Trouble with my sip trunk

General Help
1

Good day,

I am having trouble with my sip trunk, My provider of the account is not being very helpful and just say that they dont support 3rd party devices.

Could someone please tell me if my configs in the trunk are on the right track?

OUTGOING SETTINGS:

type=peer
insecure=invite
qualify=yes
dtmfmode=rfc2833
host=sip.nexus.co.za
username=00000000000 (change from original)
fromuser=00000000000 (change from original)
secret=password
disallow=all
allow=g729,ulaw,alaw,ilbc,g723,gsm

INCOMING SETTINGS:

secret=Edzarsddr7
type=user
context=from-trunk
disallow=all
allow=ulaw,alaw,ilbc,g723,gsm

I just want to set it up as a basic trunk, I only have one (this one) and I want all my calls to be forwarded to it.

How ever my Outbound route keeps giving me the congested message, but my sip account can have multiple concurrent calls so I don’t see why it would be doing this?

2

I have been doing some debugging,

I can make incoming calls happily, and all that works,
But my outbound route always gives me the congested message?
This must have something to do with the outbound calls on my trunk?

3

I have Attached the log:

[2014-05-05 04:33:28] VERBOSE[1705][C-00000005] netsock2.c: == Using SIP RTP TOS bits 184
[2014-05-05 04:33:28] VERBOSE[1705][C-00000005] netsock2.c: == Using SIP RTP CoS mark 5
[2014-05-05 04:33:28] VERBOSE[1941][C-00000005] pbx.c: – Executing [972592180362@from-sip-external:1] NoOp(“SIP/41.132.147.142-00000005”, “Received incoming SIP connection from unknown peer to 972592180362”) in new stack
[2014-05-05 04:33:28] VERBOSE[1941][C-00000005] pbx.c: – Executing [972592180362@from-sip-external:2] Set(“SIP/41.132.147.142-00000005”, “DID=972592180362”) in new stack
[2014-05-05 04:33:28] VERBOSE[1941][C-00000005] pbx.c: – Executing [972592180362@from-sip-external:3] Goto(“SIP/41.132.147.142-00000005”, “s,1”) in new stack
[2014-05-05 04:33:28] VERBOSE[1941][C-00000005] pbx.c: – Goto (from-sip-external,s,1)
[2014-05-05 04:33:28] VERBOSE[1941][C-00000005] pbx.c: – Executing [s@from-sip-external:1] GotoIf(“SIP/41.132.147.142-00000005”, “0?checklang:noanonymous”) in new stack
[2014-05-05 04:33:28] VERBOSE[1941][C-00000005] pbx.c: – Goto (from-sip-external,s,5)
[2014-05-05 04:33:28] VERBOSE[1941][C-00000005] pbx.c: – Executing [s@from-sip-external:5] Set(“SIP/41.132.147.142-00000005”, “TIMEOUT(absolute)=15”) in new stack
[2014-05-05 04:33:28] VERBOSE[1941][C-00000005] func_timeout.c: – Channel will hangup at 2014-05-05 04:33:43.039 UTC.
[2014-05-05 04:33:28] VERBOSE[1941][C-00000005] pbx.c: – Executing [s@from-sip-external:6] Log(“SIP/41.132.147.142-00000005”, "WARNING,“Rejecting unknown SIP connection from 192.168.1.1"”) in new stack
[2014-05-05 04:33:28] WARNING[1941][C-00000005] Ext. s: “Rejecting unknown SIP connection from 192.168.1.1”
[2014-05-05 04:33:28] VERBOSE[1941][C-00000005] pbx.c: – Executing [s@from-sip-external:7] Answer(“SIP/41.132.147.142-00000005”, “”) in new stack
[2014-05-05 04:33:28] VERBOSE[1941][C-00000005] pbx.c: – Executing [s@from-sip-external:8] Wait(“SIP/41.132.147.142-00000005”, “2”) in new stack
[2014-05-05 04:33:30] VERBOSE[1941][C-00000005] pbx.c: – Executing [s@from-sip-external:9] Playback(“SIP/41.132.147.142-00000005”, “ss-noservice”) in new stack
[2014-05-05 04:33:30] VERBOSE[1941][C-00000005] file.c: – <SIP/41.132.147.142-00000005> Playing ‘ss-noservice.ulaw’ (language ‘en’)
[2014-05-05 04:33:35] VERBOSE[1941][C-00000005] pbx.c: – Executing [s@from-sip-external:10] PlayTones(“SIP/41.132.147.142-00000005”, “congestion”) in new stack
[2014-05-05 04:33:35] VERBOSE[1941][C-00000005] pbx.c: – Executing [s@from-sip-external:11] Congestion(“SIP/41.132.147.142-00000005”, “5”) in new stack
[2014-05-05 04:33:40] VERBOSE[1941][C-00000005] pbx.c: == Spawn extension (from-sip-external, s, 11) exited non-zero on ‘SIP/41.132.147.142-00000005’
[2014-05-05 04:33:40] VERBOSE[1941][C-00000005] pbx.c: – Executing [h@from-sip-external:1] Hangup(“SIP/41.132.147.142-00000005”, “”) in new stack
[2014-05-05 04:33:40] VERBOSE[1941][C-00000005] pbx.c: == Spawn extension (from-sip-external, h, 1) exited non-zero on ‘SIP/41.132.147.142-00000005’
[2014-05-05 04:34:00] WARNING[1705] chan_sip.c: Retransmission timeout reached on transmission 5fd94e43cfe694236123050c01ee5a64 for seqno 1 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

4

Check your NAT settings in “Asterisk SIP Settings”. Make sure you have NAT set and have specified your external address correctly and your local networks.

Also It appears that you are getting inbound SIP connections because you probably have “Allow anonymous SIP connections” set to Yes but these are being handle by your trunk settings.

5

I have enabled them: please see

Should allow anonymous be disabled or enabled?

I did see in the log file now:

[2014-05-05 05:30:44] VERBOSE[2667][C-0000001c] pbx.c: – Executing [s@macro-dialout-trunk:22] Dial(“IAX2/6-2549”, “SIP/nexus/824419917,300,Tt”) in new stack
[2014-05-05 05:30:44] VERBOSE[2667][C-0000001c] netsock2.c: == Using SIP RTP TOS bits 184
[2014-05-05 05:30:44] VERBOSE[2667][C-0000001c] netsock2.c: == Using SIP RTP CoS mark 5
[2014-05-05 05:30:44] VERBOSE[2667][C-0000001c] app_dial.c: – Called SIP/nexus/824419917
[2014-05-05 05:30:45] VERBOSE[2667][C-0000001c] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)

it is saying the trunk is congested?

6

Anonymous SIP connections should be “No” if the trunk is configured correctly. Also “Allow SIP Guest” should be set to No as well.

Set NAT to Yes.

Try that.

7

There are so many issues with your config I don’t know where to start.

Did you read the post “read this before posting” because you did not include the information we need to fully assist you.

Right off the bat insecure=very is deprecated. Also why would you declare insecure then set a username and password?

You have a hodge podge of CODEC’s selected and I am sure you have not purchased g.729 licenses.