@faisalkhan wrote:

I have installed Freepbx 13 on Centos 7 manually following this link
https://wiki.freepbx.org/display/FOP/Installing+FreePBX+13+on+CentOS+7
all went ok but when I want to install commercial module system admin It gives the error for php zend guard and incron.
I install zend guard loader and incron but when I again go for install it prompts me error of permissions. I gave the permissions using chmod but now it prompts this error.

error.PNG1584x800 116 KB

I followed one of the link from the below link for Repos downloading.

from the below link I downloaded the repos
http://wiki.freepbx.org/display/FPG/Installation+on+CentOS+and+RHEL+based+systems
but when I
use this command "yum -y install php-5.3-zend-guard-loader sysadmin fail2ban incron ImageMagick"
it gave me this error

image.png1574x549 22.9 KB

Need help in this please anyone can help me in this really gonna appreciate your help.

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@HateTheDrake wrote:

Go easy... first post.

I'm looking to make calls via the Linphone app over the internet and need to verify if/how encrypted it is. I've been testing with TLS and SRTP with a letsencrypt cert, though calls either hang up or have one-way audio with TLS. Calling via UDP with SRTP enabled appears to work fine, though I'm concerned about the lack of tls.

Does this mean the headers are simply not encrypted but the payload is? What are the real-world risks with that?
Linphone lists the call as encrypted fwiw.

The infrastructure looks like:

iPhone > internet > FortiGate firewall (using SIP ALG) > DNAT to FreePBX VM > Internal SIP trunk to Mitel System.

Mitel then rings internally or out the PRI if necessary.

A snippet of the debug:

<--- SIP read from UDP:x.x.x.x:10038 --->
INVITE sip:11355 (at) pbx.example.com SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:10038;rport;branch=z9hG4bK+234f5a05e4a47dfcc27e5876b56efe0f1+s676+1
From: ;tag=s676+1+41400001+5ee7f7b6
To: "11355"
Call-ID: ESLYjb3nw8-S
CSeq: 20 INVITE
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 694
Contact: ;+sip.instance=""
User-Agent: Linphone_iPhone.6_iOS10.3.2/3.16.3 (belle-sip/1.6.1)

v=0
o=1901 1782 4007 IN IP4 x.x.x.x
s=Talk
c=IN IP4 x.x.x.x
b=AS:380
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 10031 RTP/SAVP 0 8 9 18 101
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:oYEUJUe+U2JygsOg4JT3k8ysyj4Sxffm84/gsodH
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:8dU4z4Wk9Pl+LdhWAmU0BJiVRlIa4bbKdmY8jrAk
a=crypto:3 AES_256_CM_HMAC_SHA1_80 inline:ymNIvIUHt4vkLOEfNttpZQ/ef0gPt69Q2uQ+e8qn6UEL6mdgbjXlylo40rr1Ng==
a=crypto:4 AES_256_CM_HMAC_SHA1_32 inline:8My2B4d8P6pD7+CMBT/xZ3YhwH2IULHDMwNppr1P6lP3rGtAJm05V+WFVVvwWQ==
a=rtcp-fb:* trr-int 5000
a=rtcp-fb:* ccm tmmbr
<------------->
--- (13 headers 16 lines) ---
Sending to x.x.x.x:10038 (no NAT)

Sending to x.x.x.x:10038 (no NAT)
Using INVITE request as basis request - ESLYjb3nw8-S
Found peer '1901' for '1901' from x.x.x.x:10038

<--- Reliably Transmitting (no NAT) to x.x.x.x:10038 --->
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP x.x.x.x:10038;branch=z9hG4bK+234f5a05e4a47dfcc27e5876b56efe0f1+s676+1;received=x.x.x.x;rport=10038
From: ;tag=s676+1+41400001+5ee7f7b6
To: "11355" ;tag=as1131dd4e
Call-ID: ESLYjb3nw8-S
CSeq: 20 INVITE
Server: FPBX-13.0.192.8(11.25.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4dc244ac"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'ESLYjb3nw8-S' in 88448 ms (Method: INVITE)

<--- SIP read from UDP:x.x.x.x:10038 --->
ACK sip:11355 (at) pbx.example.com SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:10038;rport;branch=z9hG4bK+234f5a05e4a47dfcc27e5876b56efe0f1+s676+1
From: ;tag=s676+1+41400001+5ee7f7b6
To: "11355" ;tag=as1131dd4e
Call-ID: ESLYjb3nw8-S
CSeq: 20 ACK
Max-Forwards: 70
Content-Length: 0

<------------->

--- (8 headers 0 lines) ---

<--- SIP read from UDP:x.x.x.x:10038 --->
INVITE sip:11355 (at) pbx.example.com SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:10038;rport;branch=z9hG4bK+26425e614990def04ddef93c5ce0af8d1+s676+1
From: ;tag=s676+1+41400001+70a5dfdf
To: "11355"
Call-ID: ESLYjb3nw8-S
CSeq: 21 INVITE
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 694
Contact: ;+sip.instance=""
User-Agent: Linphone_iPhone.6_iOS10.3.2/3.16.3 (belle-sip/1.6.1)
Authorization: Digest realm="asterisk", nonce="4dc244ac", algorithm=MD5, username="1901", uri="sip:11355 (at) pbx.example.com", response="abc809463b8e0bb858edc5fc862686d6"

v=0
o=1901 1782 4007 IN IP4 x.x.x.x
s=Talk
c=IN IP4 x.x.x.x
b=AS:380
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 10034 RTP/SAVP 0 8 9 18 101
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:oYEUJUe+U2JygsOg4JT3k8ysyj4Sxffm84/gsodH
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:8dU4z4Wk9Pl+LdhWAmU0BJiVRlIa4bbKdmY8jrAk
a=crypto:3 AES_256_CM_HMAC_SHA1_80 inline:ymNIvIUHt4vkLOEfNttpZQ/ef0gPt69Q2uQ+e8qn6UEL6mdgbjXlylo40rr1Ng==
a=crypto:4 AES_256_CM_HMAC_SHA1_32 inline:8My2B4d8P6pD7+CMBT/xZ3YhwH2IULHDMwNppr1P6lP3rGtAJm05V+WFVVvwWQ==
a=rtcp-fb:* trr-int 5000
a=rtcp-fb:* ccm tmmbr
<------------->

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@choeschen wrote:

I was asked to help with creating an alert system in the case someone is threatening our receptionist. Our front desk is on the first floor of our building and our offices are located on the second. If someone comes in and is threatening the receptionist we want the ability to alert the people on the second floor. The first thing that came to my mind is using the paging and intercom app. When our receptionist calls the paging number there is a beep heard on their phone. Since this will be used in an emergency situation having the receptionist's phone beep is not a good idea. Is there a way to stop the beep on the dialer's phone? We are using Snom 760s. Is there a better way to do what we are looking for?

The second part to this is we are looking at having two options, the receptionist needs help and a warning that something is happening but not to come downstairs. I know the paging and intercom app can play an announcement after the phone's auto answer but this setting is setup in the general section of the app so I can only setup one announcement that is played regardless of what paging extension is dialed. Is there some way to setup an extension that will page a list of phones and play a specific announcement?

I don't know how to get the version of our FreePBX system but our core is running version 2.11.0.1 and we have the Paging and Intercom version 2.11.0.8 installed.

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@marwan83 wrote:

Hi,

I have just completed the connection of 2 PBXs via IAX2 trunk. Here's the setup:

CloudPBX is on a public IP on the internet.
PBX1 is behind a NATing device with dynamic IP (home use internet connection)

Now, I understand that the IAX trunk has 2 ends (one from PBX1=>CloudPBX, and the other is CloudPBX=>PBX1).

The thing is, CloudPBX can not connect to PBX1, the IP keeps changing.
But, PBX1 is able to connect to CloudPBX successfully.

Would the trunk work only on one leg? I mean, will I be able to make calls from both boxes if only PBX1 registers on CloudPBX? OR both ways should register?

Also, if only one leg is enough (PBX1=>CloudPBX), so how would I set the dialing string (IAX2/**TRUNK_NAME**/EXTENSION)? What should be in place of "TRUNK_NAME?

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@charneval wrote:

Hello.
I wold like to configure a new number take from this provider " Sonetel"

Someone has ever configured a voip number by this provider?

I used this PEER detail :

type=friend
username=info
fromuser=info
fromdomain=XXXXXXXXX.it
host=sonetel.com
outboundproxy=sip.sonetel.com
disallow=all
allow=ulaw,alaw
qualify=yes
dtmf=rfc2833
nat=auto_force_rport
canreinvite=no
insecure=invite,port
context=sonetel
sendrpid=yes
secret=YYYYYY

and register string :

info@XXXXXXXX.it:YYYYYY@sip.sonetel.com

The number is registered in my freepbx but I can't receive and call.

Do you have any idea about this?

Thanks

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@RealOfAlex wrote:

I have several Cisco 7960's and a few Cisco SPA303's on my system. The SPA303s work without a problem, they can call another extension and can receive calls from other extensions. The 7960s can call another extension (the SPA303s), but they can't receive a call into the phones. When I go and look at the Asterisk Info, it shows:

Active SIP Channel(s): 0
Sip Registry: 1
Sip Peers:
Online: 0
Online-Unmonitored: 0
Offline: 1
Offline-Unmonitored: 0

Active PJSIP Channel(s): 0
PJSip Registrations: 0
PJSip Endpoints:
Available: 2
Unavailable: 3
Unknown: 3

Active IAX2 Channel(s): 0
IAX2 Registry: 1
IAX2 Peers:
Online: 0
Offline: 0
Unmonitored: 0

I'm running version: 13.14.0
Hopefully, someone could help me out with this or has had this issue before.
Thank you!

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@ghurty wrote:

I have a remote freepbx box that I do not know the root password to, but I have freepbx access. Is there a dialplan that I can write that I can use to change the root password?

Thanks

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@santicn4n wrote:

I am using FreePBX 13.0.192.8 on Ubuntu 14.04. A single phone on my network fails to call any number. The extension can receive calls from inbound routes and other extensions, but cannot dial. I constantly get NO RESPONSE errors. I have other phones on the network that operate fine. The phone is a Grandstream GXP2170. The phone was working for months and randomly stopped dialing. I started running in circles with troubleshooting so I thought I would give the forum a try.

Main Points:
Phone Registers Fine
Phone Can Receive Any Call (with two way audio)
Phone Cannot Dial Any Extension or Outbound Route

Tried upgrading phone firmware, updating freepbx modules, factory reset phone, packet captures, remaking routes, remaking extensions.

Any info or troubleshooting help would be greatly appreciated.

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@marwan83 wrote:

Hi,

Anyone knows how to disable the very first ring tone the caller hears when the extension is bring forwarded or Follow Me option is active?

Here's the scenario:

I have setup some IVR destionations to go to extensions that actually forwarded to external destinations using Follow Me option. But when I call the IVR and hit that option, I hear a first ring, the music on hold starts playing.

Any udea how to disable that ring?

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@avayax wrote:

I have been receiving ghost calls to one of my remote phones, which connects to my PBX over openvpn.
My PBX is behind a NAT firewall (which also hosts the VPN server). SIP port 5060 is open, but whitelisted to only allow traffic from my SIP provider IP addresses and a FQDN of a remote FPBX box.

Anonymous Inbound SIP Calls are not allowed, SIP guests aren't allowed either.

The phone receives calls from extensions that don't exist on my PBX, but are registered to public IP addresses.
The logs on the phone show something like this:
Call from
800@67.245.163.64
1002@67.245.167.74
10@67.245.163.64
1004@67.245.167.74

I haven't found anything in the Asterisk logs referencing those IP addresses.
Are those calls touching the FBPX server at all?

Where are those calls coming in on?
Is it the firewall in front of the remote phone that's the problem?

Also, what are those calls probing for?
Sure the incentive is toll fraud, but what information are those calls trying to gather?

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@marwan83 wrote:

Hi,

Im having a situation that is driving me crazy:

I have 2 trunks:

1 IAX to connect to PBXs
2 SIP trunk named Voxbeam used for outbound and inbound calls

my CID is 0540543700. I call PBX1, Use a call route to direct international calls to CloudPBX. CloudPBX should use Voxbeam SIP trunk for the outgoing call.

I have configured outbound routes on CloudPBX.

I have both IAX and SIP trunks on CloudPBX with correct dial manipulation rules.

However, the calls get congested. I think CloudPBX is not able to use Voxbeam trunk. It keeps trying to use the IAX trunk. Am I right?

Below are the logs:

[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [s@macro-user-callerid:1] Set("IAX2/JEDPBX-11963", "TOUCH_MONITOR=1497789963.172") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [s@macro-user-callerid:2] Set("IAX2/JEDPBX-11963", "AMPUSER=0540543700") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [s@macro-user-callerid:3] GotoIf("IAX2/JEDPBX-11963", "0?report") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [s@macro-user-callerid:4] ExecIf("IAX2/JEDPBX-11963", "1?Set(REALCALLERIDNUM=0540543700)") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [s@macro-user-callerid:5] Set("IAX2/JEDPBX-11963", "AMPUSER=") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [s@macro-user-callerid:6] GotoIf("IAX2/JEDPBX-11963", "0?limit") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [s@macro-user-callerid:7] Set("IAX2/JEDPBX-11963", "AMPUSERCIDNAME=") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [s@macro-user-callerid:8] GotoIf("IAX2/JEDPBX-11963", "1?report") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx_builtins.c: Goto (macro-user-callerid,s,15)
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [s@macro-user-callerid:15] GotoIf("IAX2/JEDPBX-11963", "1?continue") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx_builtins.c: Goto (macro-user-callerid,s,29)
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [s@macro-user-callerid:29] Set("IAX2/JEDPBX-11963", "CALLERID(number)=0540543700") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [s@macro-user-callerid:30] Set("IAX2/JEDPBX-11963", "CALLERID(name)=") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [s@macro-user-callerid:31] GotoIf("IAX2/JEDPBX-11963", "1?cnum") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx_builtins.c: Goto (macro-user-callerid,s,33)
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [s@macro-user-callerid:33] Set("IAX2/JEDPBX-11963", "CDR(cnum)=0540543700") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [s@macro-user-callerid:34] Set("IAX2/JEDPBX-11963", "CHANNEL(language)=en") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [0020233352873@from-internal:2] Set("IAX2/JEDPBX-11963", "ROUTEUSER=") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [0020233352873@from-internal:3] Set("IAX2/JEDPBX-11963", "ROUTEUSER=") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [0020233352873@from-internal:4] GotoIf("IAX2/JEDPBX-11963", "1?notblind") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx_builtins.c: Goto (from-internal,0020233352873,7)
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx.c: Executing [0020233352873@from-internal:7] GotoIf("IAX2/JEDPBX-11963", "0?,0020233352873,2:outbound-allroutes,0020233352873,2") in new stack
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] pbx_builtins.c: Goto (outbound-allroutes,0020233352873,2)
[2017-06-18 15:46:03] WARNING[25610][C-000000a5] pbx.c: Channel 'IAX2/JEDPBX-11963' sent to invalid extension but no invalid handler: context,exten,priority=outbound-allroutes,0020233352873,2
[2017-06-18 15:46:03] VERBOSE[25610][C-000000a5] chan_iax2.c: Hungup 'IAX2/JEDPBX-11963'

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@arielgrin wrote:

Hi guys: I'm coming to the forums to seek help from anyone that might be experiencing the same issue I'm experiencing with my Audiocodes 320HD phone. Whenever a BLF monitored extensions is ringing, the only way to perform a call pickup is to press the BLF key while the handset is onhook. If you take the handset offhook and press the BLF key for the ringing extension, it will dial the extension instead of picking up the call.
Does anyone on the forum owns a 320HD and is experiencing the same issue?
I have the following options set on the phone's web gui
Application server type: Asterisk
BLF activate: enabled
Call pickup: enabled

The firmware version is 1.6.0.44.43.2

I contacted Audiocodes but they just said that my phone is not under any support contact so they can't help me.

Regards, Ariel.

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@asifameer wrote:

Hi All,

Asif here, I just installed FreePBX for my home based business to setup toll free number and few extensions. I am using the version FreePBX-64bit-6.12.65. and I have installed it on VMWare 12.

The problem I am getting is when I click on Detect Network settings in Settings > Asterisk SIP Settings, I get the following error:

failed to connect to myip.freepbx.org port 5060: connection timed out.

Capture.PNG1169x369 19.2 KB

I cant figure out what to do and how to solve this problem, cuz I am new to this PBX system.
Also I have tried to create extensions locally but once I provide all the details in soft phone. my PBX server does not detect them and phones are unable to connect/register to my server.

Any help would be highly appreciated.

Thanks

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@citelum wrote:

Hello
When I change ivr default msg for "no answer" option with my custom msg into advanced option for the extension, after that msg is played, it isn't possible to record the msg, the call is inmediatly terminated.

Can you help me please ?

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@jamesnurse wrote:

After doing some random module updates that I usually find quick and easy my PBX seems to not be able to connect to the internet. I am not sure if this is just a coincidence but no network config has changed but now suddenly whenever i try to 'search online' for module updates the web control panel times out and then reloads to the dashboard. My external trunk connection has also dropped as the PBX is unable to communicate externally to the voip provider.

Also if i try to check for updates in System Admin i get this error: fopen(http://upgrades.schmoozecom.net/updates.php): failed to open stream: Connection timed out
System admin shows i am currently running version: 10.13.66-20

Core version: 13.0.119.10

I'd appreciate any suggestions

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@Leee wrote:

Hi,

I've recently set up a new FreePBX server, (latest FreePBX distro, Asterisk 11/64 bit). It's all works just great, incoming/outgoing, IVR, recording everything. Until I get a --

[2017-06-19 09:12:38] NOTICE[22522] chan_sip.c: Peer 'htel_out' is now UNREACHABLE! Last qualify: 24
[2017-06-19 09:12:38] NOTICE[22522] chan_sip.c: Peer 'htel_in' is now UNREACHABLE! Last qualify: 24

in the full log and then I'm cut off from my SIP trunk provider. Asterisk then just sits there and doesn't reconnect, and everyone gets "All circuits are busy" messages. Then the shouting and screaming starts. This can be from every a couple of days down to 10 mins.

I've been tearing my hair out for the last days, reading everything regarding having Asterisk/FreePBX connected via pfSense. I already had NAT rules in place to forward incoming requests from my SIP trunk provider through the firewall/NAT to the FreePBX VM.

I've since tried 1:1 NAT, didn't seem to help, and then an Outbound NAT rule as recommended in the pfSense docs. This doesn't seem to help either.
I've added 'keepalive=30' in the trunk settings, registertimeout and registerattempts, no improvement either.

I've tried using SIPROXD on pfSense, but the outbound traffic didn't seem to go through the proxy. And the docs seem to infer that I shouldn't be using it anyway.

I've spoken to the SIP Provider and they tell me I've jumped ports, instead of connecting to 5060 at their end, I was connecting to 38000ish. An 'aportal restart' seems to fix the issue, but of course that kills all active calls.

I'm guessing this is NAT time out issue, but I'm damned if I can find an answer, but I can't be the only person with a SIP provider who uses IP based authentication (if that's the right term where there is no username/password), who's using FreePBX behind pfSense.

Oddly, I also have another trunk that I used for testing, it's a traditional SIP setup with a username/password, and has no firewall/NAT rules in pfSense, but just works, this hasn't dropped out that I've noticed.

Can anyone help? I'm at my wits end and getting a lot of heat from the management...

Leee

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@ctznpro wrote:

I gave DISA configured to allow specific users to dial in, enter a code, receive a dial tone and call out using an internal CID. This is only a temporary solution while I setup VPN access for them. However, one of the managers was requesting access to these call logs and I can't figure out a way to grant access on a granular level like that. Both the UCP and the admin ACL only allow restricting by user/extension.

Any idea/suggestion on how to configure granular permissions for FreePBX modules?

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@NDSiouxFan wrote:

I have five FreePBX servers, all running FBPX 13.0.192.9 with all module updates. My DISA quit working on all servers. When I dial the extension tied to the DISA, it asks for the password as normal. It says "Password Incorrect" with the correct password. I have reset the DISA password, but no improvement. I guess my next step would be to delete that DISA record, and set up a new one. Has anyone else seen this?

It has to be a system issue with all of my servers exhibiting the same behavior. DISA was working in the past.

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@StaceyS wrote:

After updating some modules this morning, we are experiencing a serious issue with our FreePBX system.
Calls (whether incoming or outgoing, external or internal) are immediately placed on hold, and there doesn't appear to be any way to recover them.

We have 8 Polycom VVX 500 phones, and when the call is ringing, everything looks good, but as soon as the call is answered, the display changes from displaying the called number to "Held:XXX-XXX-XXXX", and the line shows the call on hold. X-Lite does the same.

The hold appears to be coming from something besides the device. If I establish a call between my Polycom to my cellphone, the Polycom displays "Held:CELLPHONE", and there is silence on my cellphone. If I place the call on hold from my Polycom, I hear our FreePBX's MOH through my cellphone.

Using XLite is the same, although XLite reports "On hold by other party." Again, if I place that call on hold from X-Lite, I immediately hear the MOH through my cellphone.

We can call out, but get silence once the connection is established. We can receive calls, but get silence on connection. We can call the Voicemail system, but immediately get silence. The phones appear as if the calls are on hold from the other side. If we call the voicemail, the call eventually disconnects, as there apparently is no activity, even if we dial our password.

Any help would be appreciated.

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@matt_dbp wrote:

Is it possible to pull call recordings over a Rest Interface or AMI?

I had a look at https://wiki.freepbx.org/display/FPG/REST+API but it appears this does not work from remote origins.

Thanks,

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