SPA8000 and analogue devices with FreePBX

General Help
1

Hey everyone,

I’m working on just an internal PBX structure (does not need to reach a provider or anything, just local communications between devices). I have an analogue phone behind a Cisco SPA8000, which (I believe) I’ve configured correctly. However, calls to the analogue (ext 105) from a softphone (ext 101) give the following error message in the asterisk log:

[2015-04-27 13:47:50] VERBOSE[30597][C-0000001b] app_dial.c: – Called SIP/105
[2015-04-27 13:47:50] VERBOSE[30597][C-0000001b] app_dial.c: – Connected line update to SIP/101-0000002d prevented.
[2015-04-27 13:47:50] VERBOSE[1635][C-0000001b] chan_sip.c: – Got SIP response 503 “Service Unavailable” back from 10.1.3.118:5060
[2015-04-27 13:47:50] VERBOSE[30597][C-0000001b] app_dial.c: – SIP/105-0000002e is circuit-busy

On the other hand, calls from 105 to 101 don’t show anything in the asterisk log. This leads me to believe that there is an issue with the SPA8000 configuration itself. FreePBX does show it as a registered device and that it is live.

Here’s the SPA8000 config for L1 (105):

[Album] SPA8000 Config

Also, under SIP, RTP Packet Size has already been modified to 0.020. Let me know if there’s anymore information I can provide, thanks in advance!

2

Try to change SIP Port to 5061 and the return of sip show peers in CLI.

3

Sure, here’s the result:

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
101/101 10.1.0.70 D No No A 59311 OK (2 ms)
105/105 10.1.3.118 D No No A 5061 OK (9 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]

And the log when I try to call it:

[2015-05-05 16:57:57] VERBOSE[26742][C-0000001d] netsock2.c: == Using SIP RTP TOS bits 184
[2015-05-05 16:57:57] VERBOSE[26742][C-0000001d] netsock2.c: == Using SIP RTP CoS mark 5
[2015-05-05 16:57:57] VERBOSE[26742][C-0000001d] app_dial.c: – Called SIP/105
[2015-05-05 16:57:57] VERBOSE[26742][C-0000001d] app_dial.c: – Connected line update to SIP/101-00000030 prevented.
[2015-05-05 16:57:57] VERBOSE[1635][C-0000001d] chan_sip.c: – Got SIP response 503 “Service Unavailable” back from 10.1.3.118:5061
[2015-05-05 16:57:57] VERBOSE[26742][C-0000001d] app_dial.c: – SIP/105-00000031 is circuit-busy
[2015-05-05 16:57:57] VERBOSE[26742][C-0000001d] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0)

Still seems to be the same result though, any other suggestions or information you’d like me to grab?

4

You don’t have a map for your extensions in your dialstring.

5

On the cisco side or the asterisk side?

6

You you can deduce that from :-

7

Sorry if I’m still new to this and it’s not entirely clear, but I do appreciate your help!

Where would you configure a dial plan for the asterisk host? In a trunk?

8
9

Ah ok, I think I see what you’re referring to. However, what I am still confused about is that a dial plan on the cisco would affect 105 to 101. The log, however, is from 101 to 105. Dial plans are only outbound configurations, not inbound. Still, I presume a dialstring that only dials 3 digit extensions would simply be (xxx)? There’s no need for any real phone numbers, after all.

Sorry if I’m completely off base, just trying to understand. Thanks again

10

Try thiis:








11

Is not, by any logic, 101 making an outbound call? It can dial only what you allow in your dial plan, it is connected to an fxs port on your Cisco device, not your asterisk server, correct?

[edit]
oops , I reread that 101 is on asterisk not your box,

sip show peer 105

might be helpful.

12

Thanks! One of the configuration changes in this made it work. Calls are going through now.

[2015-05-06 13:29:36] VERBOSE[24892][C-0000001f] netsock2.c: == Using SIP RTP TOS bits 184
[2015-05-06 13:29:36] VERBOSE[24892][C-0000001f] netsock2.c: == Using SIP RTP CoS mark 5
[2015-05-06 13:29:36] VERBOSE[24892][C-0000001f] app_dial.c: – Called SIP/105
[2015-05-06 13:29:36] VERBOSE[24892][C-0000001f] app_dial.c: – Connected line update to SIP/101-00000032 prevented.
[2015-05-06 13:29:36] VERBOSE[24892][C-0000001f] app_dial.c: – Connected line update to SIP/101-00000032 prevented.
[2015-05-06 13:29:36] VERBOSE[24892][C-0000001f] app_dial.c: – SIP/105-00000033 is ringing
[2015-05-06 13:29:38] VERBOSE[24892][C-0000001f] app_macro.c: == Spawn extension (macro-dial-one, s, 44) exited non-zero on ‘SIP/101-00000032’ in macro ‘dial-one’
[2015-05-06 13:29:38] VERBOSE[24892][C-0000001f] app_macro.c: == Spawn extension (macro-exten-vm, s, 16) exited non-zero on ‘SIP/101-00000032’ in macro ‘exten-vm’

Not quite sure which change specifically did it, but if I had to guess it would be something in the extension configuration.

Regardless, thanks!

EDIT:

Looks like it was on the asterisk side. Either DTMF or NAT mode