Incoming calls hang up

General Help
1

I have the latest stable version, PBX Firmware: 2.210.62-3
PBX Service Pack: 1.0.0.0

I have my SIP trunk configured with inbound route and IVR configured.
Inbound works in the fact that i reach my announcement and can select my different extensions. But incoming calls can only last for 25 to 32secs. If i just call in and listen to the welcome it repeats almost twice and the end of the twice it just hangs up, or if i call in and select an extension it stays up for 25secs or so. Does anybody know of some configuration that is causing this?
I tried making various changes on the trunk, inbound route, general settings, asterisks SIP settings.

Any help would be great!

Thanks

2

Who is your trunk provider and can you post the trunk definition?

3

Your context is wrong. It’s just from-pstn

May I ask where you got the incorrect syntax from? It makes not sense at all.

4

The trunk provider is Think-tel, the definition is=
type=peer
username=2262144610
secret=*****
host=206.80.250.100
context=from-2262144610&from-pstn
insecure=port,invite

Thanks

5

Made no difference. Do i have the correct peer details and user details?
Peer Details:
type=friend
insecure=port,invite
username=2262144610
secret=******
host=206.80.250.100
context=from-pstn

Incoming Settings
User Context: pstn
User Details: context=from-pstn

Thanks

6

I have no way of knowing if you have the correct details. They should have been provided by your provider.

It doesn’t make a tremendous amount of sense, you have a username and secret however you also have the insecure directive that tells Asterisk to ignore them.

Incoming settings is only used in the event that the inbound peer settings are different than the outbound. You only provided a context, that’s not a complete peer.

Have you read any documentation?

The Asterisk sip.conf sample explains every variable (there are about 100).

7

Not wanting to lead you on a wild goose chase but along with what SkyKing has said; have you examined the logs when placing the call… do you see a message of “Retransmission timeout reached on transmission” when the call drops?

8

Also what are your SIP NAT settings?

9

So my problem was.
Under Asterick SIP settings i had the IP configuration as static. Changed that to public and works great!