Transfer call to user

Asterisk Asterisk Support
1

Hi having a small issue,I can transfer a call to an exten but I cant seem to get it to call a users ID

for example I have in user.conf

[8000]
username=8000
secret=password
accountcode=8000
callerid="8000"
mailbox=8000
context=default
type=friend
host=dynamic
disallow = all
allow = g729,gsm,ulaw,alaw,h263

and in my extension I want all incoming call that for example say are called in 8000@IPADDRESS to call that user

I can call extensions in other context like 999 which is a simple echo test

code I use is

exten => _.,1,Answer
exten => _.,n,Dial(SIP/${EXTEN})

if anyone can help that would be great

2

Ok I was able to get it to work but now when I answer the call it hangs up and give me this error

[Jun 30 07:00:42] WARNING[4818]: channel.c:3908 ast_channel_make_compatible: No path to translate from SIP/i-338e715c.phono.com-00000000(4) to SIP/8000-00000001(524544) [Jun 30 07:00:42] WARNING[4818]: app_dial.c:1874 dial_exec_full: Had to drop call because I couldn't make SIP/i-338e715c.phono.com-00000000 compatible with SIP/8000-00000001

3

Do you have the G.729 codec installed and licensed? If not, remove it from your allowed codecs.

4

Lets say I need to transfer a call to say other sip phone example 6523

how can I transfer the call using my Zoiper phone to 6523

5

It is not clear what you mean by transfer. You appear to be trying to interconnect the phones, not do an actual transfer.

The reason you are having troubles internconnecting the phones is that you have allowed the two phones to select incompatible codecs, but you do not have the codec translation modules needed to resolve that. In particular, it is likely that one phone is using G.729 and another cannot accept it. G.729 has restrictive patents, so you must buy a licence before you can interconnect G.729 to soemthing not using G.729.