@frankb wrote:

I have a queue with static agents. These static agents do not have voicemail. These agents at times can have calls transferred to them directly, when this happens if they don't answer or if they are already on the phone we want the caller to have other options without having to hang up and call back in.

I created an IVR with a greeting and 2 options. The greeting tells the caller that this extension does not have voicemail press 1 to terminate the call or press 2 to be transferred to an attendant.

I then go to the static agents extension setup and I assign my new IVR to the Optional Destinations ... No Answer / Busy / Not Reachable.

The problem is that when this is configured and the static agent is on the phone queue callers are sent immediately to this extensions IVR. I would have expected the queue to see that the extension was in use and skip the extension.

If the static agent had voicemail and the same scenario occurred where this static agents phone was busy the caller would not be sent directly to their voicemail?? So why if I change Optional Destinations > Busy > "Busy Voicemail If Enabled" to something else that the queue acts differently?

Any suggestions on what I can do to accomplish this successfully??

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@fph wrote:

I did an update like I always do and now I can't connect to asterisk and got the following messages:

1. Exception
   /­var/­www/­html/­admin/­libraries/­BMO/­Hooks.class.php255

2. FreePBX\Hooks processHooks
   /­var/­www/­html/­admin/­modules/­sysadmin/­Sysadmin.class.php920

3. FreePBX\modules\Sysadmin getModuleLicenseInformation
   /­var/­www/­html/­admin/­modules/­sysadmin/­functions.inc/­license.php73

4. sysadmin_is_module_licensed
   /­var/­www/­html/­admin/­page.modules.php847

5. include
   /­var/­www/­html/­admin/­config.php385

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@Trezeguet17 wrote:

hi peopple , recently i have been presenting trouble with some log file that are literally eating my disk space, these files are locate in var/log/ , ad the size of them increased over 100 gb , whren y type cat to seee what are inside of these files it show me this:

php: /var/lib/asterisk/agi-bin/phpagi-asmanager.php[227]: feof(): supplied argument is not a valid stream resource

what could be a possible solution to stop that error, is it a bug? is this something dangeroues? i already look for solutions like modify asmanager.php to stop this error but i dont know ... please help me

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@tonyclewis wrote:

We are currently looking for technical oriented people to join the Sangoma Family in the UK/London area. This role would be part of our Support and Pre Sales Engineering team. Experience with FreePBX setup and installation is a must and use of other Sangoma products such as SBC, Gateways and Cards is a bonus but not required.

Here is your chance to join the FreePBX team and be on the front lines. See our job posting here. http://www.sangoma.com/company/careers/uk-customer-service-engineer/

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@Abayomi wrote:

Hello,

i just installed Asterisknow, all extension working fine but i notice i can not do video call.

please how can i activate this? i have tried adding h246 under each user extension---- allow codex but immediately i do that i will not be able to call out again from that very extension.

please help

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@munozj wrote:

On User Management 13.0.76.20 I'm having trouble adding an Active Directory Directory. I have been able to add an OpenLDAP directory but when i use the same creds for AD and hit submit the submit button goes white (pressed) but it doesn't appear to actually submit the form. tried on Safari and Chrome each with the same results. No error message or spinning browser showing any activity. Same behavior for the Legacy AD and non-legacy AD

I tested this on two different systems, a Distro 13 and a Non-Distro 13 and both behaved the same. It's almost as if the information i'm submitting isn't complete but there isn't any message as to what's missing.

I upgraded usermanager on system had had previously attached to the legacy AD and that imported properly as a legacy system but wouldn't' let me submit a new non-legacy ad setting. Same behavior, the SUBMIT button wouldn't submit.

I entered my complete credentials in the directory settings and left the rest as default. Is there maybe one of these empty fields that is required?

Screen Shot 2017-06-29 at 11.21.10 AM.png1198x912 89.4 KB

Screen Shot 2017-06-29 at 11.21.27 AM.png1203x858 79.6 KB

I opened an issue https://issues.freepbx.org/browse/FREEPBX-15204 but it was closed saying that it worked for them so i'm pretty sure i'm doing something wrong but can't seam to figure it out yet. I've tried filling in all the fields and different variations and even different AD networks all with the same results.

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@l0b0 wrote:

Hey guys,

I just installed Centos7, FreePBX 13 and asterisk 13 for the first time, everything installed good. I followed this guide: https://www.powerpbx.org/content/asterisk-freepbx-install-guide-centos-v7-asterisk-v13-freepbx-v13

But I cannot access the CLI, Sometimes I get this error "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" when I run asterisk -r, but I keep trying then I am able to log in to the console. Once logged in it immediately disconnects. The file does exist and I have checked most files and they are all under asterisk.asterisk. What did I miss in the config?

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@rnrstar wrote:

I've got a UC 40 running PBXact 13.0.190.19. I've also purchased the Call Recordings module. When I select the year, it's OK. Selecting the month is usually OK but when selecting the day I frequently get the following error message as a popup in the upper right corner of my browser.

DateTime::__construct(): Failed to parse time string (1970-01-01 30:50) at position 11 (3): Unexpected character
File:/var/www/html/admin/modules/recording_report/Recording_report.class.php:223

Sometimes the wave files will be listed but more frequently it responds with "no matching records."

Any suggestions?

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@RevGen wrote:

Upgraded this week to v10.13.66-20, but getting an error when I try to send an email to a selected user "Undefined index: directory File:/var/www/html/admin/modules/userman/Userman.class.php:2849 Line in question is

$usettings = $this->getAuthAllPermissions($request['directory']);

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@enrica_r wrote:

We have announcements in english and german. The extension's language is defined either english or german. We aren't alway sure if all phone users speaks german. That's why I want to play for german extension the announcement order "german - english" and for english extension "english - german".

I tried to upload for english first the english file and second the german file. For german I want to upload first german then english. This isn't possible. FreePBX GUI doesn't accept different file order per language.

Then I tried to load for english two files name "xxx-en-en.wav" and xxx-en-de.wav" and for german "xxx-de-de.wav" and "xxx-de-en.wav". Two files are always red in one language. Also in folder "sounds//custom" are the two corresponding files. But IVR announcement play all four files in each language. It seems that language isn't really separated.

It works only if I concatenate the two annoucements en/de in one file and upload it set to english. Afterwards I concatenate de/en in SAME filename and upload it set to english. This isn't obvious and elegant. Could you check this procedure and accept different orders of file per language.

Thank you very much.

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@cw1972 wrote:

Hi

I came in to the office this morning to find several extensions across several different site showing as UNREACHABLE.

Out of 120 extension over 10 sites around 15 of them were like this

When I found them on the network the were in a REGISTER FAILED state, a reboot of the device didn't help, a reboot of the router (which normally resolves the problem) didn't help, all of these troubled extension I had to do a factory reset and reconfigure them.

ALL of them picked up a new DHCP address and the reset which makes me think that they had come to the end of their lease but had not picked up a new P address.
After manual resetting them and reconfiguring them they are now working fine again.

Has anyone experience anything like this before? I have no idea where to even starting troubleshooting to find what really may have happened.

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@Abayomi wrote:

Dear All,

My AsteriskNOW freePBX is not registering on PJSIP with X-lite.

all config seems ok

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@cullenl wrote:

I have a FreePBX machine behind NAT. Trying to get a remote, NAT'ed TLS endpoint working. Despite forcing rport and all other NAT settings, Asterisk sends audio to the local address of the device. Below is a capture and some relevant configurations. Let me know if you'd like to see anything else:

<--- Received SIP request (985 bytes) from TLS:63.226.155.94:42460 --->
INVITE sips:*43@pbx.sk.user.domain.us:5161 SIP/2.0
Via: SIP/2.0/TLS 10.4.20.61;branch=z9hG4bK9c87554024e84194c;rport
Max-Forwards: 70
From: "2174" <sips:2174@pbx.sk.user.domain.us:5161>;tag=2660231b40
To: <sips:*43@pbx.sk.user.domain.us:5161>
Call-ID: f4f54805eac9f55d
CSeq: 1096918606 INVITE
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH
Allow-Events: aastra-xml, vdp-session, talk, hold, conference, LocalModeStatus
Contact: "2174" <sips:2174@63.226.155.94:35517>
Supported: path, 100rel, replaces
User-Agent: Aastra 6865i/4.3.0.1052
Content-Type: application/sdp
Content-Length: 314

v=0
o=MxSIP 0 1 IN IP4 10.4.20.61
s=SIP Call
c=IN IP4 10.4.20.61
t=0 0
m=audio 3000 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:OjtKRi4iSElpPDtzT01AP1BgMU1GLGBsLV99WEwz
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<--- Transmitting SIP response (505 bytes) to TLS:63.226.155.94:42460 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 10.4.20.61;rport=42460;received=63.226.155.94;branch=z9hG4bK9c87554024e84194c
Call-ID: f4f54805eac9f55d
From: "2174" <sips:2174@pbx.sk.user.domain.us>;tag=2660231b40
To: <sips:*43@pbx.sk.user.domain.us>;tag=z9hG4bK9c87554024e84194c
CSeq: 1096918606 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1499124704/e1b5bac8cd1ab11b9730ff072481bc24",opaque="223a5fca6dd9e2c3",algorithm=md5,qop="auth"
Server: FPBX-13.0.192.9(13.16.0)
Content-Length:  0


<--- Received SIP request (373 bytes) from TLS:63.226.155.94:42460 --->
ACK sips:*43@pbx.sk.user.domain.us:5161 SIP/2.0
Via: SIP/2.0/TLS 10.4.20.61;branch=z9hG4bK9c87554024e84194c;rport
Max-Forwards: 70
From: "2174" <sips:2174@pbx.sk.user.domain.us:5161>;tag=2660231b40
To: <sips:*43@pbx.sk.user.domain.us>;tag=z9hG4bK9c87554024e84194c
Call-ID: f4f54805eac9f55d
CSeq: 1096918606 ACK
User-Agent: Aastra 6865i/4.3.0.1052
Content-Length: 0


<--- Received SIP request (1257 bytes) from TLS:63.226.155.94:42460 --->
INVITE sips:*43@pbx.sk.user.domain.us:5161 SIP/2.0
Via: SIP/2.0/TLS 10.4.20.61;branch=z9hG4bK106dfa1bb9e9924ab;rport
Max-Forwards: 70
From: "2174" <sips:2174@pbx.sk.user.domain.us:5161>;tag=2660231b40
To: <sips:*43@pbx.sk.user.domain.us:5161>
Call-ID: f4f54805eac9f55d
CSeq: 1096918607 INVITE
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO, PUBLISH
Allow-Events: aastra-xml, vdp-session, talk, hold, conference, LocalModeStatus
Authorization: Digest username="2174",realm="asterisk",nonce="1499124704/e1b5bac8cd1ab11b9730ff072481bc24",uri="sips:*43@pbx.sk.user.domain.us:5161",response="b691d08a2c1ca692b8a9b4b24a599f21",algorithm=md5,opaque="223a5fca6dd9e2c3",qop=auth,cnonce="5a6b506b",nc=00000001
Contact: "2174" <sips:2174@63.226.155.94:35517>
Supported: path, 100rel, replaces
User-Agent: Aastra 6865i/4.3.0.1052
Content-Type: application/sdp
Content-Length: 314

v=0
o=MxSIP 0 1 IN IP4 10.4.20.61
s=SIP Call
c=IN IP4 10.4.20.61
t=0 0
m=audio 3000 RTP/SAVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:OjtKRi4iSElpPDtzT01AP1BgMU1GLGBsLV99WEwz
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

  == Setting global variable 'SIPDOMAIN' to 'pbx.sk.user.domain.us'
<--- Transmitting SIP response (323 bytes) to TLS:63.226.155.94:42460 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 10.4.20.61;rport=42460;received=63.226.155.94;branch=z9hG4bK106dfa1bb9e9924ab
Call-ID: f4f54805eac9f55d
From: "2174" <sips:2174@pbx.sk.user.domain.us>;tag=2660231b40
To: <sips:*43@pbx.sk.user.domain.us>
CSeq: 1096918607 INVITE
Server: FPBX-13.0.192.9(13.16.0)
Content-Length:  0


    -- Executing [*43@from-internal:1] Set("PJSIP/2174-00000002", "CONNECTEDLINE(name-charset,i)=utf8") in new stack
    -- Executing [*43@from-internal:2] Set("PJSIP/2174-00000002", "CONNECTEDLINE(name,i)=Echo Test") in new stack
    -- Executing [*43@from-internal:3] Set("PJSIP/2174-00000002", "CONNECTEDLINE(num,i)=*43") in new stack
    -- Executing [*43@from-internal:4] Answer("PJSIP/2174-00000002", "") in new stack
<--- Transmitting SIP response (968 bytes) to TLS:63.226.155.94:42460 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.4.20.61;rport=42460;received=63.226.155.94;branch=z9hG4bK106dfa1bb9e9924ab
Call-ID: f4f54805eac9f55d
From: "2174" <sips:2174@pbx.sk.user.domain.us>;tag=2660231b40
To: <sips:*43@pbx.sk.user.domain.us>;tag=f1f5ddd2-c75d-4e23-87a5-1dcb9188d3c8
CSeq: 1096918607 INVITE
Server: FPBX-13.0.192.9(13.16.0)
Contact: <sips:208.22.189.243:5161;transport=TLS>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: "Echo Test" <sips:*43@pbx.sk.user.domain.us>
Content-Type: application/sdp
Content-Length:   298

v=0
o=- 0 3 IN IP4 10.9.2.20
s=Asterisk
c=IN IP4 10.9.2.20
t=0 0
m=audio 13320 RTP/SAVP 0 101
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:E+It536b8vzTqNK0QReGOMxM1265vog3SHZOl29Y
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (656 bytes) from TLS:63.226.155.94:42460 --->
ACK sips:208.22.189.243:5161;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 10.4.20.61;branch=z9hG4bK862f8fbfa1f02b82f;rport
Max-Forwards: 70
From: "2174" <sips:2174@pbx.sk.user.domain.us>;tag=2660231b40
To: <sips:*43@pbx.sk.user.domain.us>;tag=f1f5ddd2-c75d-4e23-87a5-1dcb9188d3c8
Call-ID: f4f54805eac9f55d
CSeq: 1096918607 ACK
Authorization: Digest username="2174",realm="asterisk",nonce="1499124704/e1b5bac8cd1ab11b9730ff072481bc24",uri="sips:*43@pbx.sk.user.domain.us:5161",response="b691d08a2c1ca692b8a9b4b24a599f21",algorithm=md5,opaque="223a5fca6dd9e2c3",qop=auth,cnonce="5a6b506b",nc=00000001
User-Agent: Aastra 6865i/4.3.0.1052
Content-Length: 0


    -- Executing [*43@from-internal:5] Macro("PJSIP/2174-00000002", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("PJSIP/2174-00000002", "TOUCH_MONITOR=1499124705.16") in new stack
    -- Executing [s@macro-user-callerid:2] Set("PJSIP/2174-00000002", "AMPUSER=2174") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("PJSIP/2174-00000002", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("PJSIP/2174-00000002", "1?Set(REALCALLERIDNUM=2174)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("PJSIP/2174-00000002", "AMPUSER=2174") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("PJSIP/2174-00000002", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("PJSIP/2174-00000002", "AMPUSERCIDNAME=Alex Trebek") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("PJSIP/2174-00000002", "0?report") in new stack
    -- Executing [s@macro-user-callerid:9] Set("PJSIP/2174-00000002", "AMPUSERCID=2174") in new stack
    -- Executing [s@macro-user-callerid:10] Set("PJSIP/2174-00000002", "__DIAL_OPTIONS=Ttr") in new stack
    -- Executing [s@macro-user-callerid:11] Set("PJSIP/2174-00000002", "CALLERID(all)="Alex Trebek" <2174>") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("PJSIP/2174-00000002", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:13] ExecIf("PJSIP/2174-00000002", "0?Set(GROUP(concurrency_limit)=2174)") in new stack
    -- Executing [s@macro-user-callerid:14] ExecIf("PJSIP/2174-00000002", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:15] GotoIf("PJSIP/2174-00000002", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:16] ExecIf("PJSIP/2174-00000002", "1?Set(__CALLEE_ACCOUNCODE=)") in new stack
    -- Executing [s@macro-user-callerid:17] Set("PJSIP/2174-00000002", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:18] GotoIf("PJSIP/2174-00000002", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,29)
    -- Executing [s@macro-user-callerid:29] Set("PJSIP/2174-00000002", "CALLERID(number)=2174") in new stack
    -- Executing [s@macro-user-callerid:30] Set("PJSIP/2174-00000002", "CALLERID(name)=Alex Trebek") in new stack
    -- Executing [s@macro-user-callerid:31] GotoIf("PJSIP/2174-00000002", "0?cnum") in new stack
    -- Executing [s@macro-user-callerid:32] Set("PJSIP/2174-00000002", "CDR(cnam)=Alex Trebek") in new stack
    -- Executing [s@macro-user-callerid:33] Set("PJSIP/2174-00000002", "CDR(cnum)=2174") in new stack
    -- Executing [s@macro-user-callerid:34] Set("PJSIP/2174-00000002", "CHANNEL(language)=en") in new stack
    -- Executing [*43@from-internal:6] Wait("PJSIP/2174-00000002", "1") in new stack
    -- Executing [*43@from-internal:7] BackGround("PJSIP/2174-00000002", "demo-echotest,,,app-echo-test-echo") in new stack
    -- <PJSIP/2174-00000002> Playing 'demo-echotest.ulaw' (language 'en')
<--- Received SIP request (660 bytes) from TLS:63.226.155.94:42460 --->
BYE sips:208.22.189.243:5161;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 10.4.20.61;branch=z9hG4bK6207742a8a15b512a;rport
Max-Forwards: 70
From: "2174" <sips:2174@pbx.sk.user.domain.us>;tag=2660231b40
To: <sips:*43@pbx.sk.user.domain.us>;tag=f1f5ddd2-c75d-4e23-87a5-1dcb9188d3c8
Call-ID: f4f54805eac9f55d
CSeq: 1096918608 BYE
Authorization: Digest username="2174",realm="asterisk",nonce="1499124704/e1b5bac8cd1ab11b9730ff072481bc24",uri="sips:208.22.189.243:5161;transport=TLS",response="4d7eab00231e3dec60c38bc2506fe7b5",algorithm=md5,opaque="223a5fca6dd9e2c3",qop=auth,cnonce="5a6b506b",nc=00000002
User-Agent: Aastra 6865i/4.3.0.1052
Content-Length: 0


<--- Transmitting SIP response (357 bytes) to TLS:63.226.155.94:42460 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.4.20.61;rport=42460;received=63.226.155.94;branch=z9hG4bK6207742a8a15b512a
Call-ID: f4f54805eac9f55d
From: "2174" <sips:2174@pbx.sk.user.domain.us>;tag=2660231b40
To: <sips:*43@pbx.sk.user.domain.us>;tag=f1f5ddd2-c75d-4e23-87a5-1dcb9188d3c8
CSeq: 1096918608 BYE
Server: FPBX-13.0.192.9(13.16.0)
Content-Length:  0


  == Spawn extension (from-internal, *43, 7) exited non-zero on 'PJSIP/2174-00000002'
    -- Executing [h@from-internal:1] Macro("PJSIP/2174-00000002", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/2174-00000002", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/2174-00000002", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] Hangup("PJSIP/2174-00000002", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/2174-00000002' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/2174-00000002'

Transport:

> pjsip show transport 10.9.2.20-tls

Transport:  <TransportId........>  <Type>  <cos>  <tos>  <BindAddress....................>
==========================================================================================

Transport:  10.9.2.20-tls             tls      0      0  10.9.2.20:5161

 ParameterName              : ParameterValue
 =======================================================
 allow_reload               : true
 async_operations           : 1
 bind                       : 10.9.2.20:5161
 ca_list_file               :
 ca_list_path               :
 cert_file                  : /etc/asterisk/keys/tls.crt
 cipher                     :
 cos                        : 0
 domain                     :
 external_media_address     : 208.22.189.243
 external_signaling_address : 208.22.189.243
 external_signaling_port    : 0
 local_net                  : 10.9.2.0/255.255.255.0
 local_net                  : 10.19.2.0/255.255.255.0
 local_net                  : 10.9.102.0/255.255.255.0
 local_net                  : 192.168.1.0/255.255.255.0
 method                     : tlsv1
 password                   :
 priv_key_file              : /etc/asterisk/keys/tls.key
 protocol                   : tls
 require_client_cert        : No
 symmetric_transport        : false
 tos                        : 0
 verify_client              : No
 verify_server              : No
 websocket_write_timeout    : 100

Endpoint:

> pjsip show endpoint 2174

 Endpoint:  <Endpoint/CID.....................................>  <State.....>  <Channels.>
    I/OAuth:  <AuthId/UserName...........................................................>
        Aor:  <Aor............................................>  <MaxContact>
      Contact:  <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
  Transport:  <TransportId........>  <Type>  <cos>  <tos>  <BindAddress..................>
   Identify:  <Identify/Endpoint.........................................................>
        Match:  <criteria.........................>
    Channel:  <ChannelId......................................>  <State.....>  <Time.....>
        Exten: <DialedExten...........>  CLCID: <ConnectedLineCID.......>
==========================================================================================

 Endpoint:  2174/2174                                            Not in use    0 of inf
     InAuth:  2174-auth/2174
        Aor:  2174                                               4
      Contact:  2174/sips:2174@63.226.155.94:42460;transpo 4516433c04 Avail        68.272
   Identify:  2174-identify/2174


 ParameterName                      : ParameterValue
 =========================================================
 100rel                             : yes
 accountcode                        :
 acl                                :
 aggregate_mwi                      : true
 allow                              : (ulaw)
 allow_overlap                      : true
 allow_subscribe                    : true
 allow_transfer                     : true
 aors                               : 2174
 asymmetric_rtp_codec               : false
 auth                               : 2174-auth
 bind_rtp_to_media_address          : true
 call_group                         :
 callerid                           : "device" <2174>
 callerid_privacy                   : allowed_not_screened
 callerid_tag                       :
 connected_line_method              : invite
 contact_acl                        :
 context                            : from-internal
 cos_audio                          : 0
 cos_video                          : 0
 device_state_busy_at               : 0
 direct_media                       : true
 direct_media_glare_mitigation      : none
 direct_media_method                : invite
 disable_direct_media_on_nat        : false
 dtls_ca_file                       :
 dtls_ca_path                       :
 dtls_cert_file                     :
 dtls_cipher                        :
 dtls_fingerprint                   : SHA-256
 dtls_private_key                   :
 dtls_rekey                         : 0
 dtls_setup                         : active
 dtls_verify                        : No
 dtmf_mode                          : rfc4733
 fax_detect                         : false
 fax_detect_timeout                 : 0
 force_avp                          : false
 force_rport                        : true
 from_domain                        :
 from_user                          :
 g726_non_standard                  : false
 ice_support                        : false
 identify_by                        : username
 inband_progress                    : false
 language                           : en
 mailboxes                          : 2174@device
 media_address                      :
 media_encryption                   : sdes
 media_encryption_optimistic        : true
 media_use_received_transport       : false
 message_context                    :
 moh_suggest                        : default
 mwi_from_user                      :
 mwi_subscribe_replaces_unsolicited : true
 named_call_group                   :
 named_pickup_group                 :
 one_touch_recording                : false
 outbound_auth                      :
 outbound_proxy                     :
 pickup_group                       :
 record_off_feature                 : automixmon
 record_on_feature                  : automixmon
 rewrite_contact                    : true
 rpid_immediate                     : false
 rtcp_mux                           : false
 rtp_engine                         : asterisk
 rtp_ipv6                           : false
 rtp_keepalive                      : 0
 rtp_symmetric                      : true
 rtp_timeout                        : 0
 rtp_timeout_hold                   : 0
 sdp_owner                          : -
 sdp_session                        : Asterisk
 send_diversion                     : true
 send_pai                           : true
 send_rpid                          : false
 set_var                            :
 srtp_tag_32                        : false
 sub_min_expiry                     : 0
 subscribe_context                  :
 t38_udptl                          : false
 t38_udptl_ec                       : none
 t38_udptl_ipv6                     : false
 t38_udptl_maxdatagram              : 0
 t38_udptl_nat                      : false
 timers                             : yes
 timers_min_se                      : 90
 timers_sess_expires                : 1800
 tone_zone                          :
 tos_audio                          : 0
 tos_video                          : 0
 transport                          :
 trust_id_inbound                   : true
 trust_id_outbound                  : false
 use_avpf                           : false
 use_ptime                          : false
 user_eq_phone                      : false
 voicemail_extension                :

Some things I've noticed; there is no directmedia option for PJSIP extension configuration in the GUI. The GUI saves the PJSIP endpoint config to disk with media_address=; not sure if that matters but I couldn't find documentation suggesting that it was a good thing. Removing it and reloading asterisk didn't change anything, so it might not matter either way.

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@Abayomi wrote:

Hi,

please which open souce softphone can be used on andriod mobile phone to connect with FreePBX

Do i have to make any changes on FreePbx GUI?

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@fetoa wrote:

Hi all!

I have many SIP trunks configures in the same FreePBX 13 machine. All of them are trunks of the same provider.

Well, when an inbound call is coming, all of the are shown in the CLI appearing to be calls with teh same destination:

For example, if someone call to 9XXXXXXX this is shown:

-- Executing [9XXXXXXX@from-trunk-sip-MYPROVIDER_XX:1] Set("SIP/MYPROVIDER_XX-000005c7", "GROUP()=OUT_6") in new stack
-- Executing [9XXXXXXX@from-trunk-sip-MYPROVIDER_XX:2] Goto("SIP/MYPROVIDER_XX-000005c7", "from-trunk,9XXXXXXX,1") in new stack
-- Goto (from-trunk,9XXXXXXX,1)
-- Executing [9XXXXXXX@from-trunk:1] Set("SIP/MYPROVIDER_XX-000005c7", "__DIRECTION=INBOUND") in new stack
-- Executing [9XXXXXXX@from-trunk:2] Gosub("SIP/MYPROVIDER_XX-000005c7", "sub-record-check,s,1(in,9XXXXXXX,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/MYPROVIDER_XX-000005c7", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("SIP/MYPROVIDER_XX-000005c7", "_RECSTATUS=INITIALIZED") in new stack

...and if someone call to 9ZZZZZZZ this is shown:

-- Executing [9ZZZZZZZ@from-trunk-sip-MYPROVIDER_XX:1] Set("SIP/MYPROVIDER_XX-000005cc", "GROUP()=OUT_6") in new stack
-- Executing [9ZZZZZZZ@from-trunk-sip-MYPROVIDER_XX:2] Goto("SIP/MYPROVIDER_XX-000005cc", "from-trunk,9ZZZZZZZ,1") in new stack
-- Goto (from-trunk,9ZZZZZZZ,1)
-- Executing [9ZZZZZZZ@from-trunk:1] Set("SIP/MYPROVIDER_XX-000005cc", "__DIRECTION=INBOUND") in new stack
-- Executing [9ZZZZZZZ@from-trunk:2] Gosub("SIP/MYPROVIDER_XX-000005cc", "sub-record-check,s,1(in,9ZZZZZZZ,dontcare)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/MYPROVIDER_XX-000005cc", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("SIP/MYPROVIDER_XX-000005cc", "_RECSTATUS=INITIALIZED") in new stack

It's always answering via SIP/PYPROVIDER_XX

Any idea? Thanks!

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@ronandann wrote:

We are running FreePBX 13.0.192.9 with Asterisk version 13.15.0. Phone Apps Module is version 13.0.88.5. When using a phone app on the phone I am getting an email that the IP of the phone has just been banned by Fail2Ban after
8 attempts against apache-auth on localhost. It seems that the Phone App is working like it should but why would we continue to get these messages?

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@robfantini wrote:

when trying to add 1ST user to ZULU got this:

"SQLSTATE[23000]: Integrity constraint violation: 1062 Duplicate entry 'rob-1' for key 'username_UNIQUE'"

besides zulu, a similar error or exact same happen when trying to make other changes using user manager. so it is not just a zulu issue.

the issue seemed to be caused by same login name for multiple users.

the fix here anyway was to make sure there were no duplicate user login names.

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@jessy5765 wrote:

I have a deployment that we are looking at in which they are a Resident Care facility and each room has a call button for the nurses to come to the room.

We are looking at replacing it with Sangoma S700 for the business side and Akuvox R15P for the residents. ( Big Button Phone with SOS button and 3 large speed dial's programmable from the phone itself and has a remote SOS Pendant. Great for Elderly)

So our situation is: Resident pushes Pendant putton -> Calls a RingGroup of Nurses assigned to that wing, which will have "RESIDENT CALL:Room#" in the Caller ID -> 1 minute no one picks up it Rings the Central Station for Nurses -> Still no one picks up -> Rings the Nursing Director (she will raise hell if her phone rings with this, which is what she wants. People complain about being ignored) -> Finally terminates back to Central Station indefinitely.

However we would like a software that can have all rooms listed on it and they can see who called then go over and tap it or something to confirm it was handled if noone picks up the call. Is this possible?

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@jyates01 wrote:

Looking to possibly purchase CRM module to use with SuiteCRM, question I cannot find the answer to is can this setup link an account by IVR entry? If not, best idea?

Example: I have one customer that calls from the same CID every time, but has many different accounts. Need customer to enter account number when prompted, then, when queue agent connected, CRM pop up displays info for that account. Any help greatly appreciated.

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@neuralping wrote:

Just installed FreePBX 14.0.1.1
My issue is I can make outbound call but can't hear the other party talking
When I call into the PBX it receives the call but does not route it to the phone

Need a hint as to where to start looking for the cause of the problem

Only rules I have added to the firewall was to let the internal network talk to the SIP provider.

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