Hi,<br>I'm currently using Asterisk 1.4.23.1, and I have a problem (also on previous version).<br>Sometimes, when I try to do an attended transfer to another internal with default feature *2, Asterisk doesn't make it (it doesn't play 'pbx-transfer'). Sometimes on second time, Asterisk make transfer correctly. I have this problem on variuos type of SIP phones (GrandStream, Aastra, OKI).<br>
<br>My sip.conf is like the following account:<br><br>=======================================<br>[intphones](!)<br>type=friend<br>qualify=yes<br>host=dynamic<br>callgroup=1<br>pickupgroup=1<br>dtmfmode=sip<br><br>[1](intphones)<br>
context=IntPhones<br>username=1<br>secret=1234<br>amaflags=documentation<br>accountcode=11<br>subscribecontext=IntPhones<br>callerid="phone 11" <11><br>limitonpeers=yes<br>call-limit=100<br><br>
[2](intphones)<br>
context=IntPhones<br>
username=2<br>
secret=1234<br>
amaflags=documentation<br>
accountcode=12<br>
subscribecontext=IntPhones<br>
callerid="phone 12" <12><br>
limitonpeers=yes<br>
call-limit=100<br>=======================================<br><br>and on extensions.conf my dial lines are like:<br><br>=======================================<br>exten => _1X,1,Dial(SIP/${EXTEN:1},,tTr)<br>exten => _1X,n,Hangup()<br>
=======================================<br><br><br><br>Can anyone help me? I don't underwstand where I make the mistake!<br><br>Thanks to everyone<br><br>Marco<br>