Cant call a certain number. Is this a SIP provider issue or freepbx?

FreePBX
1

I am trying to call a number and keep getting “486 busy here” on the phone. Is this a SIP provider issue? I can call other numbers. Here is the log entry that occurs when I call this certain number:

[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] netsock2.c: Using SIP RTP TOS bits 184
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] netsock2.c: Using SIP RTP CoS mark 5
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] app_dial.c: Called SIP/fpbx-1-Account/9073607482
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] app_dial.c: SIP/fpbx-1-Account-0000104a redirecting info has changed, passing it to SIP/110-00001049
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] app_dial.c: SIP/fpbx-1-Account-0000104a is busy
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] app_dial.c: Everyone is busy/congested at this time (1:1/0/0)
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] pbx.c: Executing [s@macro-dialout-trunk:24] NoOp(“SIP/110-00001049”, “Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 19”) in new stack
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] pbx.c: Executing [s@macro-dialout-trunk:25] GotoIf(“SIP/110-00001049”, “0?continue,1:s-BUSY,1”) in new stack
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] pbx_builtins.c: Goto (macro-dialout-trunk,s-BUSY,1)
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] pbx.c: Executing [s-BUSY@macro-dialout-trunk:1] NoOp(“SIP/110-00001049”, “Dial failed due to trunk reporting BUSY - giving up”) in new stack
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] pbx.c: Executing [s-BUSY@macro-dialout-trunk:2] PlayTones(“SIP/110-00001049”, “busy”) in new stack
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] pbx.c: Executing [s-BUSY@macro-dialout-trunk:3] Busy(“SIP/110-00001049”, “20”) in new stack
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] app_macro.c: Spawn extension (macro-dialout-trunk, s-BUSY, 3) exited non-zero on ‘SIP/110-00001049’ in macro ‘dialout-trunk’
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] pbx.c: Spawn extension (from-internal, 9073607482, 7) exited non-zero on ‘SIP/110-00001049’
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] pbx.c: Executing [h@from-internal:1] Macro(“SIP/110-00001049”, “hangupcall”) in new stack
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/110-00001049”, “1?theend”) in new stack
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/110-00001049”, “0?Set(CDR(recordingfile)=)”) in new stack
[2017-04-10 15:15:33] VERBOSE[3549][C-00000633] pbx.c: Executing [s@macro-hangupcall:4] Hangup(“SIP/110-00001049”, “”) in new stack

2

From ringofsaturn.com

Cause No. 19 - no answer from user (user alerted).

This cause is used when the called party has been alerted but does not
respond with a connect indication within a prescribed period of time.
Note - This cause is not necessarily generated by Q.931 procedures but
may be generated by internal network timers.

BUT the response was immediate , so call you provider and have them check their routing to that number, and why they sent an inappropriate error.

3

The provider is SIPStation. They are semi hard to get in contact with, unless you pay for support credit.

I can call the problem number with my cell phone but not through FreePBX. So yeah, I think it is the SIP provider SIPStation.

4

Even AT&T FU sometimes :slight_smile: , but they fix it quickly when you tell them, (and never charge you for support :wink: )

5

Ugh AT&T lol. Thanks for the help!

6

We do not require support credits to get support with SIPStation. You can open a ticket under SIPStation for free or even just call us and get directed to the support department for SIPStation.

1 Like
7

Hi Tony. Thanks for the update. I just made a support ticket under sipstation.

I was under the incorrect assumption that support credit was needed (it is referenced in several places under sangoma regarding freepbx. Not sure of where the line is.)

Thanks!

8

Ok well it clearly states at support.sangoma.com main page that SIPStation support is free.

1 Like
9

Sounds great! Thanks for the update. Makes me feel better about selecting sipstation.

10

Just so people know. I started a ticket and it ended up being an issue of not having international calling. I was making a call to an Alaskan number that is considered international.

Unfortunately, freepbx did not provide an error that reflected the issue but it was my fault for not having the correct plan.

11

That is correct we state lower US 48 and Canada is part of your normal dialing.

12

Whatever Alaska is NOT international, nor is Hawaii
If you get cause 19, they are doingit wrong and they need to fix that . . .

2 Likes
13

Well it’s considered international for SIPStation because all lower US 48 and Canada is included in our trunks with no per minute charges for normal business usage. And to call areas that cost per minute money you have to enable our international calling option and agree to the per minute charges. Don’t tell me on how we run our business Dicko.

Secondly we don’t send a 486 if a user has international disabled. Asterisk may be doing something dumb but what we send is a very descriptive custom response of “480 INTERNATIONAL DISABLED” for this very reason.