David Allen
2005-Dec-20 06:39 UTC
[Asterisk-Users] 482 Loop Detected when transferring calls back to Asterisk
Hi, I want to be able to receive incoming calls via H323 to Asterisk for SIP Conversion and then send the Call to a seperate machine running SER to route the call to the end user CPE. However if the call is not answered, I want to be able to send that call back to the machine with Asterisk on it to a Voicemail Box. I have set it up this way, however I'm getting a Loopback Detected each time the call is sent back to the Asterisk Box, which then tries to connect the call as a fall back to a Local Channel listed in my default context. The Call Flow is as follows: -------->Asterisk (Performs H323 to SIP Conversion) ----------------------->Passes the call to SER -------------------------> UA/CPE is called | | | | --------------------- On timeout the call is sent back to the Asterisk Box -------------- to a Voicemail Box on that machine. Is there anyway around the loopback issue on the Asterisk Box (either by using another IP Address or by some how mangling the SIP Message)? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051220/7835e249/attachment.htm
Olle E. Johansson
2005-Dec-21 01:15 UTC
[Asterisk-Users] 482 Loop Detected when transferring calls back to Asterisk
David Allen wrote:> Hi, > > I want to be able to receive incoming calls via H323 to Asterisk for SIP > Conversion and then send the Call to a seperate machine running SER to > route the call to the end user CPE. However if the call is not answered, > I want to be able to send that call back to the machine with Asterisk on > it to a Voicemail Box. I have set it up this way, however I'm getting a > Loopback Detected each time the call is sent back to the Asterisk Box, > which then tries to connect the call as a fall back to a Local Channel > listed in my default context. > > The Call Flow is as follows: > > -------->Asterisk (Performs H323 to SIP Conversion) > ----------------------->Passes the call to SER > -------------------------> UA/CPE is called > | > | > | > | > --------------------- > On timeout the call is sent back to the Asterisk Box -------------- > to > a Voicemail Box on that machine. > > Is there anyway around the loopback issue on the Asterisk Box (either by > using another IP Address or by some how mangling the SIP Message)? >For now, you haver to make Asterisk is in control and times out on the call to the SER client and moves on to voicemail within the dialplan. /O