[asterisk-users] Trying to do a transfer in agi

David Jones david.jones at davidbrummy.com
Sat Jan 24 22:31:25 CST 2009


> Hey,
>
> I am trying to work through a use case requirement where a user  
> listens to a some advertisement and then if at the end off it they  
> press a key they press a 1 key they get transfered to a pre-defined  
> number.  I am using the asterisk java library at http://asterisk-java.org/ 
> .
>
> I was able to originate the call easily by doing a
>
> String channel="SIP/" + phoneNumber + "@"+getSipPeer();
> 			this.getAsteriskServer().originateToExtension(channel,  
> getContext(), getExtension(), getPriority(), getTimeout(),  
> getCallerId(), vars);
>
> my extensions.conf file has this defined in it
>
>>>
>>>
>>> exten => 9001,1,Agi(agi://127.0.0.1:4573/${campaign}.agi?campaign=$ 
>>> {campaign})
>>
>>>
>
>
> A fast AGI script is then called and I can play the media.
>
> My first option was at the end of the script to set the extension to  
> continue at 9002

     	
  channel.setContext("davidtest");
  channel.setExtension("9002");
  channel.setPriority("1");

>
> exten => 9002,1,Transfer(SIP/${campaignNumber}@SER1)
>
> When I tried this I got the following in the logs and the call would  
> be dropped.
>
>>> [Jan  8 23:33:35] VERBOSE[16163] logger.c: --- (8 headers 0 lines)  
>>> ---
>>> [Jan  8 23:33:39] VERBOSE[16163] logger.c:
>>> <--- SIP read from 10.128.181.23:5060 --->
>>> SIP/2.0 403 Forbidden^M
>>> Via: SIP/2.0/UDP
>>> 209.34.91.75
>>> :5060;received=10.128.181.21;branch=z9hG4bK6937f2b6;rport=5060^M
>>> From: "companyname" <sip:+14158888888 at 209.34.91.75>;tag=as1e339e25^M
>>> To: <sip:4159999999 at 209.34.91.74>;tag=gK028a94eb^M
>>> Call-ID: 4b07e5ef5f04f5dc40ce7d6b4b002d5f at 209.34.91.75^M
>>> CSeq: 103 REFER^M
>>> Content-Length: 0^Mites
>
>
> I did try to execute the Transfer command directly in the Fast AGI  
> script.  The call would not drop but it did not connect.  I saw no  
> errors in the logs.
>

int reply = channel.exec("Transfer", "SIP/4159999999 at SER1");

> I am new to Asterisk and VOIP so any help would be appreciated,
>
> cheers,
> David
>
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