@ColoradoRuss wrote:

I have a call flow that is as follows:

Someone calls in from outside

the call is routed to a virtual extension that automatically goes to followme to a second virtual extension
the call is then routed by the second virtual extension to a real extension that rings a phone

The problem I have is when the first virtual extension’s followme calls the second virtual extension, that does not call the final real extension but gives the following:

CHANUNAVAIL
Unable to create channel of type ‘VIRTUAL’ (cause 66 - Channel not implemented)

Why isnt the second virtual extension just executing its followme to the final destination.

I know - get rid of the multiple hops. If I manually do this, it works fine, but I have this these extensions created automatically to create a flexibility I want. It makes it easier to make the system have the flexibility I want and have all routing thru a similar scheme.

How can I make a virtual extension’s followme execute when called by another virtual extension’s followme to go to a third extension?

Thanks for the help!!!

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@rotary500 wrote:

More than 10 years ago I built an Asterisk pbx to run my home phone collection It consists of a Pentium 4 based computer with a 4 port analog FXO card and a single TE110p T1 PCI card connected to an Adtran TA850 channel bank with 23 analog extensions. Failing capacitors destroyed the motherboard so I built another system out of an old Pentium d computer, Installed freepbx distro instead of trying to roll my own Asterisk system again- forgot how to do it anyway. The result? I cannot get access to the pots lines through the FXO ports and the channel bank does not give dial tone or any of its extensions are active. There is no alarm on the T1 card - it has a green light. However, in the asterisk log I get this error.

[2018-12-24 01:05:15] NOTICE[2346] sig_pri.c: pri_check_event returned error 500 (Unknown error 500)
[2018-12-24 01:05:15] NOTICE[2346] chan_dahdi.c: Got DAHDI event: HDLC Abort (6) on D-channel of span 1

Not sure where to start, I remember there was a command line test for the Zaptel interfaces and have seen some info on a Dahdi tools command. Don’t know if something could have gotten changed in the Adtran itself but I plugged in a Carrier Access channel bank and had the same result.

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@ghurty wrote:

If I call somone who is on the line, it goes straight to voicemail. This is for all the extensions. Where would I change that setting?

Thank you

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@ghurty wrote:

I am trying to hook up to an existing paging system. The amp has a 500ohm input for the Tel line. Is there a way for me to a way to output to that?

Thank you

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@jimvman wrote:

Hello again everybody.
I have a Fortigate 60E firewall with Dual Internet connections for fail-over. Both Internet connections have static IP addresses. Based on some other posts, I setup a DYN DNS entry for the external IP address of my connection and used that hostname in the CHAN_SIP settings of Asterisk SIP Settings page in FreePBX.
The issue is when the primary goes down, it takes about 10 minutes so I can start hearing 2-way audio, probably because the registration to the SIP provider hasn’t updated to the new IP address yet. So after about 10 minutes things seem normal, but then it’s another 10-minutes if the connection fails back to the primary again.
Are there any timers I can change to make this process go quicker? I’m using a cron script every 5 minutes on the FreePBX host to update the DYN DNS entry, so maybe I can make that more frequent, but should I be modifying any registration timers, so maybe asterisk reloads when it fails over?

Any help would be greatly appreciated to make this fail-over quicker so phones work normally.

Thanks!

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@Hawkeye wrote:

Hi, trying to install zulu server. Followed the instructions at https://wiki.freepbx.org/display/ZU/Zulu+3+Installation

Version: FreePBX 14, 12.7.5-1807-1.sng7

fwconsole ma downloadinstall zulu

No repos specified, using: [standard,commercial] from last GUI settings

Downloading module ‘zulu’
Processing zulu
Verifying local module download…Verified
Extracting…Done
Download completed in 7 seconds
Updating tables zulu_interactions_interaction_states, zulu_softphones, zulu_tokens, zulu_interactions_contacts, zulu_interactions_interactions, zulu_interactions_owners, zulu_interactions_members, zulu_interactions_streams, zulu_interactions_stream_bodies, zulu_interactions_stream_links, zulu_interactions_stream_actions, zulu_login_tokens…Done
Installing/Updating Required Libraries. This may take a while…The following messages are ONLY FOR DEBUGGING. Ignore anything that says ‘WARN’ or is just a warning
npm WARN deprecated nomnom@1.8.1: Package no longer supported. Contact support@npmjs.com for more info.
Installed npm-cache v0.7.0
Running installation…
[npm-cache] [INFO] using /home/asterisk/.package_cache as cache directory
[npm-cache] [INFO] [composer] Dependency config file /var/www/html/admin/modules/zulu/node/composer.json does not exist. Skipping install
[npm-cache] [INFO] [npm] config file exists
[npm-cache] [INFO] [npm] cli exists
[npm-cache] [INFO] [npm] hash of /var/www/html/admin/modules/zulu/node/package.json: d6074356c76df89aff95ef2058f76228
[npm-cache] [INFO] [npm] cache exists
[npm-cache] [INFO] [npm] clearing installed dependencies at /var/www/html/admin/modules/zulu/node/node_modules
[npm-cache] [INFO] [npm] …cleared
[npm-cache] [INFO] [npm] retrieving dependencies from /home/asterisk/.package_cache/npm/5.6.0/d6074356c76df89aff95ef2058f76228.tar.gz
[npm-cache] [INFO] [bower] Dependency config file /var/www/html/admin/modules/zulu/node/bower.json does not exist. Skipping install
[npm-cache] [INFO] [npm] done extracting
[npm-cache] [INFO] successfully installed all dependencies

Finished updating libraries!
Stopping old running processes…Zulu Server is not running
Done
Start Stream Actions Migration…
Creating entries to zulu_interactions_stream_actions table…Done
Setting to NULL seen field in zulu_interactions_streams table…Done
No need to migrate XMPP
Start Public Stream Action Migration…
Creating entries to zulu_interactions_stream_actions table…Done
Starting new Zulu Process…Enabling the Asterisk ARI Server

In utility.functions.php line 207:

trying to set keyword [ENABLE_ARI] to [1] on uninitialized setting::

ma [-f|–force] [-d|–debug] [–edge] [–stable] [–color] [–skipchown] [-e|–autoenable] [–skipdisabled] [–snapshot SNAPSHOT] [–format FORMAT] [-R|–repo REPO] [-t|–tag TAG] [–onlystdout] [–willupdate] [–securityonly] [–sendemails] [–] []…

In Asterisk Rest Interface Users it displays:

The Asterisk REST Interface is Currently Disabled in Advanced Settings

In advanced settings REST and ARI are not found.

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@Hawkeye wrote:

Version: 12.7.5-1807-1.sng7, FreePBX 14.0.5.5

Ordered zulu 2-user license for 1 year on portal. Run Update Activation in FreePBX, the license does not show up.

Thanks.

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@voip2 wrote:

Hello, I am a longtime FreePBX non distro user.
Recently I built a CentOS 7 (64) server with FreePBX 14 & Asterisk 13.24.0. Using PJSIP*
Everything seems to work fine with the following exception: After setting extension and trunk and sip / PJSIP settings as well as endpoint phone to use g722 it still seems to transcode it.
Using core show channel it shows native is g722 but read and write format is slin16. When I change them back to g711 then native and read and write are all g711 like expected.

I am curious if anyone has thoughts on what could be causing this.

Thanks!

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@fropa wrote:

I use FreePBX 13 with less hardware than minimum requirements. I’ve no free HDD, maybe 50-70MBs. It works but I need more HDD until I can migrate that VPS on another, bigger one.

I’m interested, do I need /usr/src/freepbx-13.0.190.19/ content or can I delete it? It’s nearly 250MBs and it will be great to delete if it isn’t important.

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@jessy5765 wrote:

So this is kind of odd. We are hosted out of freepbxhosting.com datacenter. Everything is working really well. However when setting up the last 2 phone systems, we are using Sangoma phones, the BLF lights didnt light up right away. Everything was working with no issues otherwise.

On the first install, they didnt show up and i just sat there and fiddled with the phone system to see why. Rebooting it, re-registering extensions, nothing would work. However just randomly while fiddling they lit up on the phone i was working on. I looked over at the receptionist and asked her if her lights were on and they did in fact come on, on all the phones at the same time. This includes BLF keys and the XML-API keys both.

For the 2nd install, the same thing, however I did no fiddling with the phone system. We left for the day and came back the next morning and all lights were on, on all of the phones.

Has anyone else experienced this, what is responsible for this. The subscriptions are working so im not sure why this is happening. This is in FreePBX 14, latest updates all within the last 2 weeks.

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@superfly13 wrote:

Hi everyone,
English is not my first language so please excuse me if I over-explain or make mistakes in my writing.

I need your help.

My setup:
FreePBX 2.11.0.43
Asterisk (Ver. 11.17.1)

Currently for every user in the office I have setup a ring group(example: 605) with 2 extensions(example: 03 and 04) in it. Both phones 03 and 04 are at the users desk as one is cordless and the other one is corded - so the user can choose which one to pick up. For the same user I also have a third extension (example 02). I am forwarding extension 02 (via the phone menu) to 605(ring group) and now both 03 and 04 ring if someone diales 02. The reason I do it is because our CRM can only use (dial) extensions (not ring groups). I just can’t make it dial 605 - so I make it dial 02 and both phones 03 and 04 ring. So far, despite the fact that my solution is not elegant at all - it works.

What I would like to do is to do the forwarding of 02 -> 605 in FreePBX instead of the phone menu of 02, as now I need to keep one phone ON only for the forwarding and I can not use it for anything else. Basically I am using 3 phones per user - one(02) at the server room and 2 (03 and 04) on his desk.

So the question here is: how to forward (if possible at all) an extension to a ring group in Freepbx?

What I tried already:

• in the extension settings of 02 in the field “Dial” instead of sip/02 i put sip/605 but it did not work. It gives me busy tone if dial 02 from another extension.
• I tried the “Follow me” module and put the ring group number in the “Follow me list” of extension 02. - this did not work as well.

Thank you for your time!

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@Autourdupc wrote:

Hi all.

I’m facing an issue using a Huawei e169 dongle on my Freepbx 14.0.5.2.
Asterisk is 13.23.0.

chan-dongle is installed and working.
My dongle is recognized by asterisk.

raspbx2*CLI> dongle show device state dongle0
-------------- Status -------------
  Device                  : dongle0
  State                   : Dialing
  Audio                   : /dev/ttyUSB1
  Data                    : /dev/ttyUSB2
  Voice                   : Yes
  SMS                     : Yes
  Manufacturer            : huawei
  Model                   : E169
  Firmware                : 11.314.13.03.18
  IMEI                    : 35810xxxxxxxxxx
  IMSI                    : 20815xxxxxxxxxx
  GSM Registration Status : Registered, roaming
  RSSI                    : 21, -71 dBm
  Mode                    : GSM/GPRS
  Submode                 : EDGE
  Provider Name           : Orange F
  Location area code      : 74
  Cell ID                 : AD61
  Subscriber Number       : +3376760xxxx
  SMS Service Center      : +3369500xxxx
  Use UCS-2 encoding      : Yes
  USSD use 7 bit encoding : No
  USSD use UCS-2 decoding : Yes
  Tasks in queue          : 0
  Commands in queue       : 0
  Call Waiting            : Disabled
  Current device state    : start
  Desired device state    : start
  When change state       : now
  Calls/Channels          : 1
    Active                : 0
    Held                  : 0
    Dialing               : 0
    Alerting              : 0
    Incoming              : 0
    Waiting               : 0
    Releasing             : 0
    Initializing          : 1

When I try to send a call, I got an error…

        -- Executing [s@macro-outbound-callerid:20] ExecIf("SIP/1009-00000016", "1?Set(CALLERID(all)=07676xxxxx)") in new stack
    -- Executing [s@macro-outbound-callerid:21] ExecIf("SIP/1009-00000016", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:22] ExecIf("SIP/1009-00000016", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:23] ExecIf("SIP/1009-00000016", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:24] ExecIf("SIP/1009-00000016", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:25] Set("SIP/1009-00000016", "CDR(outbound_cnum)=0767608361") in new stack
    -- Executing [s@macro-outbound-callerid:26] Set("SIP/1009-00000016", "CDR(outbound_cnam)=") in new stack
    -- Executing [s@macro-dialout-trunk:14] GosubIf("SIP/1009-00000016", "0?sub-flp-1,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:15] Set("SIP/1009-00000016", "OUTNUM=04783xxxxx") in new stack
    -- Executing [s@macro-dialout-trunk:16] Set("SIP/1009-00000016", "custom=AMP") in new stack
    -- Executing [s@macro-dialout-trunk:17] ExecIf("SIP/1009-00000016", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)") in new stack
    -- Executing [s@macro-dialout-trunk:18] ExecIf("SIP/1009-00000016", "0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:19] Macro("SIP/1009-00000016", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1009-00000016", "") in new stack
    -- Executing [s@macro-dialout-trunk:20] GotoIf("SIP/1009-00000016", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:21] ExecIf("SIP/1009-00000016", "1?Set(CONNECTEDLINE(num,i)=0478314569)") in new stack
    -- Executing [s@macro-dialout-trunk:22] ExecIf("SIP/1009-00000016", "1?Set(CONNECTEDLINE(name,i)=CID:07676xxxxx)") in new stack
    -- Executing [s@macro-dialout-trunk:23] ExecIf("SIP/1009-00000016", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)07676xxxxx)") in new stack
    -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/1009-00000016", "1?customtrunk") in new stack
    -- Goto (macro-dialout-trunk,s,28)
    -- Executing [s@macro-dialout-trunk:28] Set("SIP/1009-00000016", "pre_num=AMP:dongle/dongle0/") in new stack
    -- Executing [s@macro-dialout-trunk:29] Set("SIP/1009-00000016", "the_num=OUTNUM") in new stack
    -- Executing [s@macro-dialout-trunk:30] Set("SIP/1009-00000016", "post_num=") in new stack
    -- Executing [s@macro-dialout-trunk:31] GotoIf("SIP/1009-00000016", "1?outnum:skipoutnum") in new stack
    -- Goto (macro-dialout-trunk,s,32)
    -- Executing [s@macro-dialout-trunk:32] Set("SIP/1009-00000016", "the_num=04783xxxxx") in new stack
    -- Executing [s@macro-dialout-trunk:33] Dial("SIP/1009-00000016", "dongle/dongle0/04783xxxxx,300,Tt") in new stack
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:34] NoOp("SIP/1009-00000016", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 44") in new stack
    -- Executing [s@macro-dialout-trunk:35] GotoIf("SIP/1009-00000016", "0?continue,1:s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/1009-00000016", "RC=44") in new stack
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/1009-00000016", "44,1") in new stack
    -- Goto (macro-dialout-trunk,44,1)
    -- Executing [44@macro-dialout-trunk:1] Goto("SIP/1009-00000016", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/1009-00000016", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 44 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:2] ExecIf("SIP/1009-00000016", "1?Set(CALLERID(number)=1009)") in new stack
    -- Executing [204783xxxxx@from-internal:6] Macro("SIP/1009-00000016", "outisbusy,") in new stack
    -- Executing [204783xxxxx@from-internal:7] Congestion("SIP/1009-00000016", "20") in new stack
  == Spawn extension (from-internal, 20478314569, 7) exited non-zero on 'SIP/1009-00000016'
    -- Executing [h@from-internal:1] Macro("SIP/1009-00000016", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1009-00000016", "1?theend") in new stack

I don’t know waht to do.
I have read so many pages on the web…

Any help would be appreciated.

Regards,
Laurent.

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@Ssergio wrote:

Hello!
Would appreciate any help.
A couple of days ago I performed a significant work on PBX:

1. updated FreePBX from 12 to 13.
2. installed VMWare tools (Debian), PBX is on VMWare 5.5.
3. changed external IP in SIP settings
4. changed firewall rulle, because of new external IP

Exactely after this, local users randomly began to face “no voice” in phones. They hang up, redial => all ok.

In CDR report it looks like this. No recording to play
Screenshots: http://joxi.ru/VrwVk6EI74GP4A

Call event log of this two calls is the same 15 events
unsuccessfull call: http://joxi.ru/8239yRdU98Q0pr
successfull call: http://joxi.ru/VrwVk6EI74Gy4A

I can’t to figure out why this happens.
I tried to SET SIP DEBUG PEER 123, but it’s difficult to catch a “no voice” situation, always loged normal call flow ))

Would appreciate any help.
Sergey.

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@toborgps wrote:

Hello all,

We switched to a new SIP Trunk provider however we can’t get inbound calls working. They gave us a pair of IP’s and basically said good luck. We successfully got outbound calls working but inbound calls still will not go through. Any ideas or thoughts would be great.

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@Nexcom wrote:

Greetings all, Hope the holidays treated everyone well!

Wanted to run something by the community, and see if anyone has any insight on a problem I have.

Site #1:
• Running two PBXs behind a sonicwall firewall. Set up NAT rules in firewall for main PBX (.19).
• 192.168.17.19 is one PBx, with 192.168.17.20 being office PBX. .19 is hooked up to SIP trunk to outside provider to provide inbound/outbound.
• .19 has a IAX trunk directly to .20. This provides outbound/inbound for office PBX (.20). Reason for this method is to hook up multiple PBX via IAX to .19 and have that be main point of contact.

Site #2: SIP trunk directly to outside provider. All traffic inbound/outbound routed through provider. 172.16.0.10. Physically located on different WAN address. Sonicwall Firewall as well.

Phones located on all three PBX can call in and out with no issues. Audio is fantastic and calls crisp.

My problem is when I use a phone located off the .20 PBX to call a phone located off the Site #2 PBX.

Phone 150 is located of PBX .19
Phone 203 is located off PBX .20
Phone 301 is located off Site #2 PBX 10

Each phone can call local and long distance no problem. Audio both ways.
When phone 203 tries to call 301 (DID) associated on Site #3 PBX, NO audio either way.
When phone 150 calls 203 3 digit dialing and DID audio is present both ways.
When 150 calls 301 DID no audio is heard either way.

Added in Site #3 with PBX 90.20 and phone 200. Sonicwall firewall, and IAX to PBX Main .19 via WAN IP and NAT. All call traffic flows from 90.20 to main PBX .19 then out SIP Trunk.

Phone 203 calls phone 200 (via DID) and audio issues persist. Both ways!
Phone 203 calls phone 301 (Via DID) and audio issues persist. Both ways!
Phone 203 calls long distance and local----perfect audio

Main provider telling me its symmetric NAT issue, and fix that and issue will be fixed. Tested this theory by trying to place call from 150 to 300 and 203. Both fail with audio issues. Call connects but no RTP pushing.

Using NAT on sonicwalls not on Freepbx. Let me know if you want some posts of anything.

Thanks for the thoughts all

J

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@dickovan wrote:

Is there a way to REQUIRE that an employee enters an account code before placing an outgoing phone call and before completing an incoming phone call?

The idea is to have a set of codes that could be used to categorize the nature of the call (i.e. new customer, customer appointment, collections, etc.) for analysis and KPIs.

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@coratec wrote:

Hi ,

i´ve been setting queues for a project with a new system.

But the surprise is that no picklist is available under IVR Break out menu (queues - caller announcements) and also on Agent announcement (queues - Timing & agent options) but works on Call confirm announce (queues - general settings).

Is there anything i can do to solve this?

Thanks in advance

FreePBX 14.0.3.1
Asterisk 13.19

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@mpforum wrote:

I have found this in my Asterisk log file and it constantly repeats.

Does this indicate that I am being hacked or just paranoid?

[2018-12-28 13:09:06] VERBOSE[3986][C-00000022] pbx.c: Executing [s@from-sip-external:7] Answer(“PJSIP/anonymous-00000022”, “”) in new stack
[2018-12-28 13:09:06] VERBOSE[3986][C-00000022] pbx.c: Executing [s@from-sip-external:8] Wait(“PJSIP/anonymous-00000022”, “2”) in new stack
[2018-12-28 13:09:06] VERBOSE[3869][C-00000020] pbx.c: Spawn extension (from-sip-external, s, 11) exited non-zero on ‘PJSIP/anonymous-00000020’
[2018-12-28 13:09:06] VERBOSE[3869][C-00000020] pbx.c: Executing [h@from-sip-external:1] Hangup(“PJSIP/anonymous-00000020”, “”) in new stack
[2018-12-28 13:09:06] VERBOSE[3869][C-00000020] pbx.c: Spawn extension (from-sip-external, h, 1) exited non-zero on ‘PJSIP/anonymous-00000020’
[2018-12-28 13:09:08] VERBOSE[3986][C-00000022] pbx.c: Executing [s@from-sip-external:9] Playback(“PJSIP/anonymous-00000022”, “ss-noservice”) in new stack
[2018-12-28 13:09:08] VERBOSE[3986][C-00000022] file.c: <PJSIP/anonymous-00000022> Playing ‘ss-noservice.ulaw’ (language ‘en’)
[2018-12-28 13:09:10] VERBOSE[3924][C-00000021] pbx.c: Executing [s@from-sip-external:10] PlayTones(“PJSIP/anonymous-00000021”, “congestion”) in new stack
[2018-12-28 13:09:10] VERBOSE[3924][C-00000021] pbx.c: Executing [s@from-sip-external:11] Congestion(“PJSIP/anonymous-00000021”, “5”) in new stack
[2018-12-28 13:09:13] VERBOSE[3986][C-00000022] pbx.c: Executing [s@from-sip-external:10] PlayTones(“PJSIP/anonymous-00000022”, “congestion”) in new stack
[2018-12-28 13:09:13] VERBOSE[3986][C-00000022] pbx.c: Executing [s@from-sip-external:11] Congestion(“PJSIP/anonymous-00000022”, “5”) in new stack
[2018-12-28 13:09:15] VERBOSE[3924][C-00000021] pbx.c: Spawn extension (from-sip-external, s, 11) exited non-zero on ‘PJSIP/anonymous-00000021’
[2018-12-28 13:09:15] VERBOSE[3924][C-00000021] pbx.c: Executing [h@from-sip-external:1] Hangup(“PJSIP/anonymous-00000021”, “”) in new stack
[2018-12-28 13:09:15] VERBOSE[3924][C-00000021] pbx.c: Spawn extension (from-sip-external, h, 1) exited non-zero on ‘PJSIP/anonymous-00000021’
[2018-12-28 13:09:17] VERBOSE[2615] pbx_variables.c: Setting global variable ‘SIPDOMAIN’ to ‘203.40.47.143’
[2018-12-28 13:09:17] VERBOSE[3988][C-00000023] pbx.c: Executing [953014763000942@from-sip-external:1] NoOp(“PJSIP/anonymous-00000023”, “Received incoming SIP connection from unknown peer to 953014763000942”) in new stack
[2018-12-28 13:09:17] VERBOSE[3988][C-00000023] pbx.c: Executing [953014763000942@from-sip-external:2] Set(“PJSIP/anonymous-00000023”, “DID=953014763000942”) in new stack
[2018-12-28 13:09:17] VERBOSE[3988][C-00000023] pbx.c: Executing [953014763000942@from-sip-external:3] Goto(“PJSIP/anonymous-00000023”, “s,1”) in new stack
[2018-12-28 13:09:17] VERBOSE[3988][C-00000023] pbx_builtins.c: Goto (from-sip-external,s,1)
[2018-12-28 13:09:17] VERBOSE[3988][C-00000023] pbx.c: Executing [s@from-sip-external:1] GotoIf(“PJSIP/anonymous-00000023”, “1?setlanguage:checkanon”) in new stack
[2018-12-28 13:09:17] VERBOSE[3988][C-00000023] pbx_builtins.c: Goto (from-sip-external,s,2)
[2018-12-28 13:09:17] VERBOSE[3988][C-00000023] pbx.c: Executing [s@from-sip-external:2] Set(“PJSIP/anonymous-00000023”, “CHANNEL(language)=en”) in new stack
[2018-12-28 13:09:17] VERBOSE[3988][C-00000023] pbx.c: Executing [s@from-sip-external:3] GotoIf(“PJSIP/anonymous-00000023”, “1?noanonymous”) in new stack
[2018-12-28 13:09:17] VERBOSE[3988][C-00000023] pbx_builtins.c: Goto (from-sip-external,s,5)
[2018-12-28 13:09:17] VERBOSE[3988][C-00000023] pbx.c: Executing [s@from-sip-external:5] Set(“PJSIP/anonymous-00000023”, “TIMEOUT(absolute)=15”) in new stack
[2018-12-28 13:09:17] VERBOSE[3988][C-00000023] func_timeout.c: Channel will hangup at 2018-12-28 13:09:32.925 AEDT.
[2018-12-28 13:09:17] WARNING[3988][C-00000023] func_channel.c: Unknown or unavailable item requested: ‘recvip’
[2018-12-28 13:09:17] VERBOSE[3988][C-00000023] pbx.c: Executing [s@from-sip-external:6] Log(“PJSIP/anonymous-00000023”, "WARNING,"Rejecting unknown SIP connection from “”) in new stack
[2018-12-28 13:09:17] WARNING[3988][C-00000023] Ext. s: "Rejecting unknown SIP connection from "

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@gregorywest wrote:

What I am trying to do if from and phone that is ringing from a ring-group. Be able to press a ‘feature’ button on one of the phones and transfer the call to a different extension. ie, call comes in for a sales person, all the sales phones start ringing. Would like to have any sales person press a button, and have the call sent to a specific voice mail box, and all the phones go quiet.

is this even possible?

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@moehammond wrote:

Dear All, I have purchased Voicemail notification yesterday in order to change the From and Subject to my own custom brand.

When trying to configure a new voicemail notification for an extension, I try to call the extension to leave a voice mail however at the end of leaving the voice mail I press # and get an error saying “I am sorry an error has occurred”

Before I setup the Voicemail Notification for this extension I used to get the regular notification with the subject “FreePBX VoiceMail Notification”

I would appreciate any help
Thanks

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