@ssi_tmckee wrote:
I’m new to FreePBX and running FreePBX 2.11 and astersik 11. All of a sudden my voicemail to email stopped working. I’ve verified my relay server can send emails, and I’ve checked the main.cf to make sure the correct host was in the config.
I don’t have the SMTP config option in the System Admin, so where can I troubleshoot why voicemail is not sending out?
Posts: 2
Participants: 2
@HDS_Nick wrote:
The remote extensions were fully working a while ago, but at some point they started to have this issue, the two remote users took quite a while to report the issue so we don’t know when exactly it started to happen thus what might have changed. Remote extensions are a Mitel 6865i and a Aastra 6731i; I’ve tried it on multiple internet connections (all AT&T with plenty of bandwidth) and all extras/helpers in the home router turned off, but same issues. FreePBX is running in ESXi with a SonicWall in front. I’ve attached screenshots of relevant settings and below is the behavior of the remote phones:
Remote > External: Two Way Audio
Remote > Internal: No Audio
Internal > Remote: Voicemail (remote phone doesn’t ring)
External > Remote: UntestedI’ve tried everything I could think of and quite a few search result suggestions, any thoughts would be appreciated…
Since apparently as a newbie I can only post one image and can’t post links, here is a link to the images:
files.starbase7(dot)net/pbx/
Posts: 1
Participants: 1
@dan_ce wrote:
Executing WhoCalled UK Searching https://who-called.co.uk/Number/02031291891 ... Average Rate: Harassing Number of Searches: 653 Site returned unexpected rate of Harassing , doing nothing result took 0.5374 seconds.Just wondering how I can also act on ‘
harassing’ or ‘dangerous’ - not just SPAM?Thanks!
Posts: 1
Participants: 1
@jgiebler wrote:
[split from unrelated thread - mod]
@lgaetz Thanks again for such a quick response.
I have struggled to understand when to use the [from-internal-custom] in the past. I had read your other post (Hooking for Fun and Income) several times and just read it again.
Based on what was just confirmed, I am not sure how to handle the below situation… Maybe it should be a separate question post?
The below solution has seemingly been working fine for quite a while, but now you have me thinking that maybe it’s causing the dialing to bypass our default [from-local] context all together when 6 or 7 digit numbers are dialed.
Background… each of our phone systems uses 3 digit extensions and 4 digit groups. However, we have a network of phone systems that all route through a central “hub”. Each phone system is assigned a unique 3 digit number that is essentially it’s “locator”. When a 6 or 7 digit number is dialed, the custom context appends the correct local “locator” before sending the call to the IAX2 trunk. This ensures that if someone calls back the number on the caller ID, they will call back the correct extension.
[extension_custom.conf]
; Add "Phone System Locator" code to front of extension when another 6 digit ; extension has been dialed. ; This will make "call backs" work correctly when calling from Office to Office ; This has been Set for vPBX-XXX-01 which is: 100 [from-internal-custom] ; Dialed 6 digits (Company Extension - Site Prefix + Destination Extension) exten => _XXXXXX,1,GotoIf($[${LEN(${CALLERID(num)})} = 3]?setcid:end) exten => _XXXXXX,n(setcid),NoOp(CUSTOM "from-internal-custom" in /etc/asterisk/extension_custom.conf_) exten => _XXXXXX,n,NoOp(CUSTOM: 6 digit extension dialed. Changing Caller ID for ${CALLERID(all)} to 100${CALLERID(number)}) exten => _XXXXXX,n,Set(CALLERID(number)=100${CALLERID(number)}) exten => _XXXXXX,n,NoOp(CUSTOM: Caller ID changed to ${CALLERID(all)}) exten => _XXXXXXX,(end)n,NoOp() ; Dialed 7 Digits (Company Groups, Queues etc - Site Prefix + Group/Queue etc Number) exten => _XXXXXXX,1,GotoIf($[${LEN(${CALLERID(num)})} = 3]?setcid:end) exten => _XXXXXXX,n(setcid),NoOp(CUSTOM "from-internal-custom" in /etc/asterisk/extension_custom.conf_) exten => _XXXXXXX,n,NoOp(CUSTOM: 7 digit extension dialed. Changing Caller ID for ${CALLERID(all)} to 100${CALLERID(number)}) exten => _XXXXXXX,n,Set(CALLERID(number)=100${CALLERID(number)}) exten => _XXXXXXX,n,NoOp(CUSTOM: Caller ID changed to ${CALLERID(all)}) exten => _XXXXXXX,(end)n,NoOp()
Posts: 5
Participants: 2
@FOGG wrote:
When clicking on a missed call number it dials the area code + number, but we need it to dial a 9+1 before dialing the rest of the number for external outbound calls. How can this be done?
Posts: 1
Participants: 1
@byerkes wrote:
We are trying to debug an issue with one of our servers and I am not sure we are getting enough information in the core dump. Right now our back trace doesn’t seem to have enough information to properly track down what is locking the threads.
They all look like this with the message no symbol information.
Thread 458 (Thread 0x7fea35c24700 (LWP 21156)):
#0 0x00007feb62ad8a5e in pthread_cond_timedwait@@GLIBC_2.3.2 () from /lib64/libpthread.so.0
No symbol table info available.
#1 0x000000000053ee8e in __ast_cond_timedwait ()
No symbol table info available.
#2 0x000000000046731e in ast_audiohook_trigger_wait ()
No symbol table info available.
#3 0x00007feabc79bdf2 in ?? () from /usr/lib64/asterisk/modules/app_mixmonitor.so
No symbol table info available.
#4 0x0000000000607c84 in ?? ()
No symbol table info available.
#5 0x00007feb62ad4aa1 in start_thread () from /lib64/libpthread.so.0
No symbol table info available.
#6 0x00007feb61e5c93d in clone () from /lib64/libc.so.6
No symbol table info available.Is there a way to get additional information from the core dump?
Posts: 1
Participants: 1
@Peter3b wrote:
I have got my server setup and running ad have all phones working internally but now it is te to get my trunks configuration done for att inflexible reach and I am not having any luck I have made many phone calls to att and no luck in any help I am hopping someone on here has set a successful trunk up using att iP flex and could help me with a sample configuration for now I have everything setup as chan_sip
Thank you
Peter
Posts: 2
Participants: 2
@ahmedalmulki wrote:
Hi, I purchased a new PC to install Freepbx on it, and this PC contains 2 m.2 SSD hard disk with Raid 1, I tried to install Freepbx on it but without any luck always “pane is dead” error , and I tried the Advanced option to set hard manually but also I can’t find it.
please, any thoughts …
Posts: 1
Participants: 1
@URTHllc wrote:
When trying to install the latest Distro I get this error message
Disk “” given in clearpart command does not exist
Posts: 1
Participants: 1
@wpns wrote:
Whenever I do a ‘module reload’ in asterisk -r or “Apply Config” in the web GUI, I get 31of the above errors, followed by:
Unable to register extension at line 3892 of /etc/asterisk/extensions_additional.conf
where the line number is in the 3900 range.that file has stuff like:
[clean]
include => clean-custom
exten => ,1,Noop(====> Room ${CALLERID(num)} is Clean now <====)
exten => ,n,AGI(mk_clean.agi,${CALLERID(num)},1)
exten => ,n,Hangup;–== end of [clean] ==–;
[receptionist]
include => receptionist-custom
exten => 100,1,Goto(from-internal,${EXTEN},1)
exten => 100,n,Hangup;–== end of [receptionist] ==–;
[mini-bar]
include => mini-bar-custom
exten => ,1,Noop(====> Room ${CALLERID(num)} ask a mini-bar <====)
exten => ,n,Playback(pms/drink-code)
exten => ,n(read_key),Read(Minibar,1,5)
exten => ,n,GotoIf($["${Minibar}" != “*”]?add_mini_bar)
exten => ,n,Set(minibar=${mini_bar})
exten => ,n,AGI(minibar.agi,${CALLERID(num)},${minibar})
exten => ,n,Playback(goodbye)
exten => ,n,Goto(end)
exten => ,n(add_mini_bar),Set(mini_bar=${mini_bar}${Minibar} )
exten => ,n,SayNumber(${Minibar:0:1},c)
exten => ,n,Goto(read_key)
exten => ,n(end),Hangupin that range. Have I done something wrong in setting things up? I have a very basic SNG7-FPBX-64bit-1904-2 build on a Mid-2010 Mac Mini with all the latest updates, a single Polycom SoundPoint IP-650 phone registered and (as yet) no trunks.
Yeah, I know it’s not a very serious problem to not be able to do the mini-bar thing with my bare-bones setup, but shouldn’t the defaults be something that doesn’t throw errors?
Thanks!
Posts: 3
Participants: 3
@junhaoooo wrote:
Im using a Zulu softphone. Incoming calls work fine, but outgoing calls only have incoming audio; when we call out, we can hear them, but they can’t hear us. Any kind advice for solutions to try? Appreciate it.
Posts: 1
Participants: 1
@Zezas wrote:
Hello,
I am working on setting up a system in which they need to have multiple options for night modes. Due to the nature of the business that we run we answer the phone with a live person 24/7 365. We have 4 employees that have ip phones in their homes and I have set up a total of 5 queues (one for each person who takes calls at night and one that routes to the office for during the day). I am trying to figure out how to set it up so that one of five dial codes could be dialed that would result in calls being routed to that queue. Any suggestions?
Posts: 1
Participants: 1
@Flurd wrote:
I’m trying to understand the best way to set up my call centre in the move over to FreePBX/PBXAct and have come up against Device & User Mode vs Extension Mode. The former is not supported but does allow an extension to exist wherever a user logs in and that would allow the user to be automatically logged into their queues. To my mind, this is exactly what I want by the term Hot Desking.
Extension mode seems to be completely opposite to what a call centre needs so how have other people handled this? Are there commercial modules that simply resolve this issue or is there some configuration I’m not seeing?
I have a test box of FreePBX that I’m playing around with and going through the Sangoma PBXAct essentials videos but nothing I’ve seen there or in the wiki/forum makes a clear suggestion of what a call centre should do to handle hot desking and queues.
Posts: 2
Participants: 2
@andrewj wrote:
I buy the Sangom S305 phone + Sysadmin Pro module.
I setup the OpenVPN and confirm from my laptop it connects fine and I can access the FreePBX
Then I provision the phone for OpenVPN and the VPN icon shows up at time. But I take the phone offsite and the VPN connection is not established, it shows in the menu VPN IP = 0.0.0.0
How can I view the logs of VPN on the phone? When I login to the web interface there is no mention of VPN
Posts: 1
Participants: 1
@BrianTullio wrote:
I currently have a working system - FreePBX 12 and many Polycom IP phones.
Right now, everything is on the default VLAN 1 (192.168.1.x).
For testing purposes, I moved 1 phone to a different VLAN 12 (192.168.12.x). At this point, communication between VLAN 1 and 12 is wide open. There are no restrictions and everybody can communicate with everybody else.
When I boot the test phone on VLAN 12, it grabs the right IP address, it can connect to the provisioning server, grabs the correct settings, it gets the phone background, and it grabs the current time. The only thing it will not do is register the line.
I can ping the phone from VLAN 1 and even connect to the phones web interface. I hooked up a laptop on VLAN 12 and I can ping the FreePBX server and login to the admin interface.
I have 192.168.1.x and 192.168.12.x listed as local networks under Settings -> Asterisk SIP Settings -> General SIP Settings -> NAT Settings -> Local Networks
If I switch the phone back to VLAN 1, the line registers fine.
What am I missing?
Posts: 2
Participants: 2
@AnthKay92 wrote:
I thought I would start a clean thread as struggling to get this working
I have two extensions registered to two different phones.
1001 on one phone
1004 on the other phoneThey can dial between each other.
I want calls to log in stasis ARI.
On both extensions, I change the context to from-internal-test
In extensions_custom.conf I enter the below dial plan:
[from-internal-test]
exten => _1XXX,1,NoOp()
same => n,Answer()
same => n,Stasis(hello-world)
same => n,Hangup()So, I have node.js and wscat installed and connected using an ARI and logging into the wscat command prompt as below (when I dial from 1001 to 1004):
My problem now is that when I dial 1004, the phone displays “talking” and it doesnt actually dial 1004…
In the logs, I see the call does not even get to 1004:
I want 1004 to ring when dialled from 1001 AND also log into the stasis ARI. At this minute I cannot get both to work.
I change the extension context back to from-internal and it rings but events do not fire into the ARI.
I change it to from-internal-test and I cannot get the other phone (1004) to dial. Really struggling with this and any help would be appreciated.Thanks in advance and sorry for the repost.
Posts: 1
Participants: 1
@AnthKay92 wrote:
Executed the wscat command in Node.Js prompt.
In Asterisk, we get a WebSocket connection and a message telling us that our Stasis application has been created
Now when I dial between extensions I get events logged with the following in my extensions_custom.conf and using from-internal-custom context:
[from-internal-custom]
exten => 2003,1,NoOp(entering ${CONTEXT} as defined in extensions_custom.conf)
same => n,Answer()
same => n,Stasis(Tess_Asterisk,PJSIP/${EXTEN}, 45)
same => n,Hangup()
[from-internal-custom]
exten => 2004,1,NoOp(entering ${CONTEXT} as defined in extensions_custom.conf)
same => n,Answer()
same => n,Stasis(Tess_Asterisk,PJSIP/${EXTEN}, 45)
same => n,Hangup()
[from-internal-custom]
exten => 1001,1,NoOp(entering ${CONTEXT} as defined in extensions_custom.conf)
same => n,Answer()
same => n,Stasis(Tess_Asterisk,PJSIP/${EXTEN}, 45)
same => n,Hangup()I have tried to replace the context with a wildcard pattern so I do not need to list an extension every time… (as below)
[from-internal-custom]
exten => XXXX,1,NoOp(entering ${CONTEXT} as defined in extensions_custom.conf)
same => n,Answer()
same => n,Stasis(Tess_Asterisk,PJSIP/${EXTEN}, 45)
same => n,Hangup()But then events do not show… Please could someone help me with the correct contect I need to use to get this working??? Any help appreciated!
Posts: 18
Participants: 5
@shellbr wrote:
Hello,
I’ve just started learning the details of VOIP. I currently have a VOIP line at home provided by 1-VOIP and had the idea maybe I could stand up a FreePBX server and have it use 1-VOIP as my outside line and if calls come in, FreePBX could handle them as IVR or forward directly to an extension like a DID. What I’m finding though is all the config guides are for trunks, and I think what I have is not a SIP “trunk” since it is only a single phone number. Or maybe it is technically a trunk with only 1 phone number? I’m not sure. I do connect to the provider with SIP server name which is a public IP, username, and password. Softphones connect to the provider fine, but I’m not sure how to adapt the FreePBX config instructions to what I have. Anyone have some guidance for me?
Posts: 3
Participants: 2
@GGCoffee wrote:
Reference:
Dave, I got called out of town, and the original thread was automatically closed!
I executed the command, and there was no response. I was logged in as root and executed it from the root directory.
Thanks,
Mike
Posts: 1
Participants: 1
@dwight wrote:
My FreePBX systems can be accessed by using one of two fully qualified domain names - for example pbx-example-com and pbx5-example-com. My phones register to pbx but when I am building or managing the servers I access it via pbx5. When it is time to build a new server I will build it as pbx6. When I am ready to transition to the new server I move the pbx name to point to pbx6.
I have no problems with getting the certificates using certbot.
First with:
sudo certbot --apache -d pbx7-example-com
(I am a new user, the “-” should be “.”)And then when I have repointed pbx:
sudo certbot --apache -d pbx.example-com -d pbx7-example-comI would like to be able to do this using the Certificate Manager in FreePBX but I can’t figure out how to get the Certificate Manager to generate a single certificate for both pbx and pbx5.
One driver in wanting to do this inside FreePBX and Certificate Manager is that I need to use WebRTC and I would prefer to use a single certificate for HTTPS and WebRTC.
I would appreciate any thoughts on how to accomplish the two names in a single certificate using FreePBX and Certificate Manager or any other ways of accomplishing this in a straight forward manner that will not involve manual intervention every time the certificates are renewed.
By the way, I am running FreePBX 16-15 on CentOS 7.
Thank you.
Posts: 1
Participants: 1
