium.com
>Subject: Re: [Asterisk-Users] How to transfer a call??
>
>  
>
>>Negative side effect with 't' option:  all the local SIP-to-SIP media
>>streams travel trough Asterisk instead of going direct.
>>
>>Right now I'm using SNOM's transfer option instead.
>>But now I can't use *  call parking  because of that. Using  #  is 
>>probably better
>>if there are no scaling problems.
>>
>>Regards Pertti
>>
>>
>>
>>Steven Critchfield wrote:
>>
>>    
>>
>>>If you search the archives you would find that for IP phone you need to
>>>add a 't' option to the end of your dial command. The 't' option will
>>>let the user dial '#' to get the systems attention, then dial an
>>>extention for the transfer.
>>>
>>>On Fri, 2003-03-14 at 03:32, =?iso-8859-1?B?V2lwZU91dCCZ ?= wrote:
>>> 
>>>
>>>      
>>>
>>>>Hi,
>>>>
>>>>Firstly let me start off by saying that asterisk is one of the most amazing pieces of open source I have seen, it rates right up there with Apache, OpenOffice, MySQL and even Linux itself.. Nice work!!
>>>>
>>>>I have just installed my first server, thanks to the astinstall script.. and I have read the Handbook (ver 1) and the white paper PDF's.. and I have managed to setup 2 extentions and make calls between them using MSN Messenger, nothing fantastic but its a start..
>>>>
>>>>One answer is still missing.. How do I transfer a call to another ext?? I am looking at only using IP phones and so for the test system I am using MSN Messenger.. The final solution will probably use a linux softphone line gnophone or linphone..
>>>>
>>>>All I have been able to find in the docs about call transfer is using a normal phone handset and hook-flash (not quite sure what that it, I am new to telephony)..
>>>>
>>>>So I guess what I am asking is what do I need to configure or do to be able to transfer a call from one IP ext to another??
>>>>
>>>>Thanks.. 
>>>>   
>>>>
>>>>        
>>>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>    
>>
>
>  
>

-- 

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Nordic LAN&WAN Communication Oy
Pertti Pikkarainen
vp of engineering
E-Mail: ppik at lanwan.fi
tel: +358-9-5024100
fax: +358-9-5023840
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