FreePBX External Calls to Internal Extensions Cannot Hear nor Speak

ISSUE: Cannot hear when dialing direct extensions from external phone source.

I have noticed many similar support requests and searched the web extensively but have not identified a potential solution to this particular scenario. I hope to have provided enough info for my request. This seems to be a firewall issue, but after careful examination, I still have not identified the exact problem especially since some external calls work and others do not.

WHAT WORKS:

  1. Internal extension to extension.
    End-user answers call and speaks/hears successfully.
  2. External to IVR selection. Example, 1 for Customer Service, 2 for Maintenance, 3 for Billing.
    End-user answers call and speaks/hears successfully.
  3. Dialing out.
    End-user makes call and speaks/hears successfully.

WHAT DOES NOT WORK:

  1. External to direct dial extension.
    End-user answers call but cannot speak/hear.
  2. External to IVR selection Ring Group, then forwarded to extension if Destination if no answer.
    End-user answers call but cannot speak/hear.

To confirm existing issue, I modified Inbound Routes “from” Set Destination–>Time Conditions–>IVR “to” Set Destination–>Extensions–>user’s extension. Thus, an external call direct to extension (bypassing IVR) still has the same result… End-user answers call but cannot speak/hear.

CONFIGURATION:
Router/firewall – Public Static IP
Forwarded/NAT ports to FreePBX server: 5061 TCP/UDP (PJSIP TLS), 10000-20000 UDP (RTP for SIP)

FreePBX 14.0.5.25 Asterisk 13.22.0, Installed using full automation, fully updated at this posting
Inbound DID - Any, CID - Any, Destination - Time Conditions/IVR/Direct Dial enabled
All extensions using PJSIP
SIP trunks provided by SIPstation (quantity 3)
FreePBX firewall enabled using wizard

Asterisk Log Snippet (tail -f full)
[2019-03-16 11:41:37] VERBOSE[25798] res_pjsip/pjsip_options.c: Contact 100/sip: [email protected]:38892;transport=TLS;rinstance=743cbcddbac10b95 has been deleted
[2019-03-16 11:41:42] VERBOSE[20032] res_pjsip/pjsip_options.c: Contact 100/sip:[email protected]:38892;transport=TLS;rinstance=743cbcddbac10b95 is now Reachable. RTT: 511.265 msec
[2019-03-16 11:43:39] VERBOSE[20032] res_pjsip/pjsip_options.c: Contact 100/sip:[email protected]:38892;transport=TLS;rinstance=743cbcddbac10b95 has been deleted
[2019-03-16 11:43:39] VERBOSE[5691] res_pjsip_registrar.c: Added contact ‘sip:[email protected]:38892;transport=TLS;rinstance=743cbcddbac10b95’ to AOR ‘100’ with expiration of 60 seconds

UPDATES:
After deleting the existing inbound route and creating a new inbound route, my previous confirmation test Set Destination–>Extensions–>user’s extension which originally failed, now works. End-user makes call and speaks/hears successfully.

Attempting the same with the IVR by deleting and creating a new IVR, Set Destination–>Extensions–>IVR still fails. End-user answers call but cannot speak/hear. Possible bug with IVR --> Extension ???

I built a new FreePBX server using the latest ISO download, updated all modules, and applied all system updates. Same behaviour as listed below. Seems to have issues with IVR transferring to Extensions. Confirmed behaviour on both server installs.

After additional attempts we now have:

WHAT WORKS:

  1. Internal extension to extension.
    End-user answers call and speaks/hears successfully.
  2. External to IVR selection. Example, 1 for Customer Service, 2 for Maintenance, 3 for Billing.
    End-user answers call and speaks/hears successfully.
  3. Dialing out.
    End-user makes call and speaks/hears successfully.
  4. External to direct dial extension.
    End-user makes call and speaks/hears successfully.

WHAT DOES NOT WORK:

  1. External to IVR selection–>Extension or IVR selection–>Ring Group–>Extension on No Answer.
    End-user answers call but cannot speak/hear.

Are you using a valid certificate?

Can you post a full call log?
https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs-PartII

Thanks for replying PitzKey. Currently I am using the default internal certificate.

grep 1552843838.1112 /var/log/asterisk/full*
/var/log/asterisk/full:[2019-03-17 13:30:38] VERBOSE[25899][C-00000424] pbx.c: Executing [[email protected]:22] Set(“SIP/fpbx-1-prem-A6T1AG5QP6aq-00000022”, “__CRM_LINKEDID=1552843838.1112”) in new stack
/var/log/asterisk/full:[2019-03-17 13:30:48] VERBOSE[25899][C-00000424] pbx.c: Executing [[email protected]:1] Set(“SIP/fpbx-1-prem-A6T1AG5QP6aq-00000022”, “TOUCH_MONITOR=1552843838.1112”) in new stack
/var/log/asterisk/full:[2019-03-17 13:30:53] VERBOSE[25899][C-00000424] pbx.c: Executing [[email protected]:4] NoOp(“PJSIP/200-00000439”, “MASTER CHANNEL: 1552843848.1116 = 1552843838.1112”) in new stack
/var/log/asterisk/full:[2019-03-17 13:31:02] VERBOSE[26044][C-00000424] pbx.c: Executing [[email protected]:4] NoOp(“PJSIP/200-00000438”, “MASTER CHANNEL: 1552843848.1115 = 1552843838.1112”) in new stack
/var/log/asterisk/full:[2019-03-17 13:31:02] VERBOSE[25899][C-00000424] pbx.c: Executing [[email protected]:4] NoOp(“SIP/fpbx-1-prem-A6T1AG5QP6aq-00000022”, “MASTER CHANNEL: 1552843838.1112 = 1552843838.1112”) in new stack

grep C-00000424 /var/log/asterisk/full-20190317
No data return.

grep C-00000424 /var/log/asterisk/full-20190317 | pastebin
title: Automatic Pastebin from Sangoma OS 7
name: [email protected]
private: 1
expire: 0
https://pastebin.freepbx.org/view/180dd47b

ls -l /tmp | grep core.
No data return.

This is empty, because there’s no full-20190317 file.

please run

grep C-00000424 /var/log/asterisk/full | pastebin

and post the new pastebin link please.

That might be an issue, try installing a Let’s Encrypt Cert, it’s free.

grep C-00000424 /var/log/asterisk/full | pastebin
title: Automatic Pastebin from Sangoma OS 7
name: [email protected]
private: 1
expire: 0
https://pastebin.freepbx.org/view/0c2cb871

@wingutechnology OK you’re going to have to do this again because this doesn’t show anything but a successful call because that’s what it was. Looking at the full log or verbose output from the Asterisk console for a no-audio or one-way audio issue is pointless. It doesn’t contain any real information about the RTP/media.

Do the test again but this time just do pjsip set logger on from the Asterisk cli so that the actual SIP packets are outputted and we can see what is happening with the SDP/RTP/media of the call.

You will have to scroll back and copy the information this will not get live output to a file for pastebin.

I do see the below.

[2019-03-17 13:30:53] VERBOSE[26044][C-00000424] bridge_channel.c: Channel PJSIP/200-00000438 joined 'simple_bridge' basic-bridge <ba0b62b4-64e1-421d-96f6-193ee28777f2&gt
[2019-03-17 13:30:53] VERBOSE[25899][C-00000424] bridge_channel.c: Channel SIP/fpbx-1-prem-A6T1AG5QP6aq-00000022 joined 'simple_bridge' basic-bridge <ba0b62b4-64e1-421d-96f6-193ee28777f2> 
[2019-03-17 13:30:54] VERBOSE[25899][C-00000424] res_srtp.c: SRTCP unprotect failed because of unable to perform desired validation 
[2019-03-17 13:30:56] VERBOSE[25899][C-00000424] res_srtp.c: SRTCP unprotect failed because of unable to perform desired validation 
[2019-03-17 13:31:01] VERBOSE[25899][C-00000424] res_srtp.c: SRTCP unprotect failed because of unable to perform desired validation 
[2019-03-17 13:31:02] VERBOSE[26044][C-00000424] bridge_channel.c: Channel PJSIP/200-00000438 left 'simple_bridge' basic-bridge <ba0b62b4-64e1-421d-96f6-193ee28777f2>

Which as far as I understand, if it can’t negotiate RTP then there’s obviously no audio…

Question to OP, which version of Asterisk are you using, and which phones do you have?

I expressly said that nothing will output the logger data to a pastbin file. You have to get the scrollback on your screen and copy and paste that. I do not need to see verbose output of the dialplan full log. I just need to see the PJSIP logger.

asterisk -r
pjsip set logger on

That’s all you need to do.

@BlazeStudios

<— Received SIP request (733 bytes) from UDP:195.154.223.231:53953 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 195.154.223.231:53953;branch=z9hG4bK1453830751
Max-Forwards: 70
From: sip:[email protected]:5060;tag=1659369554
To: sip:[email protected]:5060
Call-ID: 1891222548-2142915723-141057087
CSeq: 1 INVITE
Contact: sip:[email protected]:53953
Content-Type: application/sdp
Content-Length: 205
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH
User-Agent: fgfdhgfxjfhyjhkj

v=0
o=1 16264 18299 IN IP4 192.168.1.83
s=call
c=IN IP4 192.168.1.83
t=0 0
m=audio 25282 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<— Transmitting SIP response (318 bytes) to UDP:195.154.223.231:53953 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.154.223.231:53953;rport=53953;received=195.154.223.231;branch=z9hG4bK1453830751
Call-ID: 1891222548-2142915723-141057087
From: sip:[email protected];tag=1659369554
To: sip:[email protected]
CSeq: 1 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Content-Length: 0

[2019-03-17 18:33:19] WARNING[19578][C-000000f7]: func_channel.c:460 func_channel_read: Unknown or unavailable item requested: ‘recvip’
[2019-03-17 18:33:19] WARNING[19578][C-000000f7]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from "
<— Transmitting SIP response (815 bytes) to UDP:195.154.223.231:53953 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.154.223.231:53953;rport=53953;received=195.154.223.231;branch=z9hG4bK1453830751
Call-ID: 1891222548-2142915723-141057087
From: sip:[email protected];tag=1659369554
To: sip:[email protected];tag=468f7296-1019-4778-ad90-c9cc93c4be31
CSeq: 1 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Contact: sip:47.206.106.125:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 231

v=0
o=- 16264 18301 IN IP4 47.206.106.125
s=Asterisk
c=IN IP4 47.206.106.125
t=0 0
m=audio 17474 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

Well that’s the start of a spam call it looks like. This does not have anything for your call. You are going to need to increase the scroll buffer and you need to get the entire thing from the moment you enter the Asterisk cli and make the test call.

@BlazeStudios It’s a lot of data; constantly out-putting. Standby for another test. What you saw earlier was a call from my FreePBX system to the FreePBX we are investigating. The data I will capture next will be from a cell phone line to the FreePBX system we are investigating.

There’s more data after this. I’ll send more as required. Test using cell phone --> FreePBX system.

CLI> pjsip set logger on
PJSIP Logging enabled
<— Transmitting SIP response (815 bytes) to UDP:93.115.27.120:60332 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 93.115.27.120:60332;rport=60332;received=93.115.27.120;branch=z9hG4bK394986264
Call-ID: 1198962395-669663186-1079348521
From: sip:[email protected];tag=817196450
To: sip:[email protected];tag=830b6d15-6063-4bae-b07b-7d3cab1cbb99
CSeq: 1 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Contact: sip:47.206.106.125:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 231

v=0
o=- 16264 18301 IN IP4 47.206.106.125
s=Asterisk
c=IN IP4 47.206.106.125
t=0 0
m=audio 12242 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP response (816 bytes) to UDP:195.154.223.231:49452 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.154.223.231:49452;rport=49452;received=195.154.223.231;branch=z9hG4bK606925477
Call-ID: 1865262278-1805069610-583942032
From: sip:[email protected];tag=1221085365
To: sip:[email protected];tag=4f9c155f-e30e-4d47-9a3a-6354414c0699
CSeq: 1 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Contact: sip:47.206.106.125:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 231

v=0
o=- 16264 18301 IN IP4 47.206.106.125
s=Asterisk
c=IN IP4 47.206.106.125
t=0 0
m=audio 17378 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP request (401 bytes) to UDP:195.154.223.231:62759 —>
BYE sip:[email protected]:62759 SIP/2.0
Via: SIP/2.0/UDP 47.206.106.125:5060;rport;branch=z9hG4bKPjec298820-cc81-4a7c-b687-f3e869af5f21
From: sip:[email protected];tag=3ce2dbd0-f53e-4c97-b0f2-d4bbaa844d02
To: sip:[email protected];tag=1053488161
Call-ID: 822639560-1998847788-1958183435
CSeq: 8118 BYE
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Length: 0

<— Transmitting SIP response (814 bytes) to UDP:195.154.223.231:51021 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.154.223.231:51021;rport=51021;received=195.154.223.231;branch=z9hG4bK1651512326
Call-ID: 1170806582-290033941-986810380
From: sip:[email protected];tag=16081236
To: sip:[email protected];tag=ed5865f1-3514-41f6-be88-2d614f3bfcac
CSeq: 1 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Contact: sip:47.206.106.125:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 231

v=0
o=- 16264 18301 IN IP4 47.206.106.125
s=Asterisk
c=IN IP4 47.206.106.125
t=0 0
m=audio 19290 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP request (406 bytes) to UDP:62.210.90.247:50081 —>
BYE sip:[email protected]:50081 SIP/2.0
Via: SIP/2.0/UDP 47.206.106.125:5060;rport;branch=z9hG4bKPj2041bf6a-9a9a-480f-b91f-374ad93adf7d
From: sip:[email protected];tag=dd0d3eb6-05e6-4b37-a921-cc69b8ad5d59
To: sip:[email protected];tag=1930393139
Call-ID: 131445152-1175606891-273778049
CSeq: 13588 BYE
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Length: 0

<— Transmitting SIP request (400 bytes) to UDP:195.154.223.231:56670 —>
BYE sip:[email protected]:56670 SIP/2.0
Via: SIP/2.0/UDP 47.206.106.125:5060;rport;branch=z9hG4bKPj3e14ae96-253a-48bc-a658-45cd88a4eb82
From: sip:[email protected];tag=c43095d4-d40d-4019-8643-755ed461c826
To: sip:[email protected];tag=2101133002
Call-ID: 115930455-2028654004-1358858705
CSeq: 705 BYE
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Length: 0

<— Transmitting SIP response (817 bytes) to UDP:195.154.223.231:58616 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.154.223.231:58616;rport=58616;received=195.154.223.231;branch=z9hG4bK1185270837
Call-ID: 1505882382-827110192-1817155072
From: sip:[email protected];tag=1343353790
To: sip:[email protected];tag=efec7090-eda6-45c1-ae85-2dc79d168d7d
CSeq: 1 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Contact: sip:47.206.106.125:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 231

v=0
o=- 16264 18301 IN IP4 47.206.106.125
s=Asterisk
c=IN IP4 47.206.106.125
t=0 0
m=audio 16916 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP response (814 bytes) to UDP:195.154.223.231:50445 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.154.223.231:50445;rport=50445;received=195.154.223.231;branch=z9hG4bK206903395
Call-ID: 622137454-241971564-1737294054
From: sip:[email protected];tag=789752356
To: sip:[email protected];tag=8506067c-6b1c-4417-b053-efbc8170f09c
CSeq: 1 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Contact: sip:47.206.106.125:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 231

v=0
o=- 16264 18301 IN IP4 47.206.106.125
s=Asterisk
c=IN IP4 47.206.106.125
t=0 0
m=audio 11398 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP request (402 bytes) to UDP:195.154.223.231:50274 —>
BYE sip:[email protected]:50274 SIP/2.0
Via: SIP/2.0/UDP 47.206.106.125:5060;rport;branch=z9hG4bKPjcdc35fe1-66e7-48bb-bc0d-2cc6fe2a39eb
From: sip:[email protected];tag=c9f9c516-c8bc-413f-8127-a131b7dcd740
To: sip:[email protected];tag=2067909547
Call-ID: 1224690059-1132133650-706465965
CSeq: 12752 BYE
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Length: 0

<— Transmitting SIP request (400 bytes) to UDP:195.154.223.231:62285 —>
BYE sip:[email protected]:62285 SIP/2.0
Via: SIP/2.0/UDP 47.206.106.125:5060;rport;branch=z9hG4bKPj1c34d2e2-55f4-4ac2-8073-82f383c43cb0
From: sip:[email protected];tag=3204b9fd-b1a0-4307-b09b-53f11244f72e
To: sip:[email protected];tag=981241970
Call-ID: 1097615345-1506639358-501814916
CSeq: 7036 BYE
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Length: 0

<— Transmitting SIP response (815 bytes) to UDP:93.115.27.120:60332 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 93.115.27.120:60332;rport=60332;received=93.115.27.120;branch=z9hG4bK394986264
Call-ID: 1198962395-669663186-1079348521
From: sip:[email protected];tag=817196450
To: sip:[email protected];tag=830b6d15-6063-4bae-b07b-7d3cab1cbb99
CSeq: 1 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Contact: sip:47.206.106.125:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 231

v=0
o=- 16264 18301 IN IP4 47.206.106.125
s=Asterisk
c=IN IP4 47.206.106.125
t=0 0
m=audio 12242 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP response (816 bytes) to UDP:195.154.223.231:49452 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.154.223.231:49452;rport=49452;received=195.154.223.231;branch=z9hG4bK606925477
Call-ID: 1865262278-1805069610-583942032
From: sip:[email protected];tag=1221085365
To: sip:[email protected];tag=4f9c155f-e30e-4d47-9a3a-6354414c0699
CSeq: 1 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Contact: sip:47.206.106.125:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 231

v=0
o=- 16264 18301 IN IP4 47.206.106.125
s=Asterisk
c=IN IP4 47.206.106.125
t=0 0
m=audio 17378 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— Transmitting SIP request (402 bytes) to UDP:195.154.223.231:49452 —>
BYE sip:[email protected]:49452 SIP/2.0
Via: SIP/2.0/UDP 47.206.106.125:5060;rport;branch=z9hG4bKPj89c11a12-4b00-4ecb-89ca-183ca4b9894c
From: sip:[email protected];tag=4f9c155f-e30e-4d47-9a3a-6354414c0699
To: sip:[email protected];tag=1221085365
Call-ID: 1865262278-1805069610-583942032
CSeq: 32004 BYE
Max-Forwards: 70
User-Agent: FPBX-14.0.5.25(13.22.0)
Content-Length: 0

<— Received SIP request (736 bytes) from UDP:195.154.223.231:64252 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 195.154.223.231:64252;branch=z9hG4bK970357785
Max-Forwards: 70
From: sip:[email protected]:5060;tag=80126006
To: sip:[email protected]:5060
Call-ID: 2086447141-161414568-1937026076
CSeq: 1 INVITE
Contact: sip:[email protected]:64252
Content-Type: application/sdp
Content-Length: 207
Allow: ACK, BYE, CANCEL, INFO, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER, SUBSCRIBE, UPDATE, PUBLISH
User-Agent: fgfdhgfxjfhyjhkj

v=0
o=132 16264 18299 IN IP4 192.168.1.83
s=call
c=IN IP4 192.168.1.83
t=0 0
m=audio 25282 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

<— Transmitting SIP response (317 bytes) to UDP:195.154.223.231:64252 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 195.154.223.231:64252;rport=64252;received=195.154.223.231;branch=z9hG4bK970357785
Call-ID: 2086447141-161414568-1937026076
From: sip:[email protected];tag=80126006
To: sip:[email protected]
CSeq: 1 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Content-Length: 0

[2019-03-17 18:50:38] WARNING[22616][C-0000019c]: func_channel.c:460 func_channel_read: Unknown or unavailable item requested: ‘recvip’
[2019-03-17 18:50:38] WARNING[22616][C-0000019c]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from "
<— Transmitting SIP response (814 bytes) to UDP:195.154.223.231:64252 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 195.154.223.231:64252;rport=64252;received=195.154.223.231;branch=z9hG4bK970357785
Call-ID: 2086447141-161414568-1937026076
From: sip:[email protected];tag=80126006
To: sip:[email protected];tag=7da2904a-59b1-4976-b230-0338dcd9eca4
CSeq: 1 INVITE
Server: FPBX-14.0.5.25(13.22.0)
Contact: sip:47.206.106.125:5060
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 231

v=0
o=- 16264 18301 IN IP4 47.206.106.125
s=Asterisk
c=IN IP4 47.206.106.125
t=0 0
m=audio 15522 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[2019-03-17 18:50:39] ERROR[22612][C-0000019a]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-03-17 18:50:39] ERROR[22612][C-0000019a]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-03-17 18:50:39] ERROR[22612][C-0000019a]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-03-17 18:50:39] ERROR[22612][C-0000019a]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-03-17 18:50:39] ERROR[22612][C-0000019a]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-03-17 18:50:39] ERROR[22612][C-0000019a]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-03-17 18:50:39] ERROR[22612][C-0000019a]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-03-17 18:50:39] ERROR[22612][C-0000019a]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-03-17 18:50:39] ERROR[22612][C-0000019a]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-03-17 18:50:39] ERROR[22612][C-0000019a]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-03-17 18:50:39] ERROR[22612][C-0000019a]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.
[2019-03-17 18:50:39] ERROR[22612][C-0000019a]: res_pjsip_header_funcs.c:454 func_read_header: This function requires a PJSIP channel.

As asked earlier,

  1. Which version of Asterisk are you using?
  2. Which phones do you have?
  3. Can you try installing a LE cert?

@PitzKey

  1. Which version of Asterisk are you using?
    FreePBX 14.0.5.25 Asterisk 13.22.0 , Installed using full automation, fully updated at this posting
  2. Which phones do you have?
    GrandStream GXP2130’s and Yealink T41P’s in this particular config.
  3. Can you try installing a LE cert?
    Using only one public IP and already assigned to a reverse proxy. In other setups of similar configuration, using the local certificate has been sufficient to provide a fully working system.

On 2 of the grandstream 2130s can you set the SIP Transport to TCP/TLS and then try to make a call in between them to see what happens

… just be sure you’ve set up all of the required ports/forwards/certificates/etc. for this to work. You may also need to adjust the firewall(s) in your network to allow the additional TCP traffic to be passed.

1 Like

Does it happen with the Yealink phone’s as well?

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