GoTalk Transit calls

Asterisk Asterisk Support
1

I know I used to be able to do this with older versions of Asterisk, however since 1.8, I cannot redirect a call (from the dial plan using Local) back out the same trunk to a mobile for instance. The call works but the media is absent. When I do an RTP debug, I see that the Asterisk server is not bridging the call as I would be expecting with directmedia=no in sip.conf.

What is interesting is that it works fine with another provider and I do see the RTP which indicates that it is Asterisk that is deciding not to bridge the RTP packets. I think GoTalk uses Asterisk and the other provider uses Broadsoft.
Is there any config you can think of that could solve this problem. What is it doing?

Thanks
Mike

sip.conf

register=09448848:@202.169.178.10

[gwy1]
defaultuser=09448848
secret=
type=peer
host=202.169.178.10
context=incoming-group1
fromuser=09448848
fromdomain=sip.gotalk.com
port=5060
directmedia=no
qualify=30000

[general]
language=en_AU
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
qualify=no
insecure=port,invite
dtmfmode=rfc2833
allowguest=no
realm=ipcscm
checkmwi=7
notifyringing=yes
notifyhold=yes
useragent=IPCSCM
srvlookup=no
autocreatepeer=yes
context=DialPlan99
session-timers=refuse

2

Asterisk doesn’t have trunk concept, so that fact that it is the same “trunk” should be irrelevant.

Using the /n option on the Local channel may work as a workaround.

What was the SDP for the call? Was there any re-invite (1.8 will re-invite for connected line id)?

Why did you need a local channel? As described, Goto should be enough.