@Nelsonasp wrote:

Hello my pickup calls funcion doesn´t work.
I have already put all extensions in the same call group and the same pickup group, and still doesn´t work.

I have also try to change de pickup feature code *8 to other one and still doesn´t work.
I have already reeboot my server after all changes and still doesn´t work.

I´m an Help Desk supervisor, and i realy need to make it works

Can somebody help me please

Thanks in advance.

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@Itsm wrote:

Hello,
We are currently using freepbx,
And I would like to have a full call-center solution that would have the following features;

1. Extension login/logout - or an option to mark an extension on break (In case the employee take a lunch break etc..)
2. dashboard showing total number concurrent calls and alike in real time
3. Distributed dialer - based on a queue that will be populated by an API (the dialer should take in consideration if an employee is on break/logout, in conversation and such, configurable working time)
4. The ability to integrate with Microsoft Dynamics CRM

Just to be obvious here.. I'm looking for a payed supported solution, and any custom/additional developments according to our needs will of course be paid additionally.

So please guys, share your experience

Thank you.

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@sentinelace wrote:

I successfully imported these extensions from a csv, but now all the destinations are in error. How do you fix this?

error.PNG1234x118 6.81 KB

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@zirophyz wrote:

Hi all,

Noticed that OSS EPM doesn't download any T4X packages in the FreePBX UI - however, upon looking at /var/www/html/admin/modules/_ep_phone_modules/endpoint/yealinkv70/t4x I can see that there are configs, and json files to (I guess) support these phones.

Is there something I've missed to get these to show up under the FreePBX UI? I checked under the YealinkV70 and Yealink/Dreamwave packages.

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@igloo wrote:

Hello, I have FreePBX 13.0.187 host with Backup module 13.0.26
Backup module stopped to write backups (local storage). Tested with default backup.
Backup information appears:
- in mysql db
- backup.php
- utility.function.php
- Cron.class.php
In Cron.class.php var_dump shows cron job command (@daily ID=freepbx_backup_1 /var/lib/asterisk/bin/backup.php --id=1) the last time in function "add", before "foreach ($addArray as $add) {"
I see that modification time of /var/spool/cron/asterisk changes every time when i modify and save Backup schedule (daily to weekly and back). But job @daily ID=freepbx_backup_1 /var/lib/asterisk/bin/backup.php --id=1 doesn't appears inside.
I tried to remove Backups and create new one, but unsuccessfully (manual run of Backup works correctly).

I have another host with a similar setup, all works correctly too.
Any ideas what happens?

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@atilla_666 wrote:

Hi. Our customer gave me a task and I don't know solution. I want to block a range of numbers (1234500-99) and apply this blacklist rule to an existing inbound route. There are a lot of inbound routes in my system, but I want to apply this only for one inbound route. How can I do this? Please, help me.

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@woodpecker505 wrote:

I've upgraded a 2.11 system to FreePBX 13 and have a problem with installation of the unsupported bosssecretary module. I'm not a programmer - so I'm kindly asking for help how to try fixing this - does this module need recompiling from source or does it try to access invalid variables?

It gives following error during installation via Module Admin

 Exception

Can not modify bosssecretary_group.id_group column::
/var/www/html/admin/libraries/utility.functions.php

	if(is_object($extended_text) && method_exists($extended_text,"getMessage")) {
		$e = $extended_text;
		$extended_text = htmlentities($e->getMessage());
		$code = $e->getCode();
		throw new \Exception($text . "::" . $extended_text,$code,$e);
	} else {
		$extended_text = htmlentities($extended_text);
		throw new \Exception($text . "::" . $extended_text);
	}
}


further info:

Exception
/­var/­www/­html/­admin/­libraries/­utility.functions.php205
5. die_freepbx
/­var/­www/­html/­admin/­modules/­bosssecretary/­install.php104
4. include_once
/­var/­www/­html/­admin/­libraries/­modulefunctions.class.php2461
3. module_functions _doinclude
/­var/­www/­html/­admin/­libraries/­modulefunctions.class.php2413
2. module_functions _runscripts
/­var/­www/­html/­admin/­libraries/­modulefunctions.class.php1963
1. module_functions install
/­var/­www/­html/­admin/­page.modules.php284
0. include
/­var/­www/­html/­admin/­config.php391

and also:

Server/Request Data
SCRIPT_URL 	/admin/config.php
SCRIPT_URI 	http://10.134.63.4/admin/config.php
HTACCESS 	on
HTTP_HOST 	10.134.63.4
HTTP_USER_AGENT 	Mozilla/5.0 (Windows NT 10.0; WOW64; rv:50.0) Gecko/20100101 Firefox/50.0
HTTP_ACCEPT 	text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8
HTTP_ACCEPT_LANGUAGE 	en-US,en;q=0.5
HTTP_ACCEPT_ENCODING 	gzip, deflate
HTTP_REFERER 	http://10.134.63.4/admin/config.php?display=modules
HTTP_COOKIE 	lang=en_US; guielToggle=%7B%22extensions%23UserManagerSettings%22%3Atrue%2C%22extensions%23VmXLocater%22%3Afalse%2C%22extensions%23iSymphonySettings%22%3Afalse%2C%22extensions%23ExtensionRouting%22%3Atrue%7D; searchHide=1; bannerMessages=%5B%22b21e830264d9368d9e9d6dfe2831c170f3b051bc%22%5D; local-type=upload; PHPSESSID=vpulkni819kop5prgfikc0qrg5; __utma=49849254.788681124.1481887311.1482093100.1482137719.9; __utmz=49849254.1481887311.1.1.utmcsr=(direct)|utmccn=(direct)|utmcmd=(none); sid=9c7b8071a10b6868df6d6bd64a509b70; __utmc=49849254; testing=1; _ga=GA1.1.788681124.1481887311
HTTP_CONNECTION 	keep-alive
CONTENT_LENGTH 	0
PATH 	/sbin:/usr/sbin:/bin:/usr/bin
SERVER_SIGNATURE 	<address>Apache/2.2.15 (CentOS) Server at 10.134.63.4 Port 80</address>
SERVER_SOFTWARE 	Apache/2.2.15 (CentOS)
SERVER_NAME 	10.134.63.4
SERVER_ADDR 	10.134.63.4
SERVER_PORT 	80
REMOTE_ADDR 	10.134.63.21
DOCUMENT_ROOT 	/var/www/html
SERVER_ADMIN 	root@localhost
SCRIPT_FILENAME 	/var/www/html/admin/config.php
REMOTE_PORT 	58305
GATEWAY_INTERFACE 	CGI/1.1
SERVER_PROTOCOL 	HTTP/1.1
REQUEST_METHOD 	POST
QUERY_STRING 	display=modules&action=process&quietmode=1&online=1&modules%5Bbosssecretary%5D%5Baction%5D=install&modules%5Bbosssecretary%5D%5Btrack%5D=stable
REQUEST_URI 	/admin/config.php?display=modules&action=process&quietmode=1&online=1&modules%5Bbosssecretary%5D%5Baction%5D=install&modules%5Bbosssecretary%5D%5Btrack%5D=stable
SCRIPT_NAME 	/admin/config.php
PHP_SELF 	/admin/config.php
REQUEST_TIME 	1482226049
GET Data
display 	modules
action 	process
quietmode 	1
online 	1
modules 	Array ( [bosssecretary] => Array ( [action] => install [track] => stable ) )
POST Data
empty
Files
empty
Cookies
lang 	en_US
guielToggle 	{"extensions#UserManagerSettings":true,"extensions#VmXLocater":false,"extensions#iSymphonySettings":false,"extensions#ExtensionRouting":true}
searchHide 	1
bannerMessages 	["b21e830264d9368d9e9d6dfe2831c170f3b051bc"]
local-type 	upload
PHPSESSID 	vpulkni819kop5prgfikc0qrg5
__utma 	49849254.788681124.1481887311.1482093100.1482137719.9
__utmz 	49849254.1481887311.1.1.utmcsr=(direct)|utmccn=(direct)|utmcmd=(none)
sid 	9c7b8071a10b6868df6d6bd64a509b70
__utmc 	49849254
testing 	1
_ga 	GA1.1.788681124.1481887311
Session
module_name 	modules
module_page 	modules
AMP_user 	ampuser Object ( [username] => L3admin [id] => [password:ampuser:private] => 4d15abd1363123308f93cdda31033ac4b17e1f8a [extension_high:ampuser:private] => [extension_low:ampuser:private] => [sections:ampuser:private] => Array ( [0] => * ) [mode:ampuser:private] => database [opmode:ampuser:private] => [_deptname] => [_lastactivity] => 1482226049 )
calculated_max_calls 	0
netstats 	Array ( [eth0] => Array ( [tx] => Array ( [1481887385] => 609467 [1481887393] => 613587 ) [rx] => Array ( [1481887385] => 63143 [1481887393] => 65383 ) ) )
backup_restore_path 	/var/spool/asterisk/tmp/backuptmp-suser-1481889055-20161214-233128-1481754688-225202995.tgz
backup_restore_data 	Array ( [settings] => true [files] => Array ( [0] => /etc/asterisk [1] => /var/www ) )
DASHBOARD_FREEPBX_BRAND 	FreePBX
langdirection 	ltr
UCP_isMobile 	
UCP_isTablet 	
UCP_login_token 	40b71714dcfacbf1188636b4221dc41a
Environment Variables
empty
Registered Handlers

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@batchen wrote:

Hi,

we have -
Asterisk 13.10.0
freepbx-13.0.101

SHMZ release 6.6 (Final)

is there any wayto let user "Play" at the browser the record from the CDR report but not Download them ?

Thanks

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@batchen wrote:

Hi,

Details:
Asterisk 13.10.0
freepbx-13.0.101

SHMZ release 6.6 (Final)

i have a script that help me install new module to all of our severs.
the script :
1) transferring new module file to - /var/www/html/admin/modules/
2) setting file premissions - chown asterisk. && chmod 775
3) amportal a ma install $MODULENAME
(at this point i can see the new module at module admin in freepbx but as not installed)
4) /var/lib/asterisk/bin/module_admin install $MODULENAME
(this making the installing throw CLI )

my issue is- if i have a big server with a lot of recording for some reason the command start to set permission to all recordings! its taking forever.

can i skip it somehow ?
this is the output:

!!!!amportal is depreciated. Please use fwconsole!!!!
forwarding all commands to 'fwconsole'
Taking too long? Customize the chown command, See http://wiki.freepbx.org/display/FOP/FreePBX+Chown+Conf
Setting Permissions...
	Collecting Files...Done
 39738/39738 [============================] 100%
Finished setting permissions

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@hiro2016 wrote:

hello Everyone,

it may not properly applicable but please allow me to ask some help.
I have purchased PBXAct U and now trying to figure out CID lookup using Phonebook Module.
i tried fwconsole ma listonline but it never return anything like Phonebook .

the manual of Freepbx 13 on wiki says there is menu ander Admin.. which also can not find anything like that.

does anyone have similar problem or someone has any ideas ?

thanks.

-hiro

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@lblokland wrote:

Hi,

On our FPX 13.0.190.7 installation the responsive firewall is not working.
We have only enabled CHAN_SIP Protocol on UDP Port 5060 in asterisk and in the firewall the interface (only one, eth0) is set to 'internal' ,

I can see hundreds of registration attempts in the asterisk logs, even adding the remote IP to the blacklist in the firewall won't block it.

Can anyone help me out ?

Cheers.

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@dijkstra wrote:

HI,

I have some strange problem on one of my Phone systems.
Sometimes during a call we get Jitter on a server (TCPdump on server). we see a delay in the RTP packets inbetween of 0.1 and 1.1 seconds. What causes a silence moment in a call. There is no packet loss just this strange jitter. The jitter is visible on all the legs so in comming and outgooing. it does not matter if it is a internal call or an external call. All phones are connected directly one the network no NAT or anything only the VoIP trunk is gooing through NAT. the phone system is installed on Proxmox 4.3 (KVM virt) we have multiple phone sytems running on the same server and they dont have this problem. If we start a ping to 8.8.8.8 we dont see the delay jusst a stable ping of about 7ms.
The server has 2 NIC's and about 5 static routes (to reach the diffrent sites) disk and NIC are Virtio based.

I hope someone has an idea?

Aron.

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@MC1 wrote:

Hello
I'm going to be moving my setup from a Lap Top to a FreePBX 60 Appliance. The FreePBX 60 will also include the installation of a Sangoma FXO / FXS card. I'd like to confirm the process first within this forum.

Lap Top
V 13.0.190.7
PBX Firmware = 10.13.66-17
PBX Service Pack = 1.0.0.0

Procedure - make sure the Lap Top is upgraded to the latest firmware and the modules are up to date.
- complete a "full backup" including "voice mail" and "system audio".
- transfer this onto a USB stick.
- power up the FreePBX 60, select Asterisk 13 and upgrade it to the latest firmware, service pack and modules. Hopefully "System Admin Module" comes with the appliance.
- complete a restore from the USB key
- lastly, install the Sangoma FXO / FXS card
- test and verify.

Is this the correct procedure to follow?

Thanks in advance for any help that maybe provided.

Michael

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@jhelou wrote:

Hello Community,
I am having a strange problem with a deployment of FreePBX 13.0.190.7
i have 20+ Aastra 6731i and one Cisco 508G, the aastra phones create a distorted echo only audible by the person who is talking on the phone, and only echo the person's own voice. the receiving party does not hear the echo and it is not audible on recordings, on speaker phone the problem seems to be less pronounced. Cisco phones had no problems at first but today they too have some distorted feedback. the problem is more pronounced for external calls, and it builds up over time for long conversations. I tried replacing the handset, the phone, the codecs used (i tried g711 and g722). I also tried settings: handset tx gain: -10 in the provisioning file (aastra phones). The gain parameter helped a little but the distortion is still audible and annoying.
I have exhausted the limits of my asterisk knowledge, i was thinking of replacing the catalyst PoE switch, but i don't see how this could affect call quality. any ideas would be greatly appreciated.
Thank you

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@stevet wrote:

New installation of of the FreePBX distro on our own in-house hardware. Migrated from an Elastix VPS in the cloud, partly because I thought it might solve this problem.

Starting last week, we began to have issues with outbound calls. A call is passed to our provider (Vitelity), and as soon as it's answered, it drops. This is intermittent, but seems to happen in clumps, or everything is fine for a while, and then we can't make calls for a while, rinse, repeat.

I'd recently done an major update on the Elastix system, and figured that had caused some sort of issue. Since I was planning on moving away from that system anyway, I got FreePBX set up at the end of the week and over the weekend, and we were up and running on Monday morning, but the problem persists. Different server, different network, different distro, even different phones in a couple of cases. Figured it was Vitelity, so I opened a ticket with them and after digging into their logs, they informed me that the BYE signal was coming from my end...

I've searched everything I can think of to find people who've had the same issue, and most of what I found was from years ago and is unrelated. The most likely thing I can come up with is issues with RTP/firewall, but disabling the firewall doesn't solve the problem.

I captured the sip debug output from one of these calls, which is below. Has anyone seen this recently? I've been running various * distros for almost a decade and haven't run into anything like this before that couldn't be solved by flushing the firewall rules. Even * doesn't know why it hung up, according to the debug log.

TIA

X.X.X.X = OUR PBX IP (PUBLIC, NOT BEHIND NAT)

<--- SIP read from UDP:64.2.142.216:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK32d78e67;received=X.X.X.X
From: <sip:myusername@X.X.X.X:5160>;tag=as68734862
To: <sip:15033730000@outbound.vitelity.net>
Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160
CSeq: 103 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:15033730000@64.2.142.216>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:64.2.142.216:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK32d78e67;received=X.X.X.X
From: <sip:myusername@X.X.X.X:5160>;tag=as68734862
To: <sip:15033730000@outbound.vitelity.net>;tag=as1f0fd561
Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160
CSeq: 103 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 334

v=0
o=root 18246 18246 IN IP4 64.2.142.216
s=session
c=IN IP4 64.2.142.216
t=0 0
m=audio 19544 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (11 headers 16 lines) ---
sip_route_dump: no route/path
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g723|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk), peer - audio=(ulaw|gsm|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 64.2.142.216:19544
    -- SIP/vitel-outbound-0000000d is making progress passing it to PJSIP/102-00000019

<--- SIP read from UDP:64.2.142.216:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK32d78e67;received=X.X.X.X
From: <sip:myusername@X.X.X.X:5160>;tag=as68734862
To: <sip:15033730000@outbound.vitelity.net>;tag=as1f0fd561
Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160
CSeq: 103 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 334

v=0
o=root 18246 18247 IN IP4 64.2.142.216
s=session
c=IN IP4 64.2.142.216
t=0 0
m=audio 19544 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (11 headers 16 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g723|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140|silk|silk|silk|silk), peer - audio=(ulaw|gsm|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 64.2.142.216:19544
sip_route_dump: no route/path
[2016-12-21 10:22:52] WARNING[2431][C-0000000e]: chan_sip.c:16592 __set_address_from_contact: Invalid contact uri  (missing sip: or sips:), attempting to use anyway
[2016-12-21 10:22:52] WARNING[2431][C-0000000e]: chan_sip.c:16605 __set_address_from_contact: Invalid URI: parse_uri failed to acquire hostport
Transmitting (no NAT) to 64.2.142.216:5060:
ACK sip:15033730000@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK774eb655
Max-Forwards: 70
From: <sip:myusername@X.X.X.X:5160>;tag=as68734862
To: <sip:15033730000@outbound.vitelity.net>;tag=as1f0fd561
Contact: <sip:myusername@X.X.X.X:5160>
Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160
CSeq: 103 ACK
User-Agent: FPBX-13.0.190.7(13.13.1)
Content-Length: 0

---
Reliably Transmitting (no NAT) to 64.2.142.216:5060:
BYE sip:15033730000@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK07263c34
Max-Forwards: 70
From: <sip:myusername@X.X.X.X:5160>;tag=as68734862
To: <sip:15033730000@outbound.vitelity.net>;tag=as1f0fd561
Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160
CSeq: 104 BYE
User-Agent: FPBX-13.0.190.7(13.13.1)
Proxy-Authorization: Digest username="myusername", realm="asterisk", algorithm=MD5, uri="sip:15033730000@outbound.vitelity.net", nonce="61f2cbea", response="78bab559a93fe55939070a02f664509a"
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0




---
Scheduling destruction of SIP dialog '3e78dad92b6d54a7613617515445382a@X.X.X.X:5160' in 32000 ms (Method: INVITE)
    -- SIP/vitel-outbound-0000000d answered PJSIP/102-00000019
    -- Channel SIP/vitel-outbound-0000000d joined 'simple_bridge' basic-bridge <d87be888-6633-4be3-9af7-0ed72301cf3f>
    -- Channel PJSIP/102-00000019 joined 'simple_bridge' basic-bridge <d87be888-6633-4be3-9af7-0ed72301cf3f>
    -- Channel SIP/vitel-outbound-0000000d left 'simple_bridge' basic-bridge <d87be888-6633-4be3-9af7-0ed72301cf3f>
    -- Channel PJSIP/102-00000019 left 'simple_bridge' basic-bridge <d87be888-6633-4be3-9af7-0ed72301cf3f>
  == Spawn extension (macro-dialout-trunk, s, 23) exited non-zero on 'PJSIP/102-00000019' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 5033730000, 6) exited non-zero on 'PJSIP/102-00000019'
    -- Executing [h@from-internal:1] Macro("PJSIP/102-00000019", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/102-00000019", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/102-00000019", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] Hangup("PJSIP/102-00000019", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/102-00000019' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/102-00000019'
  == MixMonitor close filestream (mixed)
  == End MixMonitor Recording PJSIP/102-00000019
Scheduling destruction of SIP dialog '3e78dad92b6d54a7613617515445382a@X.X.X.X:5160' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 64.2.142.216:5060:
BYE sip:15033730000@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK21083f8c
Max-Forwards: 70
From: <sip:myusername@X.X.X.X:5160>;tag=as68734862
To: <sip:15033730000@outbound.vitelity.net>;tag=as1f0fd561
Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160
CSeq: 105 BYE
User-Agent: FPBX-13.0.190.7(13.13.1)
Proxy-Authorization: Digest username="myusername", realm="asterisk", algorithm=MD5, uri="sip:15033730000@outbound.vitelity.net", nonce="61f2cbea", response="78bab559a93fe55939070a02f664509a"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---

<--- SIP read from UDP:64.2.142.216:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK07263c34;received=X.X.X.X
From: <sip:myusername@X.X.X.X:5160>;tag=as68734862
To: <sip:15033730000@outbound.vitelity.net>;tag=as1f0fd561
Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160
CSeq: 104 BYE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:64.2.142.216:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP X.X.X.X:5160;branch=z9hG4bK21083f8c;received=X.X.X.X
From: <sip:myusername@X.X.X.X:5160>;tag=as68734862
To: <sip:15033730000@outbound.vitelity.net>;tag=as1f0fd561
Call-ID: 3e78dad92b6d54a7613617515445382a@X.X.X.X:5160
CSeq: 105 BYE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '3e78dad92b6d54a7613617515445382a@X.X.X.X:5160' Method: INVITE

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@cramermp wrote:

Running FreePBX 13 distro. Using commercial Endpoint Manager for provisioning.

Phone is a Yealink T32G. Latest firmware available via Endpoint Manager (1.16 release).

Upon hitting the *97 message key on the phone, we hear "You have 2 new messages and 1 old message." It then hangs up. I've deleted the extension and recreated from our template, and reset the phone to factory and upgraded firmware to the latest release in the EPM. The voicemail system is working for other users.

The logs are not showing me anything; full call log is posted in the pastebin below.
http://pastebin.com/FwQx6VYK

What's going on?

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@MSchub wrote:

I have 2 FreePBX boxes at 2 sites. I have an option in the IVR of PBX-A to dial an extension at PBX-B using a "Misc Destination" and entering that extension number. I would rather have an option at PBX-A that routes the call to the IVR at PBX-B (and vice versa). To do that using the method I'm currently using, I would need a way to direct-dial into the IVR at PBX-B, but I don't see that as a possibility. Is this the best way of achieving this outcome, or is there a better method?

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@paradonym wrote:

Installed FreePBX on Debian using - wiki.freepbx.org/display/FOP/Installing+FreePBX+13+on+Debian+8.1
I'm not using the Distro.

After installation the FreePBX Dashboard instantly warns "The file format sln48 is not supported on your system".

That's probably the reason why I can't play or upload sounds for "Music on Hold".

No such command 'file convert /var/lib/asterisk/moh/macroform-cold_day.alaw /var/spool/asterisk/tmp/temp.1482406486674.wav' (type 'core show help file convert' for other possible commands)
File:/var/www/html/admin/libraries/media/Media/Driver/Drivers/AsteriskShell.php:157

The module res_convert which should provide asterisk's ability to execute "file convert" is loaded inside asterisk and a installed sox package on Debian works properly.

How do I install the sln48 support? Is it specific for asterisk - a missing module to load or a missing package need to be installed on Debian?

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@thkavouras wrote:

Hello all,

I was wondering if anybody knows the solution for this.
I have an OpenVox A400P04 with 4 FXO modules and Yealink T29G handsets.

I want to program 4 DSS keys on the Yealinks to act as follows:
* a) As BLF in order to see the status of those channels individually &
* b) As line selectors (in order to press them and make a call using those channels/lines)

Note that I also have 4 prefixes programmed in the dialplan in order to route outgoing calls via each channel (81, 82, 83, 84). Those work fine if typed manually.

Note also that although the phone supports several SIP accounts I only have one per phone.

First of all, I assume the above is possible. Correct?
I am not sure how to do it however despite the search I have done around the net.

Any help is much appreciated.
K

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@mvogel4949 wrote:

I have a system running FreePBX 13 and Asterisk 13. The system can't resolve urls, if you try to put in a url for a sip trunk asterisk chokes and dies. If I press the detect external ip button, it can not retrieve that information. If I put in an IP for the sip trunks everything is golden call wise but I can't update my modules without dns.

I'm sitting behind a CISCO router/firewall. The system is on a 10.0.10.0/24 and the data is on a 10.0.20.0/24.

My DNS in System Admin Pro is 127.0.0.1, 8.8.8.8 and 8.8.4.4

Any ideas what might be going on network wise? Thanks

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