Incoming call - AsteriskNow gives busy signal

FreePBX Installation / Upgrade
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

I also enabled “Allow Anonymous Inbound SIP Calls” in the General Settings.

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 signal.

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:[email protected];tag=2632853575642027014

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

2

can you check it enabling sip debug from the console with:

sip set debug on

It’s possible you’ll face with something like:

-- Executing [0123456789@from-sip-external:1] NoOp("SIP/192.168.1.100-00000003", "Received incoming SIP connection from unknown peer to 0123456789") in new stack
-- Executing [0123456789@from-sip-external:2] Set("SIP/192.168.1.100-00000003", "DID=0123456789") in new stack
-- Executing [0123456789@from-sip-external:3] Goto("SIP/192.168.1.100-00000003", "s,1") in new stack

it runs the [from-sip-external] (just an example… not had to be the same) and maybe it has a definitin of Congestion in the plan…

3

Finally, I got it 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.

4

Well, you are all over the place, insecure=very is not the correct format anymore (nor has been for 4 versions). The arguments are port and invite.

Insecure tells Asterisk to ignore authentication on inbound.

You don’t have a context.

You made need to set the fromuser or fromdomain variables.

See Asterisk sample sip.conf for an explanation of all the settings.

5

Hi, I found your post and I believe I set up everything correctly as follows:

Trunk Name: 199
Outbound CallerID: 199

Outgoing Settings:
Truck Name: T199
host=10.1.80.10
username=199
secret=199
type=peer
nat=yes
insecure=very

Incoming settings is blank.

Register is "199:[email protected]/199"

I have set up an extension on the Talkswitch to match as extension 199 with a user name of 199 and a password of 199 (I will increase the security once I get this to work).

The issue is that it won’t even register. I get a connection refused in the log files which would indicate either 1) my credentials are incorrect or there is an addressing or port issue. I believe it is not 2) because I have used an external SIP client with exactly the same settings and it works perfectly and registers. Therefore I believe it must be something in the way Asterisk is passing in the credentials.

Any help greatly appreciated!!

Thanks

6

Thanks for the feedback Scott. I was working off the original post from jalmod which said to use insecure=very to fix his incoming calls. I am new to Asterisk and just testing it out so this is my first try.

I will investigate your other suggestions. Obviously something has changed since when it worked for jalmod as they weren’t listed in his (presumably) working example.