Unable to place outbound calls via sip trunk

General Help
1

Hi all,

I am faced with a very peculiar situation so bare with me:

We have PRI and SIP trunking setup with our PBX. The PRI trunk is our primary source of communication and all inbound/outbound calls work fine. The SIP trunk (with sipstation) is our backup source and also hosts some of the DIDs including our main line number.

Incoming calls from the SIP trunk work perfectly fine (it can register to the provider and we have confirmation of that from sipstation support), but outgoing calls fail. The failure appears to be within Asterisk since I am not seeing any invites coming out.

Below is a snippet of the log:

Mar 28 13:30:35] VERBOSE[8513] pbx.c: – Executing [s@macro-dialout-trunk:16] Macro(“SIP/332-000000c4”, “dialout-trunk-predial-hook,”) in new stack
[Mar 28 13:30:35] VERBOSE[8513] pbx.c: – Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/332-000000c4”, “”) in new stack
[Mar 28 13:30:35] VERBOSE[8513] pbx.c: – Executing [s@macro-dialout-trunk:17] GotoIf(“SIP/332-000000c4”, “0?bypass,1”) in new stack
[Mar 28 13:30:35] VERBOSE[8513] pbx.c: – Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/332-000000c4”, “0?customtrunk”) in new stack
[Mar 28 13:30:35] VERBOSE[8513] pbx.c: – Executing [s@macro-dialout-trunk:19] Dial(“SIP/332-000000c4”, “SIP/fpbx/XXXXXXXXX,300,”) in new stack
[Mar 28 13:30:35] VERBOSE[8513] netsock.c: == Using SIP RTP TOS bits 184
[Mar 28 13:30:35] VERBOSE[8513] netsock.c: == Using SIP RTP CoS mark 5

[Mar 28 13:30:35] WARNING[8513] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
[Mar 28 13:30:35] VERBOSE[8513] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)

[Mar 28 13:30:35] VERBOSE[8513] pbx.c: – Executing [s@macro-dialout-trunk:20] NoOp(“SIP/332-000000c4”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20”) in new stack
[Mar 28 13:30:35] VERBOSE[8513] pbx.c: – Executing [s@macro-dialout-trunk:21] Goto(“SIP/332-000000c4”, “s-CHANUNAVAIL,1”) in new stack

It looks like the problem revolves around this snippet: app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Unknown) – but unfortunately I am not quite sure how to address that…

Any ideas/suggestions?

Thanks!

2

Unable to create channel of type ‘SIP’ (cause 20 - Unknown)

I am sure you saw that, what does ‘sip show status’ show? Also ‘sip show peer fpbx’

3

I am pretty sure the SIP settings are correct. We are behind a static IP firewall and I have entered our static IP address in the field, plus all applicable local networks. Codecs are correct… nothing really stands out as a potential problem.

The outbound routes have been in place for years - at least 2 or 3 and have been working for quite some time. This is a relatively new development… anything specific you guys would like me to paste?

Thanks!

4

Also your outbound route and outbound trunk settings could have a typo or something in them as well… Sorry I can’t help much more with what I see in the snippet you provided

5

Have you checked to make sure your Asterisk Sip Settings are correct?
Settings/Asterisk SIP Settings from the pull-down menu