@ccvit wrote:

I've done my first ever installation of FreePBX and I've setup the server and around 10 phones. The extensions are setup as PJSIP phones. Phones are on same vlan as server. I can make calls between phones and I have a paging group setup with all the phones in the group. When I try to page, it just rings at all stations. It does not allow me to announce something over all the phones at once. I thought it was due to not having the airport style paging so I bought the Page Pro module. This did not give me the ability to do this either. What am I not setting up? I saw another post about using *54 to turn on Intercom and I tried that as well and still did not work.

I'm on FreePBX 10.13.66-18. Thank you so much for your time.

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@hazel wrote:

[2017-06-13 04:41:50] WARNING[2982][C-00000004]: res_rtp_asterisk.c:773 ast_rtp_ice_start: No RTCP candidates; skipping ICE checklist (0x7f6db831fbe8)
-- Channel SIP/100-0000000a joined 'simple_bridge' basic-bridge <745e6ab1-d793-4894-bf06-823a0a9a78bc>
-- Channel SIP/600-00000009 joined 'simple_bridge' basic-bridge <745e6ab1-d793-4894-bf06-823a0a9a78bc>
[2017-06-13 04:42:21] NOTICE[1766]: chan_sip.c:29402 check_rtp_timeout: Disconnecting call 'SIP/600-00000009' for lack of RTP activity in 31 seconds
-- Channel SIP/600-00000009 left 'simple_bridge' basic-bridge <745e6ab1-d793-4894-bf06-823a0a9a78bc>
== Spawn extension (macro-dial-one, s, 51) exited non-zero on 'SIP/600-00000009' in macro 'dial-one'
== Spawn extension (macro-exten-vm, s, 16) exited non-zero on 'SIP/600-00000009' in macro 'exten-vm'
== Spawn extension (ext-local, 100, 2) exited non-zero on 'SIP/600-00000009'
-- Executing [h@ext-local:1] Macro("SIP/600-00000009", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/600-00000009", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/600-00000009", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("SIP/600-00000009", "") in new stack

If I call webRTC "res_rtp_asterisk.c:773 ast_rtp_ice_start: No RTCP candidates; skipping ICE checklist (0x7f6db831fbe8)" warning appeared and hangup . What can I do? I used freepbx version 13 and asterisk 13.12.Please help me.

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@munozj wrote:

I know an SBC would be the BEST solution but until the client has the budge, is it possible to interconnect multiple sites each using a different SIP port.
i.e.
main site SIP 5060
remote site 1 SIP 5062
remote site 2 SIP 5068

PJSIP is available but also on non-standard ports. Can you specify the port to use in the SIP-TRUNK settings page?

The main site has various ITSP's all talking on 5060 so they don't want to change the main and don't want to use standard ports on all the cloud PBX's serving the remote users.

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@Pageplayer wrote:

Although I have a little experience setting up cisco equipment in a network, I am a complete noob with this voip thing.
Here's my issue: About a year ago an associate of mine installed a IP phone system, using Freepbx on a Raspberry pi to handle only the voicemail. (SIP to voicemail, analog parts to pstn)

The Rasp has corrupted itself twice since then, and this last time I wasn't even able to login to the CLI. So I downloaded the recent stable version and installed it on a Dell R210 server.

I managed to get everything working again myself, but my only issue is; when a message is left on the extension I setup for the person delegated to listen to them, The MWI indicator does not show up on the Cisco phone.
Chatting with the original associate that set it up, he says I need the script in /etc/scripts that was on the Raspberry. But as mentioned, the raspberry is dead.

Is there anyone out there with the patience to help me out with this?

Chuck

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@jessy5765 wrote:

We have a client who has an IVR at the beginning of the route that has Direct Dial enabled. They press 1 to access the directory.

1. If you dial 4011 using Direct Dial it works just fine and rings through to her.
2. If you dial 1 to access the directory and then dial 4011 it goes straight to "Dial Up Modem sounds"

This is the only extension that does this and from what I see has no special configurations. This is a new phone system and all extensions are configured the same way.

I have removed her from the directory and re-added her. No luck

Help!

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@threeeye wrote:

Hi guys,
I'm trying to connect my 2 FreePBX boxes together.
Both are FreePBX 14.0.1rc1.17
On server1, I have extensions 2xxx
On server2, I have extensions 4xxx
External calls go only to server1 (server2 cannot receive external calls, only from internal on system1 (extensions 2xxx))
Server2 cannot make any outgoing calls, not even to server1

I setup the IAX2 Trunks on both servers, with Outbound Route on server1, but when I call from server1 to any 4xxx extension, I get "all circuits busy now"
Here is the CLI output:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [4701@from-internal:1] Macro("SIP/1010-00000008", "user-callerid,LIMIT") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/1010-00000008", "TOUCH_MONITOR=1497453103.15") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/1010-00000008", "AMPUSER=1010") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/1010-00000008", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/1010-00000008", "1?Set(__REALCALLERIDNUM=1010)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/1010-00000008", "AMPUSER=1010") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/1010-00000008", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/1010-00000008", "AMPUSERCIDNAME=1010") in new stack
-- Executing [s@macro-user-callerid:8] GotoIf("SIP/1010-00000008", "0?report") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/1010-00000008", "AMPUSERCID=1010") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/1010-00000008", "_DIALOPTIONS=Ttr") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/1010-00000008", "CALLERID(all)="1010" <1010>") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/1010-00000008", "0?limit") in new stack
-- Executing [s@macro-user-callerid:13] ExecIf("SIP/1010-00000008", "1?Set(GROUP(concurrency_limit)=1010)") in new stack
-- Executing [s@macro-user-callerid:14] ExecIf("SIP/1010-00000008", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:15] GotoIf("SIP/1010-00000008", "1?continue") in new stack
-- Goto (macro-user-callerid,s,29)
-- Executing [s@macro-user-callerid:29] Set("SIP/1010-00000008", "CALLERID(number)=1010") in new stack
-- Executing [s@macro-user-callerid:30] Set("SIP/1010-00000008", "CALLERID(name)=1010") in new stack
-- Executing [s@macro-user-callerid:31] GotoIf("SIP/1010-00000008", "0?cnum") in new stack
-- Executing [s@macro-user-callerid:32] Set("SIP/1010-00000008", "CDR(cnam)=1010") in new stack
-- Executing [s@macro-user-callerid:33] Set("SIP/1010-00000008", "CDR(cnum)=1010") in new stack
-- Executing [s@macro-user-callerid:34] Set("SIP/1010-00000008", "CHANNEL(language)=en") in new stack
-- Executing [s@macro-user-callerid:35] GosubIf("SIP/1010-00000008", "0?app-check-classofservce,s,1()") in new stack
-- Executing [4701@from-internal:2] Set("SIP/1010-00000008", "ROUTEUSER=1010") in new stack
-- Executing [4701@from-internal:3] GotoIf("SIP/1010-00000008", "1?notblind") in new stack
-- Goto (from-internal,4701,6)
-- Executing [4701@from-internal:6] GotoIf("SIP/1010-00000008", "1?restrictedroute-13cee27a2bd93915479f049378cffdd3,4701,2:outbound-allroutes,4701,2") in new stack
-- Goto (restrictedroute-13cee27a2bd93915479f049378cffdd3,4701,2)
-- Executing [4701@restrictedroute-13cee27a2bd93915479f049378cffdd3:2] Gosub("SIP/1010-00000008", "sub-record-check,s,1(out,4701,never)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/1010-00000008", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("SIP/1010-00000008", "_RECSTATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("SIP/1010-00000008", "NOW=1497453103") in new stack
-- Executing [s@sub-record-check:4] Set("SIP/1010-00000008", "__DAY=14") in new stack
-- Executing [s@sub-record-check:5] Set("SIP/1010-00000008", "__MONTH=06") in new stack
-- Executing [s@sub-record-check:6] Set("SIP/1010-00000008", "__YEAR=2017") in new stack
-- Executing [s@sub-record-check:7] Set("SIP/1010-00000008", "__TIMESTR=20170614-151143") in new stack
-- Executing [s@sub-record-check:8] Set("SIP/1010-00000008", "__FROMEXTEN=1010") in new stack
-- Executing [s@sub-record-check:9] Set("SIP/1010-00000008", "_MONFMT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("SIP/1010-00000008", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("SIP/1010-00000008", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("SIP/1010-00000008", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("SIP/1010-00000008", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("SIP/1010-00000008", "3?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("SIP/1010-00000008", "1?sub-record-check,out,1") in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] NoOp("SIP/1010-00000008", "Outbound Recording Check from 1010 to 4701") in new stack
-- Executing [out@sub-record-check:2] Set("SIP/1010-00000008", "RECMODE=dontcare") in new stack
-- Executing [out@sub-record-check:3] ExecIf("SIP/1010-00000008", "1?Goto(routewins)") in new stack
-- Goto (sub-record-check,out,7)
-- Executing [out@sub-record-check:7] Gosub("SIP/1010-00000008", "recordcheck,1(never,out,4701)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("SIP/1010-00000008", "Starting recording check against never") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("SIP/1010-00000008", "never") in new stack
-- Goto (sub-record-check,recordcheck,14)
-- Executing [recordcheck@sub-record-check:14] Set("SIP/1010-00000008", "_RECPOLICY_MODE=NEVER") in new stack
-- Executing [recordcheck@sub-record-check:15] Goto("SIP/1010-00000008", "stoprec") in new stack
-- Goto (sub-record-check,recordcheck,25)
-- Executing [recordcheck@sub-record-check:25] NoOp("SIP/1010-00000008", "Stopping recording: out, 4701") in new stack
-- Executing [recordcheck@sub-record-check:26] Set("SIP/1010-00000008", "_RECSTATUS=STOPPED") in new stack
-- Executing [recordcheck@sub-record-check:27] System("SIP/1010-00000008", "/var/lib/asterisk/bin/stoprecording.php "SIP/1010-00000008"") in new stack
-- Executing [recordcheck@sub-record-check:28] Return("SIP/1010-00000008", "") in new stack
-- Executing [out@sub-record-check:8] Return("SIP/1010-00000008", "") in new stack
-- Executing [4701@restrictedroute-13cee27a2bd93915479f049378cffdd3:3] ExecIf("SIP/1010-00000008", "0 ?Set(CDR(accountcode)=)") in new stack
-- Executing [4701@restrictedroute-13cee27a2bd93915479f049378cffdd3:4] Set("SIP/1010-00000008", "INTRACOMPANYROUTE=YES") in new stack
-- Executing [4701@restrictedroute-13cee27a2bd93915479f049378cffdd3:5] Set("SIP/1010-00000008", "MOHCLASS=default") in new stack
-- Executing [4701@restrictedroute-13cee27a2bd93915479f049378cffdd3:6] Set("SIP/1010-00000008", "_NODEST=") in new stack
-- Executing [4701@restrictedroute-13cee27a2bd93915479f049378cffdd3:7] Macro("SIP/1010-00000008", "dialout-trunk,3,4701,,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/1010-00000008", "DIAL_TRUNK=3") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1010-00000008", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1010-00000008", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/1010-00000008", "DIAL_NUMBER=4701") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/1010-00000008", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/1010-00000008", "OUTBOUND_GROUP=OUT_3") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1010-00000008", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1010-00000008", "1?skipoutcid") in new stack
-- Goto (macro-dialout-trunk,s,12)
-- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/1010-00000008", "0?sub-flp-3,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/1010-00000008", "OUTNUM=4701") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/1010-00000008", "custom=IAX2/VOIP_CONNECTION") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1010-00000008", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Ttr)") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/1010-00000008", "0?Set(DIAL_TRUNK_OPTIONS=TtrM(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:17] Macro("SIP/1010-00000008", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1010-00000008", "") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1010-00000008", "0?skipcrm") in new stack
-- Executing [s@macro-dialout-trunk:19] Set("SIP/1010-00000008", "_CRMDIRECTION=OUTBOUND") in new stack
-- Executing [s@macro-dialout-trunk:20] Set("SIP/1010-00000008", "_CRMDESTINATION=4701") in new stack
-- Executing [s@macro-dialout-trunk:21] Set("SIP/1010-00000008", "_CRMSOURCE=1010") in new stack
-- Executing [s@macro-dialout-trunk:22] AGI("SIP/1010-00000008", "sangomacrm.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
-- AGI Script sangomacrm.agi completed, returning 0
-- Executing [s@macro-dialout-trunk:23] Set("SIP/1010-00000008", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
-- Executing [s@macro-dialout-trunk:24] NoOp("SIP/1010-00000008", "CRM Finished") in new stack
-- Executing [s@macro-dialout-trunk:25] GotoIf("SIP/1010-00000008", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:26] ExecIf("SIP/1010-00000008", "1?Set(CONNECTEDLINE(num,i)=4701)") in new stack
-- Executing [s@macro-dialout-trunk:27] ExecIf("SIP/1010-00000008", "1?Set(CONNECTEDLINE(name,i)=CID:1010)") in new stack
-- Executing [s@macro-dialout-trunk:28] ExecIf("SIP/1010-00000008", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)1010)") in new stack
-- Executing [s@macro-dialout-trunk:29] GotoIf("SIP/1010-00000008", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:30] Dial("SIP/1010-00000008", "IAX2/VOIP_CONNECTION/4701,300,Ttr") in new stack
-- Called IAX2/VOIP_CONNECTION/4701
-- Hungup 'IAX2/VOIP_CONNECTION-17597'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:31] NoOp("SIP/1010-00000008", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 50") in new stack
-- Executing [s@macro-dialout-trunk:32] GotoIf("SIP/1010-00000008", "0?continue,1:s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/1010-00000008", "RC=50") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/1010-00000008", "50,1") in new stack
-- Goto (macro-dialout-trunk,50,1)
-- Executing [50@macro-dialout-trunk:1] Goto("SIP/1010-00000008", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/1010-00000008", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 50 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] ExecIf("SIP/1010-00000008", "1?Set(CALLERID(number)=1010)") in new stack
-- Executing [4701@restrictedroute-13cee27a2bd93915479f049378cffdd3:8] Macro("SIP/1010-00000008", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/1010-00000008", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/1010-00000008", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/1010-00000008", "1?intracompany,1") in new stack
-- Goto (macro-outisbusy,intracompany,1)
-- Executing [intracompany@macro-outisbusy:1] Playback("SIP/1010-00000008", "all-circuits-busy-now&please-try-call-later, noanswer") in new stack
-- Playing 'all-circuits-busy-now.ulaw' (language 'en')

0x7f40d80168f0 -- Probation passed - setting RTP source address to 192.168.111.56:16432
-- Playing 'please-try-call-later.ulaw' (language 'en')
-- Executing [intracompany@macro-outisbusy:2] Congestion("SIP/1010-00000008", "20") in new stack
[2017-06-14 15:11:47] WARNING[8560][C-00000009]: channel.c:4991 ast_prod: Prodding channel 'SIP/1010-00000008' failed
== Spawn extension (macro-outisbusy, intracompany, 2) exited non-zero on 'SIP/1010-00000008' in macro 'outisbusy'
== Spawn extension (restrictedroute-13cee27a2bd93915479f049378cffdd3, 4701, 8) exited non-zero on 'SIP/1010-00000008'
-- Executing [h@restrictedroute-13cee27a2bd93915479f049378cffdd3:1] Hangup("SIP/1010-00000008", "") in new stack
== Spawn extension (restrictedroute-13cee27a2bd93915479f049378cffdd3, h, 1) exited non-zero on 'SIP/1010-00000008'
-- SIP/1010-00000008 Internal Gosub(crm-hangup,s,1) start
-- Executing [s@crm-hangup:1] NoOp("SIP/1010-00000008", "Sending Hangup to CRM") in new stack
-- Executing [s@crm-hangup:2] NoOp("SIP/1010-00000008", "HANGUP CAUSE: 34") in new stack
-- Executing [s@crm-hangup:3] ExecIf("SIP/1010-00000008", "0?Set(_CRMVOICEMAIL=)") in new stack
-- Executing [s@crm-hangup:4] NoOp("SIP/1010-00000008", "MASTER CHANNEL: 1497453103.15 = 1497453103.15") in new stack
-- Executing [s@crm-hangup:5] GotoIf("SIP/1010-00000008", "0?return") in new stack
-- Executing [s@crm-hangup:6] Set("SIP/1010-00000008", "_CRMHANGUP=1") in new stack
-- Executing [s@crm-hangup:7] AGI("SIP/1010-00000008", "sangomacrm.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
-- AGI Script sangomacrm.agi completed, returning 0
-- Executing [s@crm-hangup:8] Return("SIP/1010-00000008", "") in new stack
== Spawn extension (restrictedroute-13cee27a2bd93915479f049378cffdd3, h, 1) exited non-zero on 'SIP/1010-00000008'
-- SIP/1010-00000008 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

Here is the IAX2 Trunk configuration:
Server1:
General:

Trunk Name = VOIP_CONNECTION
Hide CallerID = No
Outbound CallerID = Blank
CID Options = Allow Any CID
Maximum Channels = Blank
Asterisk Trunk Dial Options = T (this field is grayed out, I can't change it) and the button is on System
Continue if Busy = No
Disable Trunk = No

Dial Number Manipulation Rules = Blank

IAX Settings:
Outgoing:
Trunk Name = VOIP_CONNECTION
PEER Details =

username=admin
secret=password
host=ip-of-server2
type=friend
context=from-internal
qualify=yes
qualifyfreqok=25000
transfer=no
trunk=yes
forceencryption=yes
encryption=yes
auth=md5
requirecalltoken=no

Server2 is exactly the same, with 1 difference:

in PEER Details, host=ip-of-server1

On both servers:
CLI> iax2 show peers
Server1:

Name/Username Host Mask Port Status Description
VOIP_CONNECTION ip-of-server2 (S) 255.255.255.255 4569 (T) (E) OK (3 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]

Server2:

Name/Username Host Mask Port Status Description
VOIP_CONNECTION ip-of-server1 (S) 255.255.255.255 4569 (T) (E) OK (1 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]

Can anyone help here?
Thanks

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@munozj wrote:

I have two FreePBX systems (one Distro 13 and the other on CentOS running 13)
I've setup two IAX2 trunks

Internal PBX (Distro)
Trunk Name: azsc-tie

host=xxxx
username=6301pbx
secret=xxxx
type=peer
qualify=yes
trunk=yes
insecure=port,invite
context=from-internal
auth=md5
requirecalltoken=no

Cloud PBX (CentOS FreePBX 13)
Trunk Name: 6301pbx

host=xxxx
username=azsc-tie
secret=xxxx
type=peer
qualify=yes
trunk=yes
insecure=port,invite
context=from-internal
auth=md5
requirecalltoken=no

I'm seeing the trunks connected:

iax2 show peers
Name/Username Host Mask Port Status Description
azsc-tie/6301pb 10.30.3.52 (S) 255.255.255.255 4569 (T) OK (13 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]

and I have outgoing routes on both taking calls that match extension numbers on the opposite systems and push them down the IAX trunks.

I have no specific incoming routes since it's just for station-to-station calling.

If I test calling one system to the other I get a busy/congestion message and I'm not seeing any activity on the other system's console. Same behavior on both systems.

Executing [s@macro-dialout-trunk:23] Dial("SIP/6000-0000000a", "IAX2/6301tie/5331,300,T") in new stack
-- Called IAX2/6301tie/5331
-- Hungup 'IAX2/6301tie-22104'
== Everyone is busy/congested at this time (1:0/0/1)

These systems are appearing internal to each other and don't have any external firewalls blocking. (internal IP connectivity only)

It almost appears that they trunks are not connected even though the IAX2 show peers indicates that they are.

I had these system half connected with SIP and was getting calls back and forth that way. Just can't seam to get the same results with IAX.

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@mhessels wrote:

Hi There
I am pretty new to FreePBX. I have a working phonesystem running but I am wondering if there is a way to create a dynamic local directory. For instance, have a spreadsheet on a server that when I change a user it automatically uploads to the phone? I am using Yealink T46s and T48s phones in this environment. Any ideas or help is greatly appreciated.

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@donkelly wrote:

I need to call a third-party PBX and provide an extension number to the auto attendant. Searches have indicated that a Custom Destination will handle it. My (non-working) solution is based on an old post by Lorne Gaetz. I have created a feature code that finds its way to a custom destination using the following in /etc/asterisk/extensions_custom.conf:

[custom-dial-test]
exten => s,1,Dial(SIP/enventis/6515551212,60,D(wwwwwwwwww123))

Dialing the feature code results in this:

[2017-06-16 08:05:45] VERBOSE[20154] pbx.c: -- Goto (custom-dial-test,s,1)
[2017-06-16 08:05:45] VERBOSE[20154] pbx.c: -- Executing [s@custom-dial-test:1] Dial("SIP/1074-00000060", "SIP/enventis/6515551212,300,D(wwwwwwwwww123)") in new stack
[2017-06-16 08:05:45] VERBOSE[20154] netsock2.c: == Using SIP RTP TOS bits 184
[2017-06-16 08:05:45] VERBOSE[20154] netsock2.c: == Using SIP RTP CoS mark 5
[2017-06-16 08:05:45] VERBOSE[20154] app_dial.c: -- Called SIP/enventis/6515551212
[2017-06-16 08:05:45] VERBOSE[20154] app_dial.c: -- SIP/enventis-00000061 is making progress passing it to SIP/1074-00000060
[2017-06-16 08:06:45] VERBOSE[20154] app_dial.c: -- No one is available to answer at this time (1:0/0/0)
[2017-06-16 08:06:45] VERBOSE[20154] pbx.c: -- Auto fallthrough, channel 'SIP/1074-00000060' status is 'NOANSWER'

The phone number, of course, is a valid one--not 5551212. After a minute of busy tone, the call is terminated.

When I simply dial 6515551212 at the extension, the call completes OK.

[2017-06-16 08:12:08] VERBOSE[20179] pbx.c: -- Executing [6512764838@from-internal:1] Macro("SIP/1074-0000006d", "user-callerid,LIMIT
,") in new stack
[2017-06-16 08:12:08] VERBOSE[20179] pbx.c: -- Executing [s@macro-user-callerid:1] Set("SIP/1074-0000006d", "AMPUSER=1074") in new st
ack
[2017-06-16 08:12:08] VERBOSE[20179] pbx.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1074-0000006d", "0?report") in new sta
ck
[2017-06-16 08:12:08] VERBOSE[20179] pbx.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1074-0000006d", "1?Set(REALCALLERIDNUM
=1074)") in new stack
...
[2017-06-16 08:12:08] VERBOSE[20179] pbx.c: -- Executing [s@macro-dialout-trunk:22] Dial("SIP/1074-0000006d", "SIP/enventis/6515551212,300,") in new stack
[2017-06-16 08:12:08] VERBOSE[20179] netsock2.c: == Using SIP RTP TOS bits 184
[2017-06-16 08:12:08] VERBOSE[20179] netsock2.c: == Using SIP RTP CoS mark 5
[2017-06-16 08:12:08] VERBOSE[20179] app_dial.c: -- Called SIP/enventis/6515551212
[2017-06-16 08:12:11] VERBOSE[20179] app_dial.c: -- SIP/enventis-0000006e is making progress passing it to SIP/1074-0000006d
[2017-06-16 08:12:14] VERBOSE[20179] app_dial.c: -- SIP/enventis-0000006e is making progress passing it to SIP/1074-0000006d
[2017-06-16 08:12:19] VERBOSE[20179] app_dial.c: -- SIP/enventis-0000006e answered SIP/1074-0000006d
...

There’s a lot missing at the ellipses—I’m assuming that people familiar with this stuff know what must have been in there. I’m happy to provide the full log! (Right. This stuff does not happen when using the Custom Destination.)

Both the good and the bad seem to be doing the same thing, with different results:
[2017-06-16 08:05:45] VERBOSE[20154] app_dial.c: -- Called SIP/enventis/6515551212

I’ve also tried simplifying this by omitting the “D(wwwwwwwwww123)” with the same result.

I’d really appreciate some suggestions!

--Don

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@tsollc wrote:

I've recently discovered a really strange issue. I've upgraded several locations to the latest version of FreePBX, and I discovered issues with some system recordings, such as the talking clock, and voicemail menu.

If I call into voicemail (*97) with no messages, I simply hear "You have" and the rest is silent. It seems like the vm-no.ulaw file and everything after cannot be heard. I can see in the asterisk logs that it is indeed playing the rest of the ulaw files to complete the announcement.

The same thing occurs with the clock (*60). Depending on the time, sometimes I will hear the whole announcement, unless the announcement includes a 4.ulaw or a 7.ulaw, then that file, and everything afterwards is silent. Again, by looking a the logs, I can see that is playing the correct files.

The vm-no.ulaw, 4.ulaw, and 7.ulaw files are the only ones that I've been able to consistently reproduce the issue with, but I haven't taken the time to try them all. I have tried downloading more recent versions of the .ulaw files from asterisk, and replacing the existing ones, but that didn't help. Secondly I did double check all permissions, and they match the settings on the working ulaw files.

Also this issue only occurs on the one brand of phones (ascom) I've been through the rounds with their tech support, to no avail.

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@tonyg wrote:

for anyone interested, this is a continuation of a discussion about bugs started in a different thread:

Tony, I am not sure if you intended on putting a link into you last comment, I did not see one, but I think this is the general accepted definition of a bug:

Re-producability has nothing to do with whether a behavior is a bug or support issue, but I do understand that if you cannot reproduce the issue, you cannot fix it, and in fact, you may need to close a bug ticket for that reason. That still does not make it a support issue. A support issue is a behavior that is a result of the mis-configuration or misuse of software resulting in undesired behavior. In the example of FREEPBX-14517, presumably this was closed because it could not be reproduced....that makes sense, if you cannot reproduce it, you cannot fix it. But what does not make sense in my mind is that it was labeled a support issue. Later when I raised this issue again and another user also related the same issue, userman v14.0.3.11 was produced to fixed the login issue....so clearly this is a bug, not a support issue.

Andrew asked at one point "My honest question is what do you think a ticket gains you over the forum thread?" My response to this is two fold, first, as a user, to be honest, it doesn't matter at all, as long as the issue is resolved, so as a user, i would concede this and stop using issues.freepbx.org But, from the FreePBX perspective....I would think it matters quite a lot. I would think that someone would want to see bug report metrics, to understand how many bugs are there, how many have been resolved etc. If something is worked on in the support forums without a bug report....then that skews the numbers. But maybe no one is looking? Anyway, those are my thoughts today.....I am open to hear yours.

thanks

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@cynjut wrote:

Just tried an "edge" upgrade from the UI and am getting a new error, specifically that the username directories table doesn't exist. Tried uninstalling and reinstalling several packages and am not having any luck getting this going.

FreePBX Utility version 13.0.192.14

Now that I've done all of this, I also can't log in (since userman is how my user is being managed and it was uninstalled and won't reinstall).

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@kfkenshin wrote:

Hi, I ran into a problem where we occasionally lose registration with our provider.
1. What could be the problem? Where should I start troubleshooting?
2. How / Where do I re-register? I tried doing fwconsole reload / restart but that doesn't initiate the registration. And what's even worse is even if I RESTART the system, it sill won't register. The only way I can register is when I go into my trunk settings and change some settings and hit apply config, then it registers....

Thank you.

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@DarkQuark wrote:

Greetings, I have several systems using the OSS EPM module. It looks like the latest version is either having issues with my system setups or there is just an issue with it in general. Going into Package manager you are confronted with Spanish and if you click Check for Update you get the attached. Now that is on a system that has been upgraded all the way to the latest version but never had any OSS items configured.

On a system that has previously run/setup OSS EPM, going into Package Manager and checking for update seems to work fine you just get the wrong language.

I noticed that on an system that OSS EPM was not on before the location in the error is not there (/var/www/html/admin/modules/_ep_phone_modules/endpoint). So I assume something is not generating that section where normally it would.

Any help or suggestions would be greatly appreciated. Thank You!

Capture.PNG1585x756 715 KB

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@KLB wrote:

Good afternoon,

i'm running FreePBX 13.0.192.9 with fax pro module (already licenced)

I would like to send and receive fax through a dedicated trunk that is delivered by my provider. (for the moment we use a cisco spa112 with the trunk config set that is directly connected to a pci modem card in a windows fax server...)

The codec used by the provider is G711a (alaw), a inbound route is dedicated to fax only.

Here is my current config:

[Settings, asterisk ip settings]
alaw codec is enabled
T38 Pass-Through: no

[Connectivity,INBOUND Route] Destination: fax recipient/fax user, Detect Faxes is not enabled, all other options are default

[Admin,User mangement] Fax user (linked extension: none) with email configured (server is able to send emails), fax feature is enabled (store locally: yes), Outgoing Station ID: equal to inbound route number

[Settings, fax configuration] Default Local Station Identifier: equal to inbound route number

[Connectivity,Trunks]
Outgoing:
type=peer
secret=*********
insecure=invite
host=ip_of_the_trunk (provider)
domain=ip_of_the_trunk (provider)
defaultuser=xxxxxxxxxxx

Incoming:
USER Context: trunk_user_name
type=peer
secret=*********
insecure=invite
host=ip_of_the_trunk (provider)
fromuser=trunk_user_name
fromdomain=ip_of_the_trunk (provider)
context=from-trunk

Please can you confirm settings i have to apply in freepbx:

i did several test but i'm not able to receive faxes, no logs are displayed (even if the log level have been set to full) when do a reception test (send fax from another working fax system), if i affect a extension to the user then i redirect the inbound routeto this extension it's working, i can dial the did number and the extension ring

• do i need to add faxdetect=yes in Chain sip settings ? (it seem if it's not enabled, fax logs will not appear, even if the log level have been set to full)

• do i have to add a special option in trunk config for fax?

Thank you in advance for your help

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@jessy5765 wrote:

We have a client who is a medical facility. They have calls on all lines all day long. They need a Trunk/Inbound Route Dedicated to have for JUST the doctors to call in on. That is the easy part. Now I need to modify the CallerID to prepend DOCTOR: to it so it would look something like DOCTOR:7342345555

I know you can use queues to do this, but I dont want the doctors to trudge their way through a queue. I see there is a few articles on here about modifying incoming caller ID but nothing that I can find like what I am looking for.

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@tty0744 wrote:

FreePBX version: 13.0.192.9
Asterisk version: 11.14.2
CentOS: 6.5 (Final)

I'm trying to upgrade 'Process Management,' but to no avail.

Error produced...

Please wait while module actions are performed

Downloading and Installing pm2
Downloading pm2
Installing pm2
Untarring..Done
Node is not installed
Error(s) installing pm2:
Failed to run installation scripts
Updating Hooks...Done
Setting Permissions...Done

I've tried (but to no avail):

yum upgrade nodejs

yum remove nodejs*
yum install nodejs

What am I missing here?

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@mangaskahn wrote:

Hi everyone, I've run into a problem and need some help diagnosing it. All calls between extensions ring normally, but as soon as they are answered, the display shows "held". Pressing the hold button does not change the "held" status. Inbound calls from outside do the same thing, as well as dialing voicemail. I'm using FreePBX 13.0.192.9 with Polycom SP550 phones. Please let me know what information is needed to help diagnose the problem. Any help is appreciated. Thanks!

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@ABBA wrote:

received email for upgrade note:

There are 5 modules available for online upgrades
backup 13.0.26.6 (current: 13.0.26.5)
core 13.0.120.2 (current: 13.0.119.10)
framework 13.0.192.9 (current: 13.0.192.8)
ivr 13.0.27.3 (current: 13.0.27.1)
recordings 13.0.30.11 (current: 13.0.30.9)

Unfortunately server disordered after upgraded.

can't pick incoming call
can't make outgoing call
can't call extensions
voicemail password not working

Fortunately recovered from full system backup.

I also fresh-install an new machine: still debian jessie+freepbx 13, reload failed,

Reload failed because retrieve_conf encountered an error: 1

click here for more info
1 error(s) occurred, you should view the notification log on the dashboard or main screen to check for more details.

Thus, for debian+freepbx users, be careful for above upgrades.

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@akira567jp wrote:

”A network interface that is assigned to the 'Trusted' zone has been detected. This is a misconfiguration. To ensure your system is protected from attacks, please change the default zone of interface 'eth0'.”

What should I do against eth0? Where and what should I set and how?

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