Polycom VVX410 registers but cannot dial out

FreePBX Endpoints
1

I am using Free pbx with voip.ms

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I am asking for guidance from any Polycom/FreePBX users.

I have been using 2 Grandstream GXP 2130’s. They register fine, and dial out just fine…no issues…not so with the Polycom - it registers but cannot make outbound calls…

the phone is a VVX 410, with the latest firmware and it registers fine in Freepbx as ext 400. no errors etc. … but, when trying an outside call, I get error.“all circuits are busy now”

Looking through the logs, I see:

8887[2021-06-19 12:35:43] VERBOSE[11842][C-00000047] pbx.c: Executing [s@macro-dialout-trunk:35] NoOp(“PJSIP/400-0000008c”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34”) in new stack

I can provide more details, of course…it is so strange because the Grandstreams work just fine…and the Polycom seems to register, but cannot make outbound calls.

I may be missing something in the setup of the phone…although it registers just fine…

Any pointers appreciated…

2

Have you check digit map on WV410 is according to your needs?

3

I am using the default digit map … I can dial other phones inside …

4

Most likely one of three issues: destination number incorrectly formatted, caller ID incorrectly formatted, wrong trunk.

A few log lines before what you posted, you should see a line beginning with
app_dial.c: Called
which will show the selected trunk and number sent. Compare working and failing cases.

About 25 lines further back, look for
1?Set(CONNECTEDLINE(name,i)=CID:
which will show the caller ID sent.

5

So here is what I see with the app_dial: made call from ext 200 (working) and 400 (Polycom)… (I have edited the heck out of dozens of lines to find only relevant ones…) …

Dialed 800-333-3333 (Raddison Hotels) – the trunk name is Voip_ms_1

WORKING phone (Grandstream – ext 200) :

32300[2021-06-19 16:08:14] VERBOSE[13034][C-00000051] app_dial.c: Called PJSIP/8003333333@Voip_ms_1

32301[2021-06-19 16:08:15] VERBOSE[13034][C-00000051] app_dial.c: PJSIP/Voip_ms_1-0000009e answered PJSIP/200-0000009d

32302[2021-06-19 16:08:15] VERBOSE[13034][C-00000051] app_stack.c: PJSIP/Voip_ms_1-0000009e Internal Gosub(sub-send-obroute-email,s,1(8003333333,8003333333,1,1624118894,8024443899)) start

32303[2021-06-19 16:08:15] VERBOSE[13034][C-00000051] pbx.c: Executing [s@sub-send-obroute-email:1] GotoIf("PJSIP/Voip_ms_1-0000009e", "0?sendEmail") in new stack

32304[2021-06-19 16:08:15] VERBOSE[13034][C-00000051] pbx.c: Executing [s@sub-send-obroute-email:2] NoOp("PJSIP/Voip_ms_1-0000009e", "email notifications disabled…exiting.") in new stack

32305[2021-06-19 16:08:15] VERBOSE[13034][C-00000051] pbx.c: Executing [s@sub-send-obroute-email:3] Return("PJSIP/Voip_ms_1-0000009e", "") in new stack

32306[2021-06-19 16:08:15] VERBOSE[13034][C-00000051] app_stack.c: Spawn extension (from-pstn, , 1) exited non-zero on ‘PJSIP/Voip_ms_1-0000009e’

POLYCOM Ext 400 – dial the same 800 number:

32695[2021-06-19 16:25:22] VERBOSE[17046][C-00000053] app_dial.c: Called PJSIP/8003333333@Voip_ms_1

32696[2021-06-19 16:25:23] VERBOSE[17046][C-00000053] app_dial.c: Everyone is busy/congested at this time (1:0/1/0)

32697[2021-06-19 16:25:23] VERBOSE[17046][C-00000053] pbx.c: Executing [s@macro-dialout-trunk:35] NoOp("PJSIP/400-000000a1", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack

32698[2021-06-19 16:25:23] VERBOSE[17046][C-00000053] pbx.c: Executing [s@macro-dialout-trunk:36] GotoIf("PJSIP/400-000000a1", "0?continue,1:s-CONGESTION,1") in new stack

32699[2021-06-19 16:25:23] VERBOSE[17046][C-00000053] pbx_builtins.c: Goto (macro-dialout-trunk,s-CONGESTION,1)

32700[2021-06-19 16:25:23] VERBOSE[17046][C-00000053] pbx.c: Executing [s-CONGESTION@macro-dialout-trunk:1] Set("PJSIP/400-000000a1", "RC=34") in new stack

32701[2021-06-19 16:25:23] VERBOSE[17046][C-00000053] pbx.c: Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("PJSIP/400-000000a1", "34,1") in new stack

32702[2021-06-19 16:25:23] VERBOSE[17046][C-00000053] pbx_builtins.c: Goto (macro-dialout-trunk,34,1)

6

Look about 25 lines before the Called entry to see the caller ID sent.

If that looks correct, you might temporarily change Outbound CID for ext. 400 to match what you have for ext. 200. Possibly, VoIP.ms is rejecting the CID you are sending, even though it is properly formatted.

7

Very interesting - extension 200 had nothing entered in CID field.

Ext 400 had “400” in CID, which I removed.

Rebooted phone and did a core restart now on FPBX

I Can now dial out ! — I guess FreePBX doesn’t require the outbound CID to be populated…

I cannot thank you enough…this nagging issue has been resolved - and quite well ! (And to think I was going to send the phone back (!)

Thank you all once again !
Bill Clark, Windham , VT

and, fwiw - here is now what I see in the log for ext 400:

36257[2021-06-19 17:15:49] VERBOSE[29454][C-00000002] app_dial.c: Called PJSIP/8003333333@Voip_ms_1

36258[2021-06-19 17:15:50] VERBOSE[29454][C-00000002] app_dial.c: PJSIP/Voip_ms_1-00000003 answered PJSIP/400-00000002

36259[2021-06-19 17:15:50] VERBOSE[29454][C-00000002] app_stack.c: PJSIP/Voip_ms_1-00000003 Internal Gosub(sub-send-obroute-email,s,1(8003333333,8003333333,1,1624122949,8024443899)) start

36260[2021-06-19 17:15:50] VERBOSE[29454][C-00000002] pbx.c: Executing [s@sub-send-obroute-email:1] GotoIf(“PJSIP/Voip_ms_1-00000003”, “0?sendEmail”) in new stack

36261[2021-06-19 17:15:50] VERBOSE[29454][C-00000002] pbx.c: Executing [s@sub-send-obroute-email:2] NoOp(“PJSIP/Voip_ms_1-00000003”, “email notifications disabled…exiting.”) in new stack

36262[2021-06-19 17:15:50] VERBOSE[29454][C-00000002] pbx.c: Executing [s@sub-send-obroute-email:3] Return(“PJSIP/Voip_ms_1-00000003”, “”) in new stack

36263[2021-06-19 17:15:50] VERBOSE[29454][C-00000002] app_stack.c: Spawn extension (from-pstn, , 1) exited non-zero on ‘PJSIP/Voip_ms_1-00000003’

36264[2021-06-19 17:15:50] VERBOSE[29454][C-00000002] app_stack.c: PJSIP/Voip_ms_1-00000003 Internal Gosub(sub-send-obroute-email,s,1(8003333333,8003333333,1,1624122949,8024443899)) complete GOSUB_RETVAL=

36265[2021-06-19 17:15:50] VERBOSE[29469][C-00000002] bridge_channel.c: Channel PJSIP/Voip_ms_1-00000003 joined ‘simple_bridge’ basic-bridge <602d7c4c-0dc2-49dd-b0bd-8e62c8da82a4>

36266[2021-06-19 17:15:50] VERBOSE[29454][C-00000002] bridge_channel.c: Channel PJSIP/400-00000002 joined ‘simple_bridge’ basic-bridge <602d7c4c-0dc2-49dd-b0bd-8e62c8da82a4>

36267[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] bridge_channel.c: Channel PJSIP/400-00000002 left ‘simple_bridge’ basic-bridge <602d7c4c-0dc2-49dd-b0bd-8e62c8da82a4>

36268[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] app_macro.c: Spawn extension (macro-dialout-trunk, s, 34) exited non-zero on ‘PJSIP/400-00000002’ in macro ‘dialout-trunk’

36269[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Spawn extension (from-internal, 8003333333, 12) exited non-zero on ‘PJSIP/400-00000002’

36270[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Executing [h@from-internal:1] Macro(“PJSIP/400-00000002”, “hangupcall”) in new stack

36271[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/400-00000002”, “1?theend”) in new stack

36272[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx_builtins.c: Goto (macro-hangupcall,s,3)

36273[2021-06-19 17:15:55] VERBOSE[29469][C-00000002] bridge_channel.c: Channel PJSIP/Voip_ms_1-00000003 left ‘simple_bridge’ basic-bridge <602d7c4c-0dc2-49dd-b0bd-8e62c8da82a4>

36274[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/400-00000002”, “0?Set(CDR(recordingfile)=)”) in new stack

36275[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/400-00000002”, "PJSIP/Voip_ms_1-00000003 montior file= ") in new stack

36276[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/400-00000002”, “1?skipagi”) in new stack

36277[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx_builtins.c: Goto (macro-hangupcall,s,7)

36278[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Executing [s@macro-hangupcall:7] Hangup(“PJSIP/400-00000002”, “”) in new stack

36279[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/400-00000002’ in macro ‘hangupcall’

36280[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/400-00000002’

36281[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] app_stack.c: PJSIP/400-00000002 Internal Gosub(crm-hangup,s,1) start

36282[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Executing [s@crm-hangup:1] NoOp(“PJSIP/400-00000002”, “Sending Hangup to CRM”) in new stack

36283[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Executing [s@crm-hangup:2] NoOp(“PJSIP/400-00000002”, “HANGUP CAUSE: 16”) in new stack

36284[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Executing [s@crm-hangup:3] ExecIf(“PJSIP/400-00000002”, “0?Set(__CRM_VOICEMAIL=)”) in new stack

36285[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Executing [s@crm-hangup:4] NoOp(“PJSIP/400-00000002”, “MASTER CHANNEL: 1624122949.2 = 1624122949.2”) in new stack

36286[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Executing [s@crm-hangup:5] GotoIf(“PJSIP/400-00000002”, “0?return”) in new stack

36287[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Executing [s@crm-hangup:6] Set(“PJSIP/400-00000002”, “__CRM_HANGUP=1”) in new stack

36288[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Executing [s@crm-hangup:7] AGI(“PJSIP/400-00000002”, “agi://127.0.0.1/sangomacrm.agi”) in new stack

36289[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] res_agi.c: <PJSIP/400-00000002>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0

36290[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] pbx.c: Executing [s@crm-hangup:8] Return(“PJSIP/400-00000002”, “”) in new stack

36291[2021-06-19 17:15:55] VERBOSE[29454][C-00000002] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/400-00000002’

8

FreePBX will use the CallerID that’s set at the lowest level, so if there’s no CID set at the extension, the system will use the one for the Outbound Route. If there is no Outbound Route CID and no extension CID, it will use the Trunk CID. All of these can be “inverted” by using the “Override” flag.

The idea is that, if you want to set up individual DIDs for your extensions (or groups of phones, let’s say), you can set the phone CID to something specific for that group of phones and “default” everyone else to the Trunk or Outbound Route, thereby maximizing your flexibility.

9

Very good to know – I checked my settings, and the CID is set at the Trunk and Outbould Route level… they are the same (the voip.ms DID ) … is it OK to have values at both levels (trunk and route) as long as they are the same ??

10

They can be the same or different. For most installations (a single DID and single Trunk) that would be the most common setup.

There are plenty of cases where setting them to different things can accomplish all sorts of “tricky but cool” effects. For example, let’s say you have different branches in different states and want the CID to reflect that when you call someone in Kansas that the CID reflects your Kansas office, but anyone anywhere else should get the “800” number Caller ID. You could set up a Trunk CID to the 800 number, and set up an outbound route for calls that go to Kansas to get the Kansas caller ID. This is what I was talking about when I referred to flexibility. I use this feature for calls I originate that go to local numbers (they get my local number) but everyone else gets my 800 number (since they’re not calling locally and therefore would have to pay to call my local number).

There are lots of ways to make this work and can, if you get really ahead of yourself, cause no limit to the kinds of grief you can cause yourself. In your situation, I suspect that setting the Trunk CID is probably plenty - you could probably leave the Outbound Route and Extension CIDs blank, and just go with the one CID. As you find new reasons to change the CID (with the blessing of your ITSP) you can do that.

11

Many thanks - Obviously the tip of the iceberg here … but one step at a time … trying to get a basic home voip system going on a solid foundation. So glad to have squared the circle on the CID issue. THX again.

12

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