Inbound calls give busy tone

Asterisk Asterisk Support
1

Hi,

I have AsteriskNow Server and TalkSwitch IP PBX.
I created a SIP Trunk in AsteriskNow server with the following settings:

General Settings
Trunk name: 158
Outbound Caller ID: 158
Outgoing Settings
Trunk Name: T158
PEER Details:
host=90.0.0.110
username=158
secret=Removed
type=peer
nat=yes
Incoming Settings
USER Context: 158
USER Details:
context=from-talkswitch
type=user
secret=Removed
Registration
Register String:
158: Removed@90.0.0.110/158

Where 90.0.0.110 is the IP of the TalkSwitch PBX.

The registration is successful, I can call from Asterisk to any extension in TalkSwitch also I can call PSTN and VIOP lines through TalkSwitch from Asterisk.

The problem, I cannot receive any calls into Asterisk through TalkSwitch, I always get busy tone.

Below is what I get in Asterisk log when calling from Analog phone connected to TalkSwitch into the SIP Trunk in Asterisk:
[Jan 22 10:28:20] VERBOSE[2732] netsock.c: == Using SIP RTP TOS bits 184
[Jan 22 10:28:20] VERBOSE[2732] netsock.c: == Using SIP RTP CoS mark 5

While below is what I get in Asterisk log when calling from IP phone connected to TalksSwitch into the SIP Trunk in Asterisk:
[Jan 22 19:37:48] VERBOSE[2725] netsock.c: == Using SIP RTP TOS bits 184
[Jan 22 19:37:48] VERBOSE[2725] netsock.c: == Using SIP RTP CoS mark 5
[Jan 22 19:37:48] WARNING[2725] chan_sip.c: username mismatch, have <158>, digest has <001A7E151>
[Jan 22 19:37:48] NOTICE[2725] chan_sip.c: Failed to authenticate device “My Office” sip:151@90.0.0.110;tag=2632853575642027014

Any idea why I am getting the busy tone? And how to fix this problem?

2

What makes you think that the user name is the insecure one that you are expecting rather than the secure looking one that you are receiving? I.E. why don’t you believe the user id mismatch message?

Also, note that there is a specific forum for AsteriskNow, and I don’t know the details of how AsteriskNow maps its user interface abstraction into Asterisk concepts.

3

Have you seen this thread?

http://forums.digium.com/viewtopic.php?f=1&t=75995&sid=24c1044dd0d6ebb5f1d9802616cf7544

4

Finally it is working now.

What I did is just add “insecure=very” in the outgoing settings and make the incoming settings empty.

Now, outgoing calls and incoming calls are working fine.

Both systems Talkswitch and Asterisk look like one IP PBX box only.

5

Please note that you are avoiding ( by reducing system security) the problem, not fixing it (making the authentication user match on both sides of the connection).

6

Thanks for the input. Can you explain how you can tell from that error message that the authuser setting is the problem? I’ve never quite understood how username, fromuser and authuser differ. Any info or links which explain this would be appreciated… haven’t found alot by googling and I’d prefer to understand than to use trial and error methods to cure such problems.

7

158 is yours, 001A7E151 is theirs.

8

158 is yours, 001A7E151 is theirs.[/quote]

I see that, but is that referring to username or authuser - and how can you tell?

9

Thank you for that.
Oky, how would I fix this problem without reducing the security?

10

I’ve never had to do this in anger, but it looks like match_auth_username may help.