What are the criteria here.
What do I need to change in the SIP message so
that asterisk will not consider it looped??
Thanks for any help
Regards
tomasz
On Nov 23, 2007 4:03 PM, Tomasz Zieleniewski <tzieleniewski@gmail.com> wrote:
hi,
I use asterisk as a gateway which forwards external calls from pstn to
my internal sip network.
all sip signaling is passed to sip proxy.
I also use asterisk as a voicemail server.
everything works well when calls are passed to asterisk from local network.
but when calls are forwarded from asterisk to sip proxy and then sip
proxy decides to pass it back to asterisk
waorking as a voicemail server
asterisk complains about the loop and returns 482 response.
Can it be somehow reconfigured??
Thanks in advance
TOmasz
No comments:
Post a Comment