SIP Trunk Transfer then Transfer Back

Asterisk Asterisk Support
1

Hi -
After having audio issues with IAX2 Trunks. I decided to use SIP Trunk (BoxA and BoxB). The audio sounds great. After testing things out - I noticed I could not do a double transfer. for example My phone is connected to BoxA. Dialing another phone across the SIp trunk to BoxB. That user can trasfer me to any phone connected to that box(BoxB) but if he tries to transfer me back to BoxA I get a “All circuits are busy” hangup cause 21. BUT if I use “CID NUm Alias” on the phone that’s connected to BoxA and masquerade the num. all the transfers work with that phone???

What am i doing wrong :smile:
Thanks
Joe
1.6.2.20(both servers)
Sample config
BoxA
type=friend
host=192.168.8.1
insecure=very
qualify=yes
context=from-internal
allowtransfers=yes
disallow=all
allow=ulaw

BoxB
type=friend
host=192.168.11.1
insecure=very
qualify=yes
context=from-internal
allowtransfers=yes
disallow=all
allow=ulaw

2

It’s a known bug. It may have been fixed in the latest 1.8.x. It will never be fixed in 1.6.2.x.

Some code that is intended to prevent a line being used more than once when forking a call with the & operator in Dial, also triggers on this case.

3

do you think they will fix the audio on th IAX2 trunk?

Thanks
Joe

4

Nothing will get fixed in 1.6.2 unless it is a security vulnerability or a bug introduced in fixing one.

5

Note the bug I’m thinking of is not in the SIP handling.