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NAT: global, per extension or only in the trunk definition?

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@dbet wrote:

In FreePBX there is the NAT setting for each extension, but also global under Asterisk SIP Settings. What are these two attitudes to each other? In the trunk, NAT can be defined also. Can NAT be deactivated in the global settings and enabled in the trunk?

My FreePBX is behind a router and thus behind NAT. The phones are in the same network as the FreePBX and thus they do not need NAT. Therefore I assume NAT in the global setting should be true and on the extensions false.

But I was confused by the following comment from the sip.conf. He seems to say that if the global option NAT is turned on, it also has to be switched on for the extensions NAT and vice versa. Otherwise it could be a security problem.

IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
; the nat setting in a peer definition, then the peer username will be discoverable
; by outside parties as Asterisk will respond to different ports for defined and
; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
; other, then valid peers with settings differing from those in the general section will
; be discoverable.

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Audio problem between two sip trunk interfaces and wan / lan

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@linkjet wrote:

Dears, I have a problem and I hope you can help me

I have two network interfaces on my server

eth0 -> LAN AND INTERNET
eth1 -> to my sip provider

previously I could make or receive calls without problems from and to the internet with softphone or by VPN with the public ip that goes to the eth0 network, this was until I changed the parameter (external address) because I changed my analog lines to digital and in this parameter I had the public ip that goes to the eth0 network and I changed it to the public ip that has the inferred eth1 due to audio problems, obviously I preferred to use the ip of eth1 since there are cell phone calls coming and going , local phones, other companies etc.

image

now the phones that are on the internet or vpn register to the server, make and receive calls, but do not have audio, it is probably a routeo / nat problem but I don’t know how to solve it.

Has anyone had any experience with this problem?

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FreePBX architecture

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@sb_asterisk wrote:

Hello guys,

I have been working with Asterisk for many years but just started looking into FreePBX for one of my work.
Basically I am interested in knowing FreePBX core architecture, how it interacts with Asterisk and Database.
I did search on forum and found this post where tonyclewis said

FreePBX writes everything to a database tables. Then when you run the Apply Config from the GUI it takes everything in the database, writes out .conf files and reloads asterisk.

Is there any other details available on architecture other than this?

Thanks,

P.S. I’m a new user and hence not allowed to put any link in my post.

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Grandstream gxp 2140 MWI inconsistent messages waiting

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@cfapress wrote:

We use Grandstream phones with FreePBX because they’re much more affordable than others. But we have erratic behavior with them.

This time … the MWI (message waiting indicator) lamp sometimes blinks after a message is left. Other times it waits several days before blinking.

I’ve tried adjusting the Extensions > someextnum > Advanced > MWI Subscription Type …

with Auto - it sometimes works properly
with Solicited - it sometimes works properly

Can anyone suggest other settings to check - either in FreePBX or on the Grandstream phone?

Thanks in advance, Jason.

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FreePBX 14.0.13.6 - No menu bar - possible hacked

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@davidber wrote:

I have a client that uses Flowroute for their SIP provider. He got a message asking if he wanted to block calls to the UK. He is in the US and wanted to know if that was spam or something else. I logged into his Flowroute account and noticed that over the last two days there were a lot of 2 minute phone calls. I pulled a CDR report and noticed that the origin IPs were not from his PBX. Ok, this looked interesting to me. I wondered . . . Hmm. Did his PBX get hacked. I went to log into his PBX and saw the missing menu bar. I get the system overview and feed modules, but the blue bar above has NO menus.

Ok, that seemed odd, so I logged into the box. I did a yum update then a yum upgrade then a fwconsole ma updateall. Still no menus. Then I saw another thread and the person suggested doing a fwconsole ma upgrade framework. I got a message stating I had the latest framework.

So I realize that this is really a two part question and they are somewhat related.

  1. How do I get my menu back?
  2. How do I determine if the server was hacked into. I have gone to Flowroute and whitelisted US calls only.

Thanks,

dave

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Call forwarding and caller ID

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@Scion wrote:

This is probably elementary for some of you guys…

I have a user who has a DID to his phone that prepends the CID with “sales.” He knows if someone dials his DID because of this, opposed to queue calls, etc.

When this user forwards his phone and someone calls his DID. I don’t want to pass the originating caller ID. I want him to know someone dialed his DID.

I tried find me follow me hoping to produce a different outcome, but no dice.

Thoughts, suggestions, solutions, appreciated.

Jason

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No calls, probably problem with ports opening

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@l4cky wrote:

When I tried to dial, I have no incoming or outgoing calls.

Only after a timeout, it says unable to reach the number. Call drops. Can’t hear or say anything.

I have been able t get around this by opening the ports through my router and have to manually restart it by unplugging the power.

However, now I have a new external IP and by cloning the mac address of my pc to my router.

Since then, nothing works. I went to pbx and update the external ip, still not working.

Funny thing is I didn’t touch the ports forwarding tab, which used to work… I manually restart several times, not working.

I have no idea how to trace a call (if someone can give me the newbie detailed instructions on how to do it…

I provide the screenshot of my configuration.

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Message when an extension is in use and after 6 rings

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@Dujith wrote:

I would like to be able to play a message when a call does not get picked up after a few rings (or time).
Or if 1 of the extensions is in use.

We have a small office with 3-4 ppl. When there is an incomming call and someone is busy the chances are high that it will not get picked up at once. At many times there will be only 1 or 2 at the office.
So i would like for a message to play when 1 of the extensions is in use to let ppl know they might have to wait longer.

The message after a few rings is something i can find, the extension in use condition i cannot find anything for.

Any advice or nudge in the right dirrection would be welcome.

Edit: Is Announcement the correct one for the message? i noticed it will stop ringing when playing back the message.

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How to connect to freepbx remotely to do pbx management

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@irfanbhai wrote:

I am new to Asterisk free pbx, I am just setting up my lab to test the pbx managemt remotely.
I need help to configure my Asterisk FreePBX and how to connect frm other system to do pbx telemanagement

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Technical information on the CDR report

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@Cricchetto wrote:

Hello, I need help to understand what all these calls are, the firewall is activated and also the file 2ban. It’s a normal thing? Can they be blocked? Need to lock?

Thanks

Sun, 3 Nov 2019 12:43 1572781392.237 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:43 1572781389.236 60345 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:43 1572781386.235 12955 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:43 1572781385.234 8225 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:43 1572781384.233 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:42 1572781377.232 70605 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:42 1572781375.231 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:42 1572781374.230 291 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:42 1572781372.229 90453 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:42 1572781366.228 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:42 1572781357.227 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:42 1572781347.226 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:42 1572781346.225 60344 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:42 1572781344.224 8225 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:42 1572781344.223 12955 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:42 1572781339.222 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:42 1572781339.221 70605 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:42 1572781330.220 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:42 1572781329.219 90452 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:41 1572781316.218 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:41 1572781304.217 8225 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:41 1572781303.216 60344 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:41 1572781302.215 12955 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:41 1572781299.214 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:41 1572781296.213 70605 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:41 1572781289.212 90452 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:41 1572781278.211 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:41 1572781269.210 290 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:41 1572781262.209 8225 Congestion s [from-sip-external] ANSWERED 00:11
Call Date Recording System CallerID Outbound CallerID DID App Destination Disposition Duration Userfield Account CDR Table CDR Graph
Sun, 3 Nov 2019 12:41 1572781261.208 12955 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:41 1572781260.207 60344 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:40 1572781258.206 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:40 1572781257.205 70605 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:40 1572781249.204 90452 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:40 1572781236.203 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:40 1572781225.202 5004 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:40 1572781221.201 8225 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:40 1572781219.200 70605 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:40 1572781219.199 12954 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:40 1572781217.198 60344 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:40 1572781208.197 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:40 1572781208.196 90452 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:39 1572781184.195 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:39 1572781179.194 8224 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:39 1572781177.193 12954 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:39 1572781175.192 70604 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:39 1572781174.191 60344 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:39 1572781174.190 290 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:39 1572781168.189 90452 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:39 1572781161.188 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:39 1572781140.187 8224 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:38 1572781137.186 70604 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:38 1572781136.185 12954 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:38 1572781133.184 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:38 1572781127.183 90451 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:38 1572781123.182 60343 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:38 1572781114.181 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:38 1572781101.180 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:38 1572781100.179 8224 Congestion s [from-sip-external] ANSWERED 00:11
Call Date Recording System CallerID Outbound CallerID DID App Destination Disposition Duration Userfield Account CDR Table CDR Graph
Sun, 3 Nov 2019 12:38 1572781099.178 70604 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:38 1572781095.177 12954 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:38 1572781089.176 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:38 1572781087.175 90451 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:38 1572781080.174 60343 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:37 1572781077.173 290 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:37 1572781077.172 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:37 1572781063.171 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:37 1572781056.170 70604 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:37 1572781054.169 12954 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:37 1572781053.168 8224 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:37 1572781052.167 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:37 1572781046.166 90451 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:37 1572781039.165 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:37 1572781037.164 60343 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:37 1572781026.163 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:36 1572781018.162 70604 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:36 1572781013.161 8224 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:36 1572781012.160 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:36 1572781010.159 12953 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:36 1572781006.158 90451 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:36 1572781001.157 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:36 1572780995.156 60343 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:36 1572780987.155 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:36 1572780981.154 290 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:36 1572780979.153 70603 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:36 1572780975.152 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:36 1572780972.151 8223 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:36 1572780969.150 12953 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:36 1572780965.149 90451 Congestion s [from-sip-external] ANSWERED 00:11
Call Date Recording System CallerID Outbound CallerID DID App Destination Disposition Duration Userfield Account CDR Table CDR Graph
Sun, 3 Nov 2019 12:35 1572780951.148 60343 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:35 1572780950.147 0081 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:35 1572780940.146 70603 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:35 1572780933.145 8223 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:35 1572780927.144 12953 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:35 1572780923.143 90450 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:35 1572780907.142 60342 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:35 1572780902.141 70603 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:34 1572780895.140 8223 Congestion s [from-sip-external] ANSWERED 00:11
Sun, 3 Nov 2019 12:34 1572780886.139 12953 Congestion s [from-sip-external] ANSWERED 00:11

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Inbound/Outbound calls using existing cellular service and Cisco phone through FreePBX?

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@VoiceNetwork wrote:

I apologize in advance for the length of this post and if the topic was already covered (I searched as well as I could with limited time). I purchased network equipment years ago (Cisco 2950’s/7960G/1841’s). I finally setup my rack and installed/configured a FreePBX/Asterisk server on the network. After using a time machine to travel back to the early 2000’s - I figured out how to setup a 7960G, tftp, dhcp, ntp, voice network,etc. Something I had not done from scratch before. My network is purring - dynamic routing protocols, tftp, dhcp, dns, firewalls galore…and a solid voice network.

After doing some research, I came across sip-ua. That allowed me to create a SIP Trunk for testing. I subsequently created extensions, outbound routes, etc. I have a dial plan, directory, soft phone using Zoiper and some others. Everything seems to work just fine and I think I understand how things work now.

I should mention that my main goal is studying for the CCNA (finally doing getting it before the February change). However, I wanted to have a voice network for testing/learning, but also to make calls without the small form factor of my pocket sized iPhone SE. I also like the sound of my desk phone and soft phone better on occasion. I like the soft phones (a few) on the PC, occasionally the iPhone if I need to with Zoiper, and a CTU24 ring tone loaded Cisco phone on my desk (as I age).

Equipment Summary:

  1. FreePBX/Asterisk Server (FreePBX 14 - OnPrem)
  2. Cisco IP Phone 7960G
  3. iPhone SE (Wi-Fi Calling Enabled)
  4. PAYG celluar plan
  5. Various Cisco Ethernet Network Equipment
  6. Wireless Network (Separate)

So, my question(s).

  1. Is it possible to make a call using the 7960G (desk phone) through my cellular carrier (existing PAYG plan) without utilizing a SIP Trunk? e.g. somehow use the iPhone SE as a proxy to cellular service and avoid needing a SIP trunk at all?
  2. Would this be Wi-Fi or can I use my existing wired network somehow?
  3. I have not fully researched Wi-Fi calling, so please feel free to enlighten me. However, I don’t mind if the connection is over 3G/4G at this time. Wi-Fi is preferred, but not a must have here.

Hopefully this is further clarification. I know right now I can hook up a “real” SIP trunk like UniTel Voice or when I find out what the rates are for SIP-UA, maybe them (waiting on an email). However, I am hoping to essentially use my hard wired desk phone or PC softphone to utilize the GSM or Wi-Fi network calling I already pay for and also utilize FreePBX/Asterisk in the process. If perhaps the suggestion is to use a “SIP2GSM” hardware device (additional purchase), please let me know that this is the only option with suggestions on model and configuration. However, I am looking for a way to avoid paying additional fees. I might consider a one time purchase of a hardware device if it is reasonable and take my GSM chip out when I need to make calls (I don’t mind that). I also may pay for UniTel if this doesn’t work out. The current SIP-UA connection is just for testing and the calls cut out super quick (since they are just for testing).

Thank you for your time. :slight_smile:

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No CDR written with default installation of FreePBX 15 + Asterisk 16 on Debian 9

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@watoan wrote:

Fresh installed FreePBX 15 with Asterisk 16 on Debian 9 using the official wiki guide

Everything smooth during the installation, ODBC connection is fine, but no CDR written neither in MySQL asteriskcdrdb table nor CSV file.

root@pbx:/usr/src# isql -v MySQL-asteriskcdrdb
+---------------------------------------+
| Connected!                            |
|                                       |
| sql-statement                         |
| help [tablename]                      |
| quit                                  |
|                                       |
+---------------------------------------+
SQL> select 1
+---------------------+
| 1                   |
+---------------------+
| 1                   |
+---------------------+
SQLRowCount returns 1
1 rows fetched
SQL>


Connected to Asterisk 16.6.1 currently running on pbx (pid = 1424)
pbx*CLI> odbc show asteriskcdrdb

ODBC DSN Settings
-----------------

  Name:   asteriskcdrdb
  DSN:    MySQL-asteriskcdrdb
    Number of active connections: 1 (out of 5)
    Logging: Disabled

pbx*CLI> cdr show status

Call Detail Record (CDR) settings
----------------------------------
  Logging:                    Enabled
  Mode:                       Simple
  Log unanswered calls:       No
  Log congestion:             No

* Registered Backends
  -------------------
    csv
    cdr_manager (suspended)
    Adaptive ODBC


root@pbx:~# cd /var/log/asterisk/cdr-csv/
root@pbx:/var/log/asterisk/cdr-csv# ll
total 8.0K
drwxrwxr-x 2 asterisk asterisk 4.0K Nov  4 10:47 .
drwxrwxr-x 5 asterisk asterisk 4.0K Nov  4 11:53 ..
root@pbx:/var/log/asterisk/cdr-csv# ls -l
total 0

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CDR using "start" column instead of "calldate" column in database

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@DaBorg wrote:

Hi,

Newer versions of Asterisk (13.21) like to log data in the “start”,“answer” and “end” columns in the database, but the cdr module in freePBx (v13.0.195.1) is hard-coded to use the calldate column. Eg:

$query = "SELECTcalldate,clid,did,src,dst, ...

I’m aware I can use the the alias directive in cdr_adaptive_odbc.conf to make the ODBC CDR module log to the calldate field, but there is other software on the server which is built around the “start” column.

All I really want to ask is: Is there a version of the FreePBX CDR module which can use the “start” column instead of “calldate”? I tried updating the source code myself but then FreePBX gave scary-looking warnings about modules being tampered with.

Thanks.

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Asterisk 17

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@dux wrote:

Asterisk 17 has already been released. When will it be available in FreePBX installation repository?

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DID manipulation on incoming pjsip Trunk

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@kyriakos33 wrote:

Hello to all.

Is there a way that I can prepend a couple of digits at an incoming pjsip trunk DID before I pass it to my inbound route.

The case scenario is the below:
Two carriers will both send me calls at a DID number 2222222222. I want at the calls of the first carrier that come from the 1st pjsip trunk to prepend 11 in front of the 2222222222 so that I have 112222222222 DID and make an inbound route so that I can route the call to my sangoma A104 port1.
At the calls from the second carrier which come form the pjsip trunk 2 I want to prepend 33 infront of the DID so that I will have the DID 332222222222 and create another inbound route to route the call in my A104 port 2.

The easy way of course would be to ask my two carriers to prepend the digits and send me the final DID’s but I’m looking for an alternative in case they can’t.

Thank you in advance for any information.

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Freepbx 13, CID via cisco trunk suggestions

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@nick_radaev wrote:

Dear Freepbx community, I am not an expert in freepbx therefore i am looking for some advice to point me in the right direction, any advice would be greatly appreciated.

I am Running Freepbx 13 and i have 2 trunks, one with our local office running cisco cme and the second one is our telecommunications provider sip trunk service.

My problem is when i set a followme on my freepbx to an external city number, Freepbx forwards the user extension from cisco cme which for example is 200 via external trunk.

This is clearly not a problem of freepbx but my lack of knowledge because freepbx doesn’t have any rule in place to handle cme ext that call outside, I don’t know how to set a DID number or range for numbers coming from CME.

Within freepbx i set a cid number via Applications, Extensions, outbound cid but i don’t understand how i can do it with extensions that are coming from another trunk like in my case CME.

How can i assign an extension calling from CME trunk to Freepbx a CID number within Freepbx system?

Sorry if i am not being very clear and once again any help would be greatly appreciated. Tnx

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Connect freepbx to GSM module

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@AmidouFlorian92 wrote:

Hi
I need some configuration about a UMTS GSM OfficeRoute with freepbx
I’m reading a manual for the 2N OfficeRoute.It’s a equipment that can hold 4 sims cards and has hos own admin GUI (configuration of
extentions of IVR…).
The thing I want to make that is only to use this GSM equipment for receiving a call on the numbers that I’ll insert on the Office Route https://www.2n.cz/officeroute
All the others configurations must be done using using freepbx (Such as extensions,Inbound and Outbound Routes).
How can I connect this equipment to freepbx ? And how can I configure freepbx to receive calls directly on the SIMS cards of my OfficeRoute?
Thanks

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Upgrade only Warm Spare to Freepbx 15?

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@ebctechsupport wrote:

I have two Freepbx Phone System 60s, with one running as a warm spare. The warm spare backs up the primary server every night. I saw a video where it is possible to restore backups across versions as of Freepbx 15. Would it be possible for me to update my warm spare to v15 for testing and leave my primary server on v14, leaving me running a warm spare on v15 running a nightly backup and restore of my primary server running v14?

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Unable to dial extension

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@ChuckCoggins wrote:

Hello everyone,

Today I am experiencing an issue with our phone system.
Everything is working correctly - I am able to call out, customers can call in.
However, on our IRV customers are able to dial an extension to directly connect with people.

Whenever you dial anyone extension it auto directs you to another extension instead it always happens to be the same extension as well no matter what extension you dial.

I went into Reports > Asterisks Info >
I see that PJSip Endpoints
Available: 24
Unavailable: 23

I think this is likely the issue, but I have no idea how to re-register all the phones.
Would anyone be able to help me with this, please?

Thank you!

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How can I set priorities to extensions?

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@roespimo wrote:

Hello All,

I just want to know how I can set priorities for extensions. In this way the extension can take priority when a call goes into a queue.

My idea is to have a queue with 30-40 dynamic agents, but we want these agents to have a different “skill or priority” to answer calls. For example: 10 of those agents have the highest priority because they are our best sellers. Another 10 agents have a different priority and the other 10 have another priority.

I was checking the extension settings and queue settings, but I don’t get anything about priorities or skills.

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