@mvogel4949 wrote:

I took a backup of System A. Downloaded it. Then restored it to System B. I browse to the file and then press restore. It cycles through the % sign at the bottom and then just comes back to a blank screen. If I restore just a CDR then it comes up with a check box asking me if I want to restore the CDR but a full backup offers me no choices. Any thoughts?

Posts: 1

Participants: 1

Read full topic

@bajramia wrote:

HI All,
I start having issue everytime i do a change on freepbx not keep it when i look im getting this error

db.c:350 ast_db_put: Couldn’t execute statment: SQL logic error or missing database
[2019-01-03 11:32:01] WARNING[208797]: db.c:350 ast_db_put: Couldn’t execute statment: SQL logic error or missing database
[2019-01-03 11:32:01] WARNING[208797]: db.c:350 ast_db_put: Couldn’t execute statment: SQL logic error or missing database
[2019-01-03 11:32:01] WARNING[208797]: db.c:350 ast_db_put: Couldn’t execute statment: SQL logic error or missing database
[2019-01-03 11:32:01] WARNING[208797]: db.c:350 ast_db_put: Couldn’t execute statment: SQL logic error or missing database
[2019-01-03 11:32:01] WARNING[208797]: db.c:350 ast_db_put: Couldn’t execute statment: SQL logic error or missing database

Please help thank you

Posts: 1

Participants: 1

Read full topic

@sxarthur wrote:

I have a server set up, and i can easily make calls between softphones here on the company, but i want to make/receive calls from outside, how do i do it?
i have Fxo and Fxs hardware set up, but i’ve seen little to none info on sip/analog trunking, most just say to pay a sip Company… (or a normal phone number and set up the fxo/fxs hardware) is it really the only way?

Posts: 2

Participants: 2

Read full topic

@GGroce63 wrote:

I’m attempting to setup faxing on our server and have also purchased Fax Pro. Outbound faxing works fine but I cannot see where to setup inbound termination to email. I do not get the option to add an email in the inbound route or extension.

Any advice is appreciated.

Posts: 1

Participants: 1

Read full topic

@sentinelace wrote:

I am guessing this is a provider issue but the provider is saying it’s a PBX issue. Incoming DTMF is sending the wrong dial tones for all numbers on my PBX. I have about 8. My sip settings are as follows:

disallow=all
type=friend
secret=username
username=password
;dtmfmode=rfc2833
dtmfmode=info
context=from-trunk
canreinvite=no
allow=ulaw
insecure=port,invite
host=x.x.x.x (real IP is here)

I have tried DTMF is auto, info, rfc2833 with all the same results. A test call log shows this after dialing 132 at the IVR:

Full log here: Log Files

Posts: 1

Participants: 1

Read full topic

@SilkBC wrote:

Hello,

We have a site that has 120+ phones. We have the reception phone (Aastra 6869i) with three sidecars and all the extensions on as BLF/Xfer, however only the first 50 or so extensions show proper status and the rest show question marks.

I learned from a previous post that this is a limitation of the hardware (phones) and was wondering if there are any phones out there that can handle 120+ BLF status’ properly. I don’t care about brand.

They really like having a physical phone. We installed FOP2 for them and they don;t like it; it is very awkward for them as they have a PC with two monitors at the desk already and the additional monitor and PC for the FOP2 is too awkward

(for reasons of their own, they can’t work with FOP2 on the PC at reception desk with the two monitors, so we had to set up a separate PC and monitor; using an extra mouse and keyboard with it confused them so we used a touch screen monitor, but working with FOP2 or anything PC-based on a touch screen sucks badly and is very awkward)

I might be able to sell them on using a softphone (not Zulu) if there can be a “virtual” side car that can show BLF status’ properly, if anyone knows of anything that might do the job.

Thoughts and ideas are appreciated.

Posts: 2

Participants: 2

Read full topic

@chrisfx wrote:

Hello,

I have an IVR with 3 different options, based on the option selected I would like an announcement to be played to the agent answering the call before the caller is connected. Can someone help me with this? thanks in advance.

Posts: 1

Participants: 1

Read full topic

@msmcknight wrote:

Hi everyone,

I’m trying to setup follow-me from a ring group to a cell phone aliased as an extension… and it’s not going so well.

I have the ring group set to call a set of extensions and one of those extensions has a follow-me setup that bounces to a cell. With follow-me setup as ringall, it appears as though the handoff is processing as expected. The hard extension rings for INITIAL_RING_TIME and then the call bounces to the cell extension and is forwarded out a SIP trunk to my cell phone. But then it gets weird…

The call rings on the cell for 10 seconds and then drops with the last messages in asterisk showing:

[Jan 3 21:26:41] – SIP/FlowRoute-SIP-00000c03 is ringing
[Jan 3 21:26:41] – SIP/FlowRoute-SIP-00000c03 is making progress passing it to Local/755@from-internal-00000012;2
[Jan 3 21:26:41] – Local/755@from-internal-00000012;1 is ringing
[Jan 3 21:26:52] == Spawn extension (macro-dial, s, 22) exited non-zero on ‘Local/202@from-internal-00000011;2’ in macro ‘dial’
[Jan 3 21:26:52] == Spawn extension (followme-sub, 202, 37) exited non-zero on ‘Local/202@from-internal-00000011;2’
[Jan 3 21:26:52] == Spawn extension (macro-dial-one, s, 54) exited non-zero on ‘Local/755@from-internal-00000012;2’ in macro ‘dial-one’
[Jan 3 21:26:52] == Spawn extension (macro-exten-vm, s, 14) exited non-zero on ‘Local/755@from-internal-00000012;2’ in macro ‘exten-vm’
[Jan 3 21:26:52] == Spawn extension (macro-dial, s, 22) exited non-zero on ‘SIP/2200-00000c01’ in macro ‘dial’
[Jan 3 21:26:53] == Spawn extension (ext-local, 755, 2) exited non-zero on ‘Local/755@from-internal-00000012;2’
[Jan 3 21:26:53] – Executing [h@ext-local:1] Macro(“Local/755@from-internal-00000012;2”, “hangupcall,”) in new stack
[Jan 3 21:26:53] – Executing [s@macro-hangupcall:1] GotoIf(“Local/755@from-internal-00000012;2”, “1?theend”) in new stack
[Jan 3 21:26:53] – Goto (macro-hangupcall,s,3)
[Jan 3 21:26:53] == Spawn extension (ext-group, 610, 14) exited non-zero on ‘SIP/2200-00000c01’
[Jan 3 21:26:53] – Executing [h@ext-group:1] Macro(“SIP/2200-00000c01”, “hangupcall,”) in new stack
[Jan 3 21:26:53] – Executing [s@macro-hangupcall:3] ExecIf(“Local/755@from-internal-00000012;2”, “0?Set(CDR(recordingfile)=)”) in new stack
[Jan 3 21:26:53] – Executing [s@macro-hangupcall:4] Hangup(“Local/755@from-internal-00000012;2”, “”) in new stack
[Jan 3 21:26:53] == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘Local/755@from-internal-00000012;2’ in macro ‘hangupcall’
[Jan 3 21:26:53] == Spawn extension (ext-local, h, 1) exited non-zero on ‘Local/755@from-internal-00000012;2’
[Jan 3 21:26:53] – Executing [s@macro-hangupcall:1] GotoIf(“SIP/2200-00000c01”, “1?theend”) in new stack
[Jan 3 21:26:53] – Goto (macro-hangupcall,s,3)
[Jan 3 21:26:53] – Executing [s@macro-hangupcall:3] ExecIf(“SIP/2200-00000c01”, “0?Set(CDR(recordingfile)=)”) in new stack
[Jan 3 21:26:53] – Executing [s@macro-hangupcall:4] Hangup(“SIP/2200-00000c01”, “”) in new stack
[Jan 3 21:26:53] == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/2200-00000c01’ in macro ‘hangupcall’
[Jan 3 21:26:53] == Spawn extension (ext-group, h, 1) exited non-zero on ‘SIP/2200-00000c01’

What’s odd is that if I answer the cell phone within this short 10s window, all seems ok. But, if I dont, it fails as shown above. But… the cell phone gets an odd recording. As soon as I see the last “hangup” message in the asterisk console, the message on the cell phone says:

“Please enter your area code and phone number followed by pound. If you make a mistake press blah blah blah…”

If I enter a number and area code and press pound, it then asks me to enter my PIN. I have no idea where this message is coming from. My guess is it’s something on the cell phone side of things b/c there are no more asterisk logs after the final “hangup” shown above.

But… this 10s drop only happens when an outside call is routed to an outside cell phone. If I call the follow-me extension from an internal extension, it rings through and keeps ringing.

At this exact 10s mark, I see debug messages showing SIP 102 CANCEL coming from the Linksys SPA3201 that’s handling the inbound POTS line. The outbound line to the cell is over a SIP trunk. Finally it all closes down with a SIP 103 CANCEL.

I have other inbound SIP trunks that auto forward to outbound SIP trunks to cell phones and they work without issue, but I’m not trying to use ring-groups or follow-me with those.

The precise and consistent 10s drop time is very suspicious to me. I just don’t know where to look for it.

This leaves me with a few perplexing questions…

1. If the odd recording is something on the cell phone side, what is it about the asterisk “hangup” thats triggering it, and how do I stop it. It does this with Consumer Cellular and Verizon cell phones. (so far)

2. How do I increase the allowed ring-time once the call is handed to the cell phone so the cell will ring long enough for the cell phone email to answer the call for calls that pass through from outside to outside? I have the RING_TIME set to 33 seconds. A native call allowed to ring on the cell goes to voicemail at 30s. But, no matter the setting, it all crashes at 10s.

It’s almost like there’s another timer somewhere that I’m missing that’s causing asterisk to drop the call after it’s handed to the cell phone. What’s odd is that no matter what I set the RING_TIME to, the hangup happens at the same 10s mark.

3. Could this be some kind of weird interaction between the SPA3201 POTS line and the outbound SIP Trunk? Is the precise 10s drop an indicator of some sort?

4. Is this a POTS-to-SIP issue and has nothing to do with ring groups and follow-me at all, or are these apps some how causing the problem?

I’m stumped. I thought I had it working with the calls being passed to the cell, caller ID being passed right along and then boom… calls drop right away. If I call the cell directly, or through the extension alias, there is no problem; so I don’t think there’s a routing issue or anything like that. I only have trouble when bouncing the cell through one of the apps (ring-groups / follow-me).

If anyone has any idea what this might be, or where I could look for answers, please let me know.

Thanks to you all in advance.

Posts: 1

Participants: 1

Read full topic

@Billyg wrote:

Hi Guys very new to freepbx
How do I set up remote access to user’s mailboxes?

Cheers

Posts: 2

Participants: 1

Read full topic

@mvogel4949 wrote:

This morning I was running some upgrades from the command line and after each upgrade (other than framework) I see the error below:

[root@Nimbus ~]# fwconsole ma upgradeall
No repos specified, using: [standard,extended,commercial,unsupported] from last GUI settings

Module(s) requiring upgrades: announcement, bulkhandler, calllimit, callrecording, cdr, conferencespro, contactmanager, core, dashboard, endpoint, fax, faxpro, findmefollow, framework, irc, ivr, languages, miscdests, music, paging, parking, parkpro, pm2, queueprio, queues, qxact_reports, restapps, sipsettings, sysadmin, ucp, userman, vega, voicemail
Upgrading module ‘framework’ from 14.0.5.5 to 14.0.5.20
Downloading module ‘framework’
Processing framework
Downloading…
12997025/12997025 [============================] 100%
Finished downloading
Extracting…Done
Download completed in 3 seconds
Updating tables admin, ampusers, cronmanager, featurecodes, freepbx_log, freepbx_settings, globals, module_xml, modules, notifications…Done
installing files to /var/www/html…done
installing files to /var/lib/asterisk/bin…done
installing files to /var/lib/asterisk/agi-bin…done
Checking for upgrades…
2 found
Upgrading to 14.0.5.9…
-> Running PHP script /var/www/html/admin/modules/framework/upgrades//14.0.5.9/upgrade.php
Upgrading to 14.0.5.9…OK
Upgrading to 14.0.5.12…
-> Running PHP script /var/www/html/admin/modules/framework/upgrades//14.0.5.12/upgrade.php
Upgrading to 14.0.5.12…OK
framework file install done, removing packages from module
file/directory: /var/www/html/admin/modules/framework/amp_conf removed successfully
file/directory: /var/www/html/admin/modules/framework/upgrades removed successfully
file/directory: /var/www/html/admin/modules/framework/start_asterisk removed successfully
file/directory: /var/www/html/admin/modules/framework/install removed successfully
file/directory: /var/www/html/admin/modules/framework/installlib removed successfully
Building Packaged Scripts…Done
Generating CSS…Done
Module framework successfully installed
Updating Hooks…Done
Upgrading module ‘core’ from 14.0.18.37 to 14.0.18.45
Downloading module ‘core’
Processing core
Downloading…
1108970/1108970 [============================] 100%
Finished downloading
Extracting…Done
Download completed in 1 seconds

The “–onlystdout” option does not exist.

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

Upgrading module ‘sipsettings’ from 14.0.27.5 to 14.0.27.7
Downloading module ‘sipsettings’
Processing sipsettings
Downloading…
259586/259586 [============================] 100%
Finished downloading
Extracting…Done
Download completed in 0 seconds

The “–onlystdout” option does not exist.

Posts: 1

Participants: 1

Read full topic

@hardocp wrote:

Asterisk 16 Freepbx 14

Received a notification about a bunch of module upgrades including framework

Went in to do the update and got the following error for Framework (which is now disabled)

fwconsole ma install framework
Updating tables admin, ampusers, cronmanager, featurecodes, freepbx_log, freepbx_settings, globals, module_xml, modules, notifications…Done
installing files to /var/www/html…done
installing files to /var/lib/asterisk/bin…done
installing files to /var/lib/asterisk/agi-bin…done
Checking for upgrades…
2 found
Upgrading to 14.0.5.9…
-> Running PHP script /var/www/html/admin/modules/framework/upgrades//14.0.5.9/upgrade.php

In upgrade.php line 3:

syntax error, unexpected ‘(’ in /etc/asterisk/asterisk.conf on line 2

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

Current framework version is 14.0.5.5 and the latest is 14.0.5.20

Can you please help me fix this

thanks

Posts: 2

Participants: 2

Read full topic

@threeeye wrote:

Hi guys,
I’m configuring a Cisco 8961, all is good (I think)…
I have a little problem, I can’t open missed calls, and I can’t get rid of them…
There isn’t any softkeys when there is a missed call…

Thanks for the help

Posts: 1

Participants: 1

Read full topic

@osantos13 wrote:

Hello,

I set up a new PBX box running PJSIP and I am receiving a warning stating my softphone client is offering an unsupported crypto suite and simply defaults to using AES_CM_128_HMAC_SHA1_80. Is there a way to adjust this to use the stronger cipher suite? Please see warning below:

[2019-01-04 12:13:10] WARNING[16962]: sdp_srtp.c:261 ast_sdp_crypto_process: Unsupported crypto suite: AES_256_CM_HMAC_SHA1_80
[2019-01-04 12:13:10] WARNING[16962]: sdp_srtp.c:261 ast_sdp_crypto_process: Unsupported crypto suite: AES_256_CM_HMAC_SHA1_32

I have upgraded sipsettings to v13.0.27.5 which now includes support to TLS1.1 and TLS1.2 as tracked in the Sangoma Issue Tracker ticket #18597.

Any help would be greatly appreciated. Thank you.

Posts: 1

Participants: 1

Read full topic

@pakenvs wrote:

I have looked for this problem and cannot find it anywhere.

We are running Freepbx 14 with Asterisk 13. Our extensions are pjsip. When we do an attended transfer, if the transfer is completed while the final destination is still ringing, there is no audio.

Example:
User 1 calls User 2; User 2 does an attended transfer to User 3. If User 2 completes the transfer while it is still ringing User 3 then User 1 and User 3 have no audio at all. If User 3 puts the call on hold and picks it back up, audio comes back both directions.

An attended transfer where User 2 waits for User 3 to answer, works fine; both User 1 and User 3 have audio both directions. A blind transfer also works fine.

Posts: 1

Participants: 1

Read full topic

@JohnCO wrote:

Hey guys, I’m having a weird issue. We’re running FreePBX 13 (planning to upgrade to 14 eventually, but it’s a bear with Hyper-V) and use FollowMe with call confirm extensively. This has worked fine in the past, where every now and again when we hit 1 to confirm the call it doesn’t take it. However, starting today every single time when we hit 1 the PBX just ignores it. I saw there was issue 18728 which seems to describe my exact issue, I installed the relevant update, no dice. Here’s an excerpt from the log when it’s trying to forward to my cell:

Local/<MYCELL#>@from-internal-0000000f;1 answered Local/RG-601*-<MYCELL#>#@from-internal-0000000e;2

– Executing [s@macro-confirm:1] Set(“Local/<MYCELL#>@from-internal-0000000f;1”, “LOOPCOUNT=0") in new stack

– Executing [s@macro-confirm:2] Set(“Local/<MYCELL#>@from-internal-0000000f;1”, “__MACRO_RESULT=ABORT”) in new stack

– Executing [s@macro-confirm:3] Set(“Local/<MYCELL#>@from-internal-0000000f;1”, “MSG1=incoming-call-1-accept-2-decline”) in new stack

– Executing [s@macro-confirm:4] BackGround(“Local/<MYCELL#>@from-internal-0000000f;1”, “incoming-call-1-accept-2-decline,m,en,macro-confirm”) in new stack

– <Local/<MYCELL#>@from-internal-0000000f;1> Playing ‘incoming-call-1-accept-2-decline.slin’ (language ‘en’)

– Channel SIP/VP-SIPJFKA-00000005 joined ‘simple_bridge’ basic-bridge <54f52b9e-d772-4b87-a573-d881f07701a0>

– Channel Local/<MYCELL#>@from-internal-0000000f;2 joined ‘simple_bridge’ basic-bridge <54f52b9e-d772-4b87-a573-d881f07701a0>

> 0x7f37d018d180 – Strict RTP switching to RTP target address 52.45.28.115:19960 as source

[2019-01-04 14:27:48] WARNING[2667]: chan_sip.c:4127 retrans_pkt: Timeout on CqvHElmTh6bLuWMY.Call on non-critical invite transaction.

[2019-01-04 14:27:48] WARNING[2667]: chan_sip.c:4127 retrans_pkt: Timeout on CqvHElmTh6bLuWMY.Call on non-critical invite transaction.

> 0x7f37d018d180 – Strict RTP learning complete - Locking on source address 52.45.28.115:19960

– Executing [s@macro-confirm:5] Read(“Local/<MYCELL#>@from-internal-0000000f;1”, “INPUT,1,4") in new stack

– Accepting a maximum of 1 digits.

– User entered nothing.

– Executing [s@macro-confirm:6] GotoIf(“Local/<MYCELL#>@from-internal-0000000f;1”, “0?,1:t,1") in new stack

I have confirmed that outbound calls work fine with audio both directions, so I don’t think it’s an RTP issue of the PBX not “hearing” the keypress. I have also rebuilt asterisk to the latest version of 13 (13.23 I think?). I’ve installed installed every module update available, no dice. There were no changes made to our PBX around the time this issue started, AFAIK.

Thanks for your time!

Posts: 1

Participants: 1

Read full topic

@duli wrote:

The caller id pattern the telco provides for me is 211199999999, where 21 is the long distance operator company, 11 is the area/city code, and the rest is the actual phone number.

I can successfuly make this detection with a normal/analog telephone with a caller id display directly connected to the voip port on the modem.

But my box (FreePBX 14.0.5.20), it always skips the first number (2 in this case).

So the number 211135386355 is detected like 11135386355:

-- Executing [in@sub-record-check:2] Set("DAHDI/1-1", "FROMEXTEN=unknown") in new stack                 
-- Executing [in@sub-record-check:3] ExecIf("DAHDI/1-1", "11?Set(FROMEXTEN=11135386355)") in new stack

This is a VoIP line that arrives at the telco modem and, from there, I connect it with a normal telephone 6p4c > 4p4c cable to my Sangoma analog card (A200).

I’m using the default configuration (bell for cidsignalling and ring for cidstart). If I change it to dtmf, it does not make any detection.

Wht is FreePBX / Asterisk missing the first number?

Any pointers would be appreciated. Thanks.

Posts: 1

Participants: 1

Read full topic

@4allbusiness wrote:

We have multiple Fpbx 14 machines which cannot acces web gui anymore after auto update modules last night.
Is there a way, we can downgrade a module through cli?
See screenshot.

sshot-46.png1121x809 60.3 KB

Posts: 3

Participants: 2

Read full topic

@Trunk wrote:

I’m using FreePBX 14.0.5.23 and Asterisk 15.5.0. I have a Chan_SIP trunk to german Telekom. With incoming calls, Telekom sends the DID in the to header, so the context from-pstn-toheader is necessary and so i specified context=from-pstn-toheader in the User details in incoming sip settings of the trunk. Unfortunately it was completely ignored. For incoming calls the context from-trunk was used and the result was that the displayed DID was always just “s” and no inbound routes matched “s”. As the next step, i also set default context to from-pstn-toheader (in Settings - Asterisk SIP Settings - Chan-SIP-Settings). Since i did that, the behaviour is really strange. After reboot it works fine, the context from-pstn-toheader is used for incoming calls, DID and inbound routing is correct. But some time later, it suddenly does not work anymore, for incoming calls the context from-trunk is used instead of pstn-to-header.until next reboot. This behaviour occurs though no changes are made in the GUI or CLI of freepbx. Sometimes the “context change” occurs after the first incoming call or after a few minutes, sometimes it occurs after about 20 incoming calls or after a few hours. I also don’t see any particular entries In the log when the context changes. There are log entries of the normal Hangup procedure of the preceding call directly followed by the entries of the normal start procedure of the next call.

Any suggestions are appreciated.
Many Thanks

Posts: 1

Participants: 1

Read full topic

@arielgrin wrote:

FreePBX version: 10.13.66-22
Asterisk Version: 13.23.1
ucpnode module version: 13.0.34.9
kernel version: 2.6.32-642.6.2.el6.i686

Hi all. While executing fwcnonsole ma updateall, I get the following error when ucpnode is being updated from 13.0.34.9 to 13.0.34.10

npm ERR! Linux 2.6.32-642.6.2.el6.i686
npm ERR! argv “/usr/bin/node” “/usr/bin/npm” “install”
npm ERR! node v0.12.18
npm ERR! npm v2.15.11
npm ERR! code ELIFECYCLE

npm ERR! node-expat@2.3.17 install: node-gyp rebuild
npm ERR! Exit status 1
npm ERR!
npm ERR! Failed at the node-expat@2.3.17 install script ‘node-gyp rebuild’.
npm ERR! This is most likely a problem with the node-expat package,
npm ERR! not with npm itself.
npm ERR! Tell the author that this fails on your system:
npm ERR! node-gyp rebuild
npm ERR! You can get information on how to open an issue for this project with:
npm ERR! npm bugs node-expat
npm ERR! Or if that isn’t available, you can get their info via:
npm ERR!
npm ERR! npm owner ls node-expat
npm ERR! There is likely additional logging output above.

npm ERR! Please include the following file with any support request:
npm ERR! /var/www/html/admin/modules/ucpnode/node/npm-debug.log
[npm-cache] [ERROR] [npm] error running npm install
[npm-cache] [ERROR] error installing dependencies

Failed updating libraries!

After this, ucpnode remains stopped, but I was able to restart it with fwconsole start ucpnode.

I don’t know if this error should be fixed or is just for information.

Any help appreciated.

Posts: 1

Participants: 1

Read full topic

@genscript wrote:

We make Time Group and Time Condition for our users.
When user type *271, they can hear “Time Activated”.
But when we call an external call from extension to extension, it seems not to apply.
Could you help on this?

Posts: 2

Participants: 2

Read full topic