@simone686 wrote:

Hi all

i inherited from a customer a multi tenant freepbx Server and i was charged to move it to a newer freepbx version…
Almost all fine but they used to add call forward by astdb via command line (dont’ ask me why) like so:
"database put cf 200 032xxxxxxxxx "
On prior version, CF extensions used the trunk assigned for outbound calls…this new version uses the first route defined on outbound routes…
I know you can’t answer precisely for my case so i’m asking any directive or info about the configuration file,macro or similar involved on CF to check for, so i can compare old configuration files with newer…
Thanks

Posts: 2

Participants: 2

Read full topic

@WLS_IT_Guy wrote:

I am getting the following error when trying to run updates through the updates module and CLI. I am not sure how to get the updates to work. I don’t want to break anything by trying to remove or reboot prematurely if I can help it

Loaded plugins: fastestmirror, versionlock
Loading mirror speeds from cached hostfile
Resolving Dependencies
--> Running transaction check
---> Package openssl.x86_64 1:1.0.2k-12.el7 will be updated
---> Package openssl.x86_64 1:1.0.2k-13.el7 will be an update
---> Package openssl-devel.x86_64 1:1.0.2k-12.el7 will be updated
---> Package openssl-devel.x86_64 1:1.0.2k-13.el7 will be an update
---> Package openssl-libs.x86_64 1:1.0.2k-12.el7 will be updated
---> Package openssl-libs.x86_64 1:1.0.2k-13.el7 will be an update
---> Package python-zmq.x86_64 0:15.3.0-2.el7 will be updated
---> Package python-zmq.x86_64 0:15.3.0-3.el7 will be an update
---> Package salt.noarch 0:2018.3.2-1.el7 will be updated
---> Package salt.noarch 0:2018.3.3-1.el7 will be an update
---> Package salt-minion.noarch 0:2018.3.2-1.el7 will be updated
---> Package salt-minion.noarch 0:2018.3.3-1.el7 will be an update
--> Finished Dependency Resolution
Error:  Multilib version problems found. This often means that the root
       cause is something else and multilib version checking is just
       pointing out that there is a problem. Eg.:
       
         1. You have an upgrade for openssl-libs which is missing some
            dependency that another package requires. Yum is trying to
            solve this by installing an older version of openssl-libs of the
            different architecture. If you exclude the bad architecture
            yum will tell you what the root cause is (which package
            requires what). You can try redoing the upgrade with
            --exclude openssl-libs.otherarch ... this should give you an error
            message showing the root cause of the problem.
       
         2. You have multiple architectures of openssl-libs installed, but
            yum can only see an upgrade for one of those architectures.
            If you don't want/need both architectures anymore then you
            can remove the one with the missing update and everything
            will work.
       
         3. You have duplicate versions of openssl-libs installed already.
            You can use "yum check" to get yum show these errors.
       
       ...you can also use --setopt=protected_multilib=false to remove
       this checking, however this is almost never the correct thing to
       do as something else is very likely to go wrong (often causing
       much more problems).
       
       Protected multilib versions: 1:openssl-libs-1.0.2k-13.el7.x86_64 != 1:openssl-libs-1.0.2k-12.el7.i686

Posts: 1

Participants: 1

Read full topic

@mcomtec wrote:

First of all I want to thank everyone.

I installed the distro "STABLE
SNG7-PBX-64bit-1805-1 "with Asterisk 15, but I can not receive faxes 99% of the time the reception fails and incoming faxes do it by G.711 without using Spandsp T.38.

I have tried to change from 9600 bps to 14400 bps, but I can not get it to work.

I have tried many things, and I have been trying and testing for more than a month without any results. Can you help me?

Thank you!

fax show stats

FAX Statistics:

Current Sessions : 0
Reserved Sessions : 0
Transmit Attempts : 0
Receive Attempts : 18
Completed FAXes : 18
Failed FAXes : 17

Spandsp G.711
Success : 1
Switched to T.38 : 0
Call Dropped : 6
No FAX : 0
Negotiation Failed : 0
Train Failure : 0
Retries Exceeded : 7
Protocol Error : 0
TX Protocol Error : 0
RX Protocol Error : 4
File Error : 0
Memory Error : 0
Unknown Error : 0

Spandsp T.38
Success : 0
Call Dropped : 0
No FAX : 0
Negotiation Failed : 0
Train Failure : 0
Retries Exceeded : 0
Protocol Error : 0
TX Protocol Error : 0
RX Protocol Error : 0
File Error : 0
Memory Error : 0
Unknown Error : 0

Posts: 4

Participants: 2

Read full topic

@bend94 wrote:

Hi

I installed freebpx. I created one user 6005 and one extensions
i try to connect from my laptop with zoiper but i received these messages

res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘REGISTER’ from failed for ‘192.168.0.3:49438’ (callid: 7CJGrjGQqat5JhZY1WjmUw…) - No matching endpoint found
res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘REGISTER’ from failed for ‘192.168.0.3:49438’ (callid: 7CJGrjGQqat5JhZY1WjmUw…) - No matching endpoint found
res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘REGISTER’ from failed for ‘192.168.0.3:49438’ (callid: 7CJGrjGQqat5JhZY1WjmUw…) - Failed to authenticate

please advise
Regards

Posts: 2

Participants: 2

Read full topic

@Keshka wrote:

Our company is encountering a situation where calls have one or both sides of a conversation stop working. Very much like all of us encounter with cell phones “can you hear me now?”. We have tried to find a pattern to this, what line, phone, in or out, time of day, call id…you name it. No pattern so far but the problem has worsened to the point it happens in nearly every call.

Our system is hosted with Cyberlynk and we have two trunks, one via SipStation and the other with Flowroute. We are running FreePBX 14.0.5.2 ‘VoIP Server’. Our main office has a 100/100 fiber line from our ISP and tests have never shown a glitch. We are using Sangoma phones on our end.

As a test I placed a call from an S705 (ext. 541) at my office off site via another ISP to the corporate office, also a S705 (ext. 8300). Around six minutes into the call our receptionist could no longer hear me (nor I her) for about 15-20 seconds.

I set this system up but am very new to VOIP and many of the terms. I do have over 30 years IT experience. I made a copy of the Asterisk log pertinent to that call and replaced all IP addresses for security (please advise if I missed something!)

…well it wouldn’t fit so an attempt at creating a link:
log file

well that link assumes you have a MS account…
try this one:
log file on filesanywhere

Posts: 5

Participants: 2

Read full topic

@kmumm111 wrote:

I am having an issue where a Yealink phone will not register. I have it set to extension 111 but when I reboot it I get registration failed in the phone and this message 4 times in the log.

[2018-11-28 11:58:31] NOTICE[4549]: res_pjsip/pjsip_distributor.c:525 log_failed _request: Request 'SUBSCRIBE' from '<sip:MAC0015657152f9@224.0.1.75>' failed for '10.100.10.203:5059' (callid: 26945@10.100.10.203) - No matching endpoint found

The device registered fine before it was rebooted. I never see it send the correct username.

Posts: 2

Participants: 2

Read full topic

@JohnCO wrote:

Hey guys, I have a few deployments running FreePBX 13 on Hyper-V. As a result I need to run the conversion tool. In the instructions it advises me to build a new install and activate it. I was wondering if I should activate it as a new deployment, or if I should deactivate and reuse the deployment ID from my source installs since I have commercial licenses tied to those deployments.

Thanks!

Posts: 3

Participants: 3

Read full topic

@sentinelace wrote:

I am calling into a mitel conference call. When I dial, it doesn’t take the dial tones. Usually these means the Sip provider is sending the wrong key strokes. For example 11111 when I only pushed 1. How do you trouble shoot this from cli?

Posts: 2

Participants: 2

Read full topic

@bajramia wrote:

Hi All,
When zulu ios app will be GA release.

Posts: 1

Participants: 1

Read full topic

@fropa wrote:

For some reason, I disabled asterisk from startup ( #systemctl disable asterisk) and then from a script I restart fwconsole. Everything works fine, but when I check asterisk status (systemctl status asterisk), there are some things I don’t understand.

[root@XX~]# systemctl status asterisk
● asterisk.service - LSB: Asterisk PBX
Loaded: loaded (/etc/rc.d/init.d/asterisk; bad; vendor preset: disabled)
Active: inactive (dead)
Docs: man:systemd-sysv-generator(8)
[root@XX ~]#

What does this mean?

Posts: 3

Participants: 2

Read full topic

@anscomputer wrote:

Hello,

I can’t make a originate call using AMI.

There is /etc/asterisk/manager.conf

[general]
enabled = yes
port = 5038
bindaddr = 127.0.0.1
#bindaddr = 0.0.0.0
displayconnects=no ;only effects 1.6+

[admin]
secret = 123456
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
write = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate,message
writetimeout = 5000

[ANS]
secret = 123456
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user,originate,agi
write = system,call,log,verbose,command,agent,user,originate,agi

[myasterisk]
secret = 123456
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user,originate
write = system,call,log,verbose,command,agent,user,originate

#include manager_additional.conf
#include manager_custom.conf

---

There is how i want to make a call

telnet 127.0.0.1 5038

action : login
username: admin
secret : 123456
Action: Originate
Channel: SIP/489
Exten: 2015
Callerid: 489
Priority: 1
Async: true

I have every time this message (for every users) :

Response: Error
Message: Permission denied

Can you help me ?

Best regards,
Luca

Posts: 1

Participants: 1

Read full topic

@Henry_power wrote:

How can i setup a rotating trunk within a outgoing route

Posts: 11

Participants: 4

Read full topic

@mvogel4949 wrote:

I followed the wiki directions for warm spare and they worked like a champ. I created the backup on the standby server, saved it, ran it and it worked. I wanted to go back into the backup I created a make a change but that backup I had created is not visible in the web gui. All I see is the default backup. When the backup restores does it erase the backup I created?

Posts: 7

Participants: 2

Read full topic

@Hansg wrote:

Hi I am using FreePBX 14.0.5.2 with Asterisk 13.19.1

I am having an issue where only calls within the last 2 days are showing up in the call history within the UCP. For example one extension is showing 69 records and the oldest one being Nov 27 12:35:04

When I check the CDR I can find calls and their recordings going back to when the system was first used.

Why is my UCP not showing these records?

I have also ran a repair on the database incase of any issues with that but everything was fine.

Posts: 1

Participants: 1

Read full topic

@SolarW wrote:

Hello.

I am use FreePBX Distro with
PBX Firmware 12.7.5-1807-1.sng7
and Asterisk 14.7.5
And i am can’t find
res_ari_mailboxes.so
res_stasis_mailbox.so
modules in my system.
Without this modules not work “mailboxes” in ARI (Asterisk REST Interface).

Where i can get this modules?

Posts: 3

Participants: 3

Read full topic

@kb9mfd wrote:

Is there a way to separate the mobile app / pc app from your desk phone. I like the app, but when I am at my desk they both ring, I would like the follow me like I have setup with a SIP client where I can have my desk phone ring first and if I do not answer in a time then ring the zulu mobile app. Right now when someone calls me I have my computer (also running zulu), my desk phone and my cell ringing at the same time. It’s a bit annoying. Thanks!

Posts: 1

Participants: 1

Read full topic

@digitalo wrote:

Hi Guys,

I’m investigating an issue. We are using the commerical module voicemail notifications that monitors a mailbox.
I’ve configured it so that on failure or on success an email is sent.
I’m getting complaints that sometimes no email is sent after listening and accepting the voicemail.
When I check the mail log I can see that no attempt was made to send an email.
How can this happen?
Here is part of the log which seamed relevant to me.

[2018-11-29 19:15:09] VERBOSE[3745] res_agi.c: AGI Script Executing Application: (NoOp) Options: (0)
[2018-11-29 19:15:09] VERBOSE[3745] res_agi.c: <Local/XXXXXXXXXX@vmnotify-main-00002b69;1> Playing 'silence/1.slin16' (escape_digits=) (sample_offset 0) (language 'en')
[2018-11-29 19:15:09] VERBOSE[3888][C-00000819] bridge_channel.c: Channel Local/XXXXXXXXXX@vmnotify-main-00002b69;2 left 'simple_bridge' basic-bridge <78bfd054-9a14-4204-8edb-4308de746cdc>
[2018-11-29 19:15:09] VERBOSE[3888][C-00000819] bridge_channel.c: Channel Local/XXXXXXXXXX@from-internal-00002b6a;2 left 'simple_bridge' basic-bridge <0971c1a1-9b30-4448-a68c-fe6ad3cf6562>
[2018-11-29 19:15:09] VERBOSE[3888][C-00000819] bridge_channel.c: Channel Local/XXXXXXXXXX@vmnotify-main-00002b69;2 swapped with Local/XXXXXXXXXX@from-internal-00002b6a;2 into 'simple_bridge' basic-bridge <0971c1a1-9b30-4448-a68c-fe6ad3cf6562>
[2018-11-29 19:15:09] VERBOSE[3888][C-00000819] bridge_channel.c: Channel Local/XXXXXXXXXX@from-internal-00002b6a;1 left 'simple_bridge' basic-bridge <78bfd054-9a14-4204-8edb-4308de746cdc>
[2018-11-29 19:15:09] VERBOSE[3748][C-00000819] app_macro.c: Spawn extension (macro-dialout-trunk, s, 24) exited non-zero on 'Local/XXXXXXXXXX@from-internal-00002b6a;2' in macro 'dialout-trunk'
[2018-11-29 19:15:09] VERBOSE[3748][C-00000819] pbx.c: Spawn extension (outbound-allroutes, XXXXXXXXXX, 7) exited non-zero on 'Local/XXXXXXXXXX@from-internal-00002b6a;2'
[2018-11-29 19:15:10] VERBOSE[3745] res_agi.c: <Local/XXXXXXXXXX@vmnotify-main-00002b69;1> Playing 'vmnotify-greeting.slin' (escape_digits=) (sample_offset 0) (language 'en')
[2018-11-29 19:15:13] VERBOSE[3745] file.c: <Local/XXXXXXXXXX@vmnotify-main-00002b69;1> Playing 'digits/4.slin16' (language 'en')
[2018-11-29 19:15:14] VERBOSE[3745] file.c: <Local/XXXXXXXXXX@vmnotify-main-00002b69;1> Playing 'digits/0.slin16' (language 'en')
[2018-11-29 19:15:15] VERBOSE[3745] file.c: <Local/XXXXXXXXXX@vmnotify-main-00002b69;1> Playing 'digits/3.slin16' (language 'en')
[2018-11-29 19:15:15] VERBOSE[3745] res_agi.c: AGI Script Executing Application: (NoOp) Options: (0)
[2018-11-29 19:15:15] VERBOSE[3745] res_agi.c: <Local/XXXXXXXXXX@vmnotify-main-00002b69;1> Playing 'vmnotify-instructions.slin' (escape_digits=123) (sample_offset 0) (language 'en')
[2018-11-29 19:15:19] VERBOSE[3745] res_agi.c: AGI Script Executing Application: (NoOp) Options: (rcinput:)
[2018-11-29 19:15:20] VERBOSE[3745] res_agi.c: AGI Script Executing Application: (NoOp) Options: (0)
[2018-11-29 19:15:20] VERBOSE[3745] res_agi.c: <Local/XXXXXXXXXX@vmnotify-main-00002b69;1> Playing '/var/lib/asterisk/sounds/vmnotify/403/29/msg0000.slin' (escape_digits=123) (sample_offset 0) (language 'en')
[2018-11-29 19:16:07] VERBOSE[3745] res_agi.c: AGI Script Executing Application: (NoOp) Options: (offset:)
[2018-11-29 19:16:07] VERBOSE[3745] res_agi.c: AGI Script Executing Application: (NoOp) Options: (pbinput:)
[2018-11-29 19:16:07] VERBOSE[3745] res_agi.c: AGI Script Executing Application: (NoOp) Options: (0)
[2018-11-29 19:16:07] VERBOSE[3745] res_agi.c: <Local/XXXXXXXXXX@vmnotify-main-00002b69;1> Playing 'vmnotify-instructions2.slin' (escape_digits=123) (sample_offset 0) (language 'en')
[2018-11-29 19:16:12] VERBOSE[3745] res_agi.c: AGI Script Executing Application: (NoOp) Options: (0)
[2018-11-29 19:16:12] VERBOSE[3745] res_agi.c: <Local/XXXXXXXXXX@vmnotify-main-00002b69;1> Playing 'vmnotify-thankyou.slin' (escape_digits=) (sample_offset 0) (language 'en')
[2018-11-29 19:16:13] VERBOSE[3745] res_agi.c: <Local/XXXXXXXXXX@vmnotify-main-00002b69;1> Playing 'vmnotify-goodbye.slin' (escape_digits=) (sample_offset 0) (language 'en')
[2018-11-29 19:16:14] VERBOSE[3884][C-00000819] bridge_channel.c: Channel SIP/TELENET-00001be7 left 'simple_bridge' basic-bridge <0971c1a1-9b30-4448-a68c-fe6ad3cf6562>
[2018-11-29 19:16:14] VERBOSE[3746][C-00000819] bridge_channel.c: Channel Local/XXXXXXXXXX@vmnotify-main-00002b69;2 left 'simple_bridge' basic-bridge <0971c1a1-9b30-4448-a68c-fe6ad3cf6562>
[2018-11-29 19:16:14] VERBOSE[3746][C-00000819] pbx.c: Spawn extension (vmnotify-main, XXXXXXXXXX, 2) exited non-zero on 'Local/XXXXXXXXXX@vmnotify-main-00002b69;2'
[2018-11-29 19:16:14] VERBOSE[3745] res_agi.c: <Local/XXXXXXXXXX@vmnotify-main-00002b69;1>AGI Script vmnotify-main.php completed, returning 4
[2018-11-29 19:16:14] NOTICE[3745] pbx_spool.c: Call completed to Local/XXXXXXXXXX@vmnotify-main
[2018-11-29 19:17:44] VERBOSE[4242] pbx_spool.c: Attempting call on Local/1@vmnotify-dummy for application AGI(vmnotify-audit.php) (Retry 1)
[2018-11-29 19:17:44] VERBOSE[4247][C-0000081a] pbx.c: Executing [1@vmnotify-dummy:1] Answer("Local/1@vmnotify-dummy-00002b6b;2", "") in new stack
[2018-11-29 19:17:44] VERBOSE[4242] dial.c: Called 1@vmnotify-dummy
[2018-11-29 19:17:44] VERBOSE[4242] dial.c: Local/1@vmnotify-dummy-00002b6b;1 answered
[2018-11-29 19:17:44] VERBOSE[4242] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/vmnotify-audit.php
[2018-11-29 19:17:44] VERBOSE[4242] res_agi.c: AGI Script Executing Application: (NoOp) Options: (1)
[2018-11-29 19:17:44] VERBOSE[4242] res_agi.c: <Local/1@vmnotify-dummy-00002b6b;1> Playing 'vmnotify-too-late.slin' (escape_digits=) (sample_offset 0) (language 'en')
[2018-11-29 19:17:44] VERBOSE[4247][C-0000081a] pbx.c: Executing [1@vmnotify-dummy:2] Wait("Local/1@vmnotify-dummy-00002b6b;2", "300") in new stack
[2018-11-29 19:17:48] VERBOSE[4242] res_agi.c: <Local/1@vmnotify-dummy-00002b6b;1>AGI Script vmnotify-audit.php completed, returning 4
[2018-11-29 19:17:48] NOTICE[4242] pbx_spool.c: Call completed to Local/1@vmnotify-dummy
[2018-11-29 19:17:48] VERBOSE[4247][C-0000081a] pbx.c: Spawn extension (vmnotify-dummy, 1, 2) exited non-zero on 'Local/1@vmnotify-dummy-00002b6b;2'

My system is completely up to date and I’m on v14, asterisk 13

Posts: 1

Participants: 1

Read full topic

@hardocp wrote:

Continuing the discussion from Asterisk pjsip sip tls:

The old thread got closed while i was working on a solution so i am continuing it here

After a bunch of back and forth with Yealink I had an epiphany about this issue this morning.

Yealink told me that the reason why the softphone was working but their phone was not was because the softphone only required 1 way authentication but Yelinks phone required 2 way authentication

So it hit me – I realized that there are 2 parts to the SSL connection for 2 way authentication – the server part and the client piece.

I had – as per the thread linked – setup the server certificates through Freepbx – both a Comodo and a LE certs

What i was missing was the client cert

So as per the wiki https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial I created a client file

Example: ./ast_tls_cert -m client -c /etc/asterisk/keys/ca.crt -k /etc/asterisk/keys/ca.key -C phone1.mycompany.com -O “My Super Company” -d /etc/asterisk/keys -o malcolm (obviously making the requisite changes to fit my scenario)

Then as per the wiki i downloaded both the server and client pieces into the phone as follows:

Security tab has 2 sections –

Trusted Certificates – client piece
Server Certificates – server piece

First i uploaded the client cert to trusted certificates and the server cert to server certificates

The under trusted certificates i changed the CA certificates to All certificates
and under Server Certificates i hanged device certificates to Custom Certificates

Confirm – and reboot and it looks like we are in business!

The endpoint seem to work regardless of whether the setting for only accept trusted certificates is set to enabled or disabled

So there you have it – TLS and SRTP on a Yealink T5x series

thank you to all who helped – this was a really tough one that took me a while to work out – very satisfying in the end to get it working

I can now finalize my PBX in the cloud knowing that remote endpoints can connect securely to the hosted server

Posts: 1

Participants: 1

Read full topic

@rhall wrote:

SHMZ release 6.6 (Final)
Asterisk 13.8.1
PBX Firmware:10.13.66-11
PBX Service Pack:1.0.0.0

Trying to upgrade pm2 to 13.0.5.1
It kept telling me to yum upgrade nodejs
I got this error:
Error: Cannot find a valid baseurl for repo: dag

Content of dag.repo
[dag]
name=DAG RPM Repository
baseurl-http://repository.it4i.cz/mirrors/repoforge/redhat/el5/en/x86_64/rpmforge/RPMS
gpgcheck=1
enabled=1

Why it’s pointing to the Czech Republic is beyond me.

At one point someone said to uninstall nodejs and then install it.
I uninstalled it get the same error trying to install it.

Is there a different Baseurl I should be using?

Posts: 1

Participants: 1

Read full topic

@muhammadbilal wrote:

Hi Guys,
I have some self installed unsigned modules and also custom changes / development in code of freePBX and i don’t want it to be updated because it will revert all changes i’ve done in backend. It actually updated the frame work under vulnerable to security threats which reverted all my work which ive done.

Is there any way i can stop every type of update on FeePBX / server and sure myself it will not happen

Thanks guys.

Posts: 1

Participants: 1

Read full topic