Blind Transfer Drops after 30 second

Asterisk Asterisk SIP
1

Hi Fellas,
I have a situation
We have an issabel server (asterisk 11) which has 2 NICs
NIC1 for the connection to providers Intranet (10.105.0.0 network)
NIC2 for our local network and IP-Phones (172.16.0.0 network)
We have register sip accounts as a trunk with this configuration:
username=2222222
type=peer
trustrpid=yes
sendrpid=yes
secret=222222
qualify=yes
insecure=port,invite
host=10.105.X.X
disallow=all
allow=ulaw&alaw&gsm

and in my advanced sip i have:
NAT=Route
IPCONF=Static
ExternalIP: 10.105.Y.Y (MY NIC1 IP)
Local Net: 172.16.0.0 255.255.255.0
Reinvite Behavior: Yes

Everything works fine, but when i blind transfer a call to another phone or when i set always forward to another phone, the call will drop afters 30 second

2

You appear to be using some sort of GUI. Please use the support resources for the GUI.

3

Hi,
yes. but can you help me with troubleshoot it or the steps i should take.
Im in some difficult situation here

4

Read and understand the Asterisk dialplan involved in the call and transfer.

Uncomment full in logger.conf. Set verbosity to 5. Enable logging for the underlying channel technology. Look at the logs. If no obvious reason, set debug to at least five.

1 Like
5

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