@USA_Phone_Man wrote:

I have multiple installations that we use DSS/MPK buttons to show a visual lamp indication of specific park orbits. This way a receptionist can answer a call, press a Green Park button (which then turns red) Make a Page, Sales you have a call on Park 71, the sales team can then go to any phone that has the same park orbit programmed on a key, press the Park Key that is programmed for Park 71 and take the call.

This has worked very well for many of our customers. the problem is, if two people try to park a call on the same park button at the same time, the caller are conferenced together and the FreePBX can no longer have access to the callers.

This has proved to be a problem for some or our customers.

We are using Free PBX 11 and Asterisk 11 distro.

Is there a solution to this problem ?

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@fetoa wrote:

Hi,

I have System Admin Pro module activated.

I've realized tha VPN service is restarted when creating an extensión, so all calls of registered remote extensions that are connected via vpn falls down.

Is that an error? Why is openvpn restarted? Is really necesary?

Regards

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@ttrm007 wrote:

I'm new to Freepbx and have it working quite well with one exception. I cannot seem to figure out how to get ddns service working / set up properly. I'm sure I'm just missing something with NAT or something. I am not a network expert either so sorry. I've been through Wiki and all seems simple but it just doesn't work.

My setup:
FreePBX 13.0.192.9
DDNS 99999999.deployments.pbxact.com
SIPSTATION for trunk service

I can access my network by going to this DDNS address so I know that part is working. Everything works great when, in Asterisk SIP settings, I have external address as my WAN Ip Address, Chan SIP Settings IP Configuration set to Static.

My ISP changes my IP address daily and I don't realize the phones are down until I try to make a call. At this very moment, if I put my DDNS address in Chan SIP settings and change it to dynmaic it works. If I change my External Address to the 99999999.deployments.pbxact.com it works. BUT, when my IP address changes again today, it will all stop working. I have the time interval in system admin DDNS set to lowest 15 mins. 15 Mins is too long for phones to be down for one, but for 2, it NEVER updates and system never reconnects after an IP address change.

Thanks for any help you might be able to offer.

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@chrisishardcore wrote:

I have an odd problem that was happening, then wasn't happening, now happening again with FreePBX and Polycom VVX500 phones. I'll do my best to describe it. A new call comes in to a ring group that has let's say three extensions. After about two or three rings, instead of the phone just showing the new call, it lists multiple calls - basically it lists the other ringing extensions as if they are also calls saying "Remote User: Name EXT" and then "To: sip:XXX@server" and then "From:"

If you scroll all the way down that list on the screen, you can answer the original call at the bottom, but if you just pick up the phone, it tries to answer the fake extension ringing calls first. Any ideas?

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@gatozgz wrote:

Hello, good afternoon:

When I Install FreePBX from the FreePBX.ISO I can see that it is created 2 MySQL databases:

• asteriskcdrdb
  I understand that on this database is it saved the info. related with the CDR (using odbc+mysl or only mysl) which is taken from the Asterisk dialplan (extensions.conf, etc).
  The "CDR reporting module" of FreePBX read only the info. which is written on MySQL asteriskcdrdb DB, true? It doesnt use the info of .csv, sqlite or postgree CDR's, true?

• asterisk
  What info. there is exactly on this database? what module is reading/writting on this database? in which cases is used this database, on the dialplan of Freepbx?

And another question, you can read/write on both MySQL databases directly or through ODBC connector, true? If I only use cdr_mysql.so and res_config_mysql.so (and dont use any module related with odbc, postgre and sqlite) on theory there is no problem related with the correct asterisk-freepbx working, true?

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@kbocek wrote:

I made a blacklist entry for:

"800 Service" <8444181595>

But they called again and it went through. Does anyone have any idea why?

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@GSnover wrote:

And yes, I have done "fwconsole chown" several times - here is the error:

Saving Backup 7...done!
Initializing Backup 7
Backup Lock acquired!
Running pre-backup hooks...
Adding items...
rsync: send_files failed to open "/usr/sbin/acpid": Permission denied (13)
rsync: send_files failed to open "/usr/sbin/build-locale-archive": Permission denied (13)
rsync: send_files failed to open "/usr/sbin/glibc_post_upgrade.x86_64": Permission denied (13)
rsync: send_files failed to open "/usr/sbin/groupadd": Permission denied (13)
rsync: send_files failed to open "/usr/sbin/groupdel": Permission denied (13)
rsync: send_files failed to open "/usr/sbin/groupmems": Permission denied (13)
rsync: send_files failed to open "/usr/sbin/groupmod": Permission denied (13)
rsync: send_files failed to open "/usr/sbin/suexec": Permission denied (13)
rsync: send_files failed to open "/usr/sbin/tzdata-update": Permission denied (13)
rsync: send_files failed to open "/usr/sbin/useradd": Permission denied (13)
rsync: send_files failed to open "/usr/sbin/userdel": Permission denied (13)
rsync: send_files failed to open "/usr/sbin/userhelper": Permission denied (13)
rsync: send_files failed to open "/usr/sbin/usermod": Permission denied (13)
rsync error: some files/attrs were not transferred (see previous errors) (code 23) at main.c(1039) [sender=3.0.6]

In what context does the Backup run - root?

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@mlpalumbo wrote:

Hi all,

I am using Zoiper to try and make external calls, however FPBX keeps giving me issues, Zoiper(on my android phone) shows a 503 Service unavailable. And FreePBX logs is showing the following:

[2017-06-22 18:40:00] VERBOSE[13680][C-0000135a] pbx.c: -- Executing [1330766***@from-internal:1] ResetCDR("SIP/330766****-000018ae", "") in new stack
[2017-06-22 18:40:00] VERBOSE[13680][C-0000135a] pbx.c: -- Executing [1330766***@from-internal:2] NoCDR("SIP/1330766***-000018ae", "") in new stack
[2017-06-22 18:40:00] VERBOSE[13680][C-0000135a] pbx.c: -- Executing [1330766***@from-internal:3] Progress("SIP/1330766***-000018ae", "") in new stack
[2017-06-22 18:40:00] VERBOSE[13680][C-0000135a] pbx.c: -- Executing [1330766***@from-internal:4] Wait("SIP/1330766***-000018ae", "1") in new stack
[2017-06-22 18:40:01] VERBOSE[13680][C-0000135a] pbx.c: -- Executing [1330766***@from-internal:5] Progress("SIP/1330766***-000018ae", "") in new stack
[2017-06-22 18:40:01] VERBOSE[13680][C-0000135a] pbx.c: -- Executing [1330766***@from-internal:6] Playback("SIP/1330766***-000018ae", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2017-06-22 18:40:01] VERBOSE[13680][C-0000135a] file.c: -- Playing 'silence/1.gsm' (language 'en')
[2017-06-22 18:40:02] VERBOSE[13680][C-0000135a] file.c: -- Playing 'cannot-complete-as-dialed.gsm' (language 'en')
[2017-06-22 18:40:04] VERBOSE[13680][C-0000135a] pbx.c: == Spawn extension (from-internal, 1330766***, 6) exited non-zero on 'SIP/1330766***-000018ae'

I am at a lose on this one, i have tried everything i can think of, the extension is set for NAT and the extension works on Ekiga. I am connecting to wifi so it is not a IPSec issue. The account is showing as registered. Anyone have any insight?

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@matsyuf wrote:

Hello,

I am newbie and request help with FreePBX.

I have FreePBX 13.0.190.7 on my virtual machine and Huawei Modem/dongle (Model: E153u - 2 ). I would like to configure the modem with my Freepbx.

Kindly help take me through the steps i should take to have the modem recognized by the freepbx system and also configure inbound trunk.

Thanks

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@DarrynL wrote:

Hi There everyone.

This forum has been so much help to me all through my FreePBX Usage, It is an awesome Product.

I have one question, when i dial a number that is busy, for example i dial my cell phone while it is in the middle of another call, it tells me that "all circuits are busy now" I was wondering if there was a way to change it so when the other end is busy it would let me know that specifically?

Kind Regards

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@skamm71 wrote:

Hey guys

I do have an issue with FreePBX changing my "FROM" field in the invite to German Telekom in case of call diversion. I want to send the real originator of the call in the FROM field, but Freepbx is changit it back to the trunk CID. I set "allow any CID" in the configure trunks section but this does not seem to work.

I am using Freepbx as a kind of session border controller in between my Avaya PBX and German Telekom as SIP provider. So the calls are coming from Avaya CM via Avaya SM to Asterisk and then routed to German Telekom.

Here is the Invite I am sending to Asterisk (traced with tshark on the Freepbx):

INVITE sip:0xx75051645@sip.saschaxxxx.local SIP/2.0
P-AV-Message-Id: 2_1
Route: sip:192.168.1.89;lr;phase=terminating;m-type=audio
Diversion: "Sascha prv." sip:0xxxx749476@sip.saschaxxxx.local;reason=unknown;privacy=off;screen=no
P-Asserted-Identity: "Sascha SLT Wohnzimmer" sip:0yyyy6223489@sip.saschaxxxx.local
Supported: 100rel, join, replaces, sdp-anat, timer
Max-Breadth: 60
P-Charging-Vector: icid-value="AAS:2584-fe347d80e7410658c0078bce3fdb829"
Session-Expires: 1200;refresher=uac
Record-Route: sip:3923e9e@192.168.1.167;transport=udp;lr
Record-Route: sip:192.168.1.67:15060;transport=udp;ibmsid=local.1495200556906_1605392_1606059;lr;ibmdrr
Record-Route: sip:192.168.1.67:15061;transport=tls;ibmsid=local.1495200556906_1605392_1606059;lr;ibmdrr
Record-Route: sip:3923e9e@192.168.1.167;transport=tls;lr
Record-Route: sip:192.168.1.60:5061;transport=tls;lr
Min-SE: 1200
Alert-Info: cid:internal@sip.saschaxxxx.local;avaya-cm-alert-type=internal
Accept-Language: en
Contact: "Sascha SLT Wohnzimmer" sip:+58323@192.168.1.60:5061;transport=tls;avext=58323;gsid=807cfafc-5806-41e7-bc73-000c29b8fde3
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY, REFER, INFO, PRACK, PUBLISH, UPDATE
Via: SIP/2.0/UDP 192.168.1.167;rport;branch=z9hG4bK626753709840614-AP;ft=192.168.1.167~13c4
Via: SIP/2.0/UDP 192.168.1.67:15060;rport=15060;ibmsid=local.1495200556906_1605393_1606060;branch=z9hG4bK626753709840614
Via: SIP/2.0/UDP 192.168.1.67:15060;rport;ibmsid=local.1495200556906_1605392_1606059;branch=z9hG4bK463129475910819
Via: SIP/2.0/TLS 192.168.1.167;branch=z9hG4bK807d4a3458641e7bc7b0c29b8fde3-AP;ft=4;received=192.168.1.167;rport=54777
Via: SIP/2.0/TLS 192.168.1.60;branch=z9hG4bK807d4a3458641e7bc7b0c29b8fde3
Via: SIP/2.0/TCP 192.168.1.248;branch=z9hG4bK807d4a3458641e7bc7b0c29b8fde3
User-Agent: Avaya CM/R017x.00.0.441.0 AVAYA-SM-7.0.1.1.701114
From: "Sascha SLT Wohnzimmer" sip:0yyyy6223489@sip.saschaxxxx.local;tag=807d42a58641e7bc790c29b8fde3
To: sip:0xx75051645@sip.saschaxxxx.local
Call-ID: 807d423c58641e7bc7a0c29b8fde3
Max-Forwards: 66
CSeq: 1 INVITE
Content-Type: application/sdp
Content-Length: 212
Av-Global-Session-ID: 807cfafc-5806-41e7-bc73-000c29b8fde3
P-Location: SM;origlocname="saschaxxxx.local";origsiglocname="saschaxxxx.local";origmedialocname="saschaxxxx.local";termlocname="AsteriskVoIPServer";termsiglocname="AsteriskVoIPServer";smaccounting="true"

v=0
o=- 1498217060 1 IN IP4 192.168.1.60
s=-
c=IN IP4 192.168.1.248
b=AS:64
t=0 0
a=avf:avc=n prio=n
a=csup:avf-v0
m=audio 2070 RTP/AVP 8 0 127
a=sendrecv
a=rtpmap:127 telephone-event/8000
a=ptime:20

And this is what Asterisk is sending to my provider:

INVITE sip:0xx75051645@tel.t-online.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.89:5060;branch=z9hG4bK088d2916;rport
Max-Forwards: 70
From: "Sascha SLT Wohnzimmer" sip:0xxxx7472266@tel.t-online.de;tag=as52018363
To: sip:0xx75051645@tel.t-online.de
Contact: sip:0xxxx7472266@192.168.1.89:5060
Call-ID: 538dfffa25a33bb8150852a12fc71d9b@tel.t-online.de
CSeq: 102 INVITE
User-Agent: FPBX-AsteriskNOW-13.0.192.8(13.16.0)
Date: Fri, 23 Jun 2017 11:24:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Diversion: "Sascha prv." sip:0xxxx749476@192.168.1.89;reason=unknown
Content-Type: application/sdp
Content-Length: 323

v=0
o=root 1601472644 1601472644 IN IP4 192.168.1.89
s=Asterisk PBX 13.16.0
c=IN IP4 192.168.1.89
t=0 0
m=audio 5086 RTP/AVP 8 9 10 0 127
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:10 L16/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-16
a=ptime:20
a=maxptime:70
a=sendrecv

What do I need to do that Asterisk keeps my from field as it was originally?
I need to use that - hopefulle German Telekom supports that - to tunnel the real calling number in case of call diversions.

TIA
Cheers
Sascha

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@frouty wrote:

Hi,
My config: Sangoma appliance, ip phones: sangoma s500 et polycom vx400
vega50 gateway for one BRI line, an sangoma FXO card 4 ports, 2 analogic lines.
Cisco SG500-28P switch

When I do :
*97 / 0 / record my unavailable message on sangoma or polycom ipphones
2 to listem to my recording sound is OK.
then call from outside , wait for the voicemail system to play my unavailable message is barely audible, but the system message like 'good bye' are OK low but OK.

I have others audio problems with the analogic lines but not the BRI lines. I don't know if there could a relation with the voicemail system audio problem.

The office was closed, pc off . Except mine of course.

Where should I go to improve that?
Many thanks
Laurent FRANCOIS

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@Alan_uk wrote:

Hi

I have a Raspberry Pi Zero with RasPBX with Asterisk 13.13.1 upgraded to 13.15.0 & FreePBX 13.0.190.11 upgraded to 13.0.192.8

I am not using any SIP provider. The aim initially is to filter incoming PSTN calls via an OBi110.

I've defined 2 ring groups: 500 for softphones from 501 etc and 600 for hardphones from 601 (the OBi110 handset 601 is the only hardphone at the moment)

When a PSTN call comes in the Dial Plan has this command:

same => n,Dial(SIP/500&SIP/600)

However this gives errors:

[2017-06-23 19:20:52] VERBOSE[6557][C-00000009] pbx.c: Executing [0@day:2] Dial("SIP/OBiTrunkSP2-00000007", "SIP/500&SIP/600") in new stack
[2017-06-23 19:20:52] WARNING[6557][C-00000009] chan_sip.c: Purely numeric hostname (500), and not a peer--rejecting!
[2017-06-23 19:20:52] WARNING[6557][C-00000009] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-06-23 19:20:52] WARNING[6557][C-00000009] chan_sip.c: Purely numeric hostname (600), and not a peer--rejecting!
[2017-06-23 19:20:52] WARNING[6557][C-00000009] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-06-23 19:20:52] VERBOSE[6557][C-00000009] app_dial.c: Everyone is busy/congested at this time (2:0/0/2)
[2017-06-23 19:20:52] VERBOSE[6557][C-00000009] pbx.c: Executing [0@day:3] Hangup("SIP/OBiTrunkSP2-00000007", "") in new stack

I have tested calling the two ring groups from softphones and that works.

I know I can list all the extensions separately or create a macro, but 1) I want to avoid having to remember to change the code when adding an extension; and 2) it just seems a natural /logical thing to ring a ring group.

I've spent a couple of hours on the web and surprised that not more people have done this.

BTW, FreePBX is an excellent product

Thanks for reading.
Alan

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@bigbear wrote:

I have been using hosted FreePBX and now have the new Yealink T58V phones installed. I was able to get video working with PJSIP with the help of comments on this bug report... https://issues.freepbx.org/browse/FREEPBX-11577

Now I am hoping I can get the calls using H264 to go direct between handsets on the local LAN instead of to the cloud and back. I know it was possible with Freeswitch to do this, not sure about Asterisk and FreePBX.

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@dicko wrote:

Don't use PJSIP yet is is very buggy

Mod edit: this was taken from http://community.freepbx.org/t/new-installation-reload-failed-still/42138/8

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@Alan_uk wrote:

Hi

I have a Raspberry Pi Zero with RasPBX with Asterisk 13.13.1 upgraded to 13.15.0 & FreePBX 13.0.190.11 upgraded to 13.0.192.8 . PSTN calls come in via an OBi110.

When a call comes into the OBi110 its Call History shows
Peer Name: blank
Peer Number: 012345678

In FreePBX there is a Trunk called =OBiTrunkSP2 and an Inbound Route that has a Custom Destination. In the Custom Destination context the command:

exten => 012345678,1,Noop(Incoming pstn call from ${CALLERID(all)})
gives Incoming pstn call from "0789123456" 0789123456

If a number is withheld the OBi110 shows
Peer Name: WITHHELD
Peer Number: blank

and the context now shows
Incoming pstn call from "OBiTrunkSP2" OBiTrunkSP2

Whilst the first case is no problem (the PSTN does not send Caller Name), the second case is problematic if there are other different status from the PSTN (e.g. INTERNATIONAL, though I guess I can check for 00 if the number is provided)

How do I figure out if 1) OBi110 is wrongly sending or 2) Asterisk/FreePBX is wrongly receiving.

I'm sure that when I incorrectly had the Trunk context set to my custom code (rather that it set to from-pstn and using Custom Extension) that the caller ID was correctly shown as "" <0789123456> but I can't find any log to prove it.

I'm reluctant to dump a whole log in this post but some things caught my eye:

>  Executing [in@sub-record-check:2] Set("SIP/OBiTrunkSP2-00000019", "FROMEXTEN=unknown"  ) in new stack
>  Executing [in@sub-record-check:3] ExecIf("SIP/OBiTrunkSP2-00000019", "11?Set(FROMEXTEN=OBiTrunkSP2)") in new stack

> Executing [s@ext-did:5] ExecIf("SIP/OBiTrunkSP2-00000019", "1 ?Set(CALLERID(name)=OBiTrunkSP2)") in new stack

Thanks for reading
Alan

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@DarrynL wrote:

Hi,

Am am getting a really odd issue where when my ISP drops my internet connection my PBX doesn't come back online on the external link correctly all the time, Sometimes it works fine though.
Does anyone have any suggestions on how to fix this maybe?
If this is not a PBX issue then i will gladly take it up with My ISP as they have been giving me shocking service lately.

My trunks deregister when the internet drops out but dont re-register when it comes back up.
If i do a manual IFdown and IFup it starts working immediately afterwards.
Like i said random question.

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@Msh wrote:

I've backed up my configs from an FPBX13 server and want to restore it on an FPBX14 fresh install without a luck. When uploading the file in restore page, seems the file uploads but nothing happens. No errors, no signs of uploaded file. Has anyone a clue?

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@fgunno wrote:

I try to setup WebRTC with HTTPS,
With chrome and Safari nothing work, but its seem to be normal...

So I do my test with Firefox, the WEBRTC work great without HTTPS, but with HTTPS the phone doesn't look to register.
Error:

[2017-06-26 14:43:09] ERROR[6542] tcptls.c: Problem setting up ssl connection: error:00000005:lib(0):func(0):DH lib, System call EOF
[2017-06-26 14:43:09] WARNING[6542] tcptls.c: FILE * open failed!
[2017-06-26 14:43:13] ERROR[6548] tcptls.c: Problem setting up ssl connection: error:00000005:lib(0):func(0):DH lib, System call EOF
[2017-06-26 14:43:13] WARNING[6548] tcptls.c: FILE * open failed!
[2017-06-26 14:43:15] ERROR[2561] chan_sip.c: Serious Network Trouble; _sipxmit returns error for pkt data
[2017-06-26 14:43:15] ERROR[6578] tcptls.c: Problem setting up ssl connection: error:00000005:lib(0):func(0):DH lib, System call EOF
[2017-06-26 14:43:15] WARNING[6578] tcptls.c: FILE * open failed!

I try with the organisation Certificate and with a Let's encrypt certificate. same error.

Any advice?

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@gwntc wrote:

whenever we try to press any of the buttons (like parking, apps, dnd, etc) on the S500 telephone we get "Failed to load XML file".

This is happening when we are using an internal profile that points direct to the PBXact IP using the internal IP. There are no firewall policies that are blocking it as it's internal. I have tried to enter the "fwconsole upgrade restapps" command as stated in this post however it doesn't work and throws -- [InvalidArgumentException]
Command "upgrade" is not defined. --

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