@unison wrote:
Hi all
Does anyone know of a way to see the number of concurrent calls to an inbound DID number within the dial plan? One of the DID's is 300, so I was looking along the lines of the following:
exten => s,n,Set(GROUP()=${300})
exten => s,n,NoOp(Number of calls to this DID: [${GROUP_COUNT(${300})})Cheers
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@tdixon81 wrote:
Hi, there knowers of VoIP. I have FreePBX 14.0.1.1 running on a VMWare VM w/ 2 CPU, 3 GB, and a 50 Mbit asynchronous fiber line with ALG disabled on our Sonicwall. Phone system is working very well when dialing into conference lines, making phone calls to cell phones, etc.,..for the most part. One caller in particular with a Salisbury, MD number, every time her and I talk on the phone there is a choppy sound during our call. I do not get this when I make an outbound or receive an inbound call with east coaster located in the DC area on her mobile or office lines.
I suspect that I'm dealing with a codec issue during negotiation between the two system when experiencing the choppy calls. Thoughts?
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Participants: 1
@Janthro wrote:
Hey guys, I have a big problem. My phone server hardware failed on me and I had to rebuild my FreePBX server from scratch. Everything with inbound and outbound calls seem to be working but when we try to transfer someone directly to their voicemail with the Polycom phones we have it just disconnects instead of going to their voicemail.
I am not saavy enough about things to figure this out. I have read numerous posts about this problem and none of them have fixed mine. We can transfer to the person just fine, and we can get their voicemail if they don't answer, but it drops when transferring directly to their voicemail.
Anyone have experience with this and might know what I need to do for this?
I am running FreePBX 14.0.1.4 direct from CD install.
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@ajr3v wrote:
Is there a way that I can hard set DSCP values globally?
I am using pjsip and have been able to set SIP DSCP values in pjsip.transports_custom_post.conf successfully as followed
[0.0.0.0-udp](+) tos=0xB8 cos=184And i Have been able to set individual extensions to the correct values in pjsip.endpoint_custom_post.conf as followed
[74876](+) tos_audio=ef cos_audio=5I have many extensions and to go through and add them one by one would be tedious, not to mention anytime a user is added, it will need to be done.
Is there another _custom.conf file that I can edit and set this globally to all extensions?
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@mvogel4949 wrote:
Is there a maximum number of endpoints you can associate with a single pjsip extension? Also would I run into issues if one of the endpoints was remote while the other was local to the freepbx system? Thanks!
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@chrischevy wrote:
Whenever a temporary greeting is present, VmX options are ignored.
I press 2 and the DTMF is ignored, as seen in the log below[2017-09-20 16:03:08] VERBOSE[18707][C-00008d0e] file.c: -- <SIP/206-0000628d> Playing '/var/spool/asterisk/voicemail/default/206/temp.slin' (language 'fr') [2017-09-20 16:03:10] DTMF[18707][C-00008d0e] channel.c: DTMF begin '2' received on SIP/206-0000628d [2017-09-20 16:03:10] DTMF[18707][C-00008d0e] channel.c: DTMF begin ignored '2' on SIP/206-0000628dSome users record a temporary message before going on vacation and they say that they can be reached on their cellphone by pressing 2 but it doesn't work. It does work when the temporary message is removed.
Is this a normal behavior ?
There is an old bug report that I think was related to this problem back in 2008
https://issues.freepbx.org/browse/FREEPBX-2650EDIT:
Also found this: https://issues.freepbx.org/browse/FREEPBX-5818
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@steve_pbuk wrote:
Hello
I have recently started to look at FreePBX and so far I like what I see. I can also see that a perfect phone to pair up with FreePBX is a Sangoma model.
If possible I would like help with the below general questions I have:
Has anyone got any thoughts/feedback for hosting FreePBX within an AWS instance rather than on site. Does this work well with handsets registering with a PBX in the cloud?
I understand the main benefit to enable the VPN feature for the handsets, that is, the extra layer of security. Are there any downsides when using the VPN feature? Does it affect call quality? Do handsets "talk" directly to each other or does ALL traffic go through the VPN to the PBX?
Is the VPN feature an SSL VPN or a IPSEC VPN. Does it use standard 443 ports, meaning it will work with 99% of firewalls by default?
By default can users "login" to handsets or does this require extra configuration? Is it complex?
How well do people rate Sangoma for things such as technical support and firmware security updates?
I'm sure I will have many more questions as I dig deeper but would appreciate any feedback from the community at this stage.
Thanks
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Participants: 2
@dwright1542 wrote:
Does anyone have a suggestion on a client friendly CDR reporting tool? They just want some basic stats, calls in, calls out etc. I tried to show them the current tool, and admittedly, it was a mess for an office manager.
Even a simple inbound call to a ring group, answered, parked, and picked back up shows a whole slew of entries in the CDR.
I read some old threads about using CEL, but that seems overkill for this.
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Participants: 2
@adtopkek wrote:
Our CDR stopped working. Trying to figure out whats going on with it. I can login to the Mysql database with
Its running the FreePBX 13 Distro: 10.13.66-21
Asterisk 13.17.1
I can connect to MySQL with the freepbxuser and the associated password and I can access the database asteriskcdrdb.
Call Detail Record (CDR) settings
Logging: Enabled
Mode: Simple
Log unanswered calls: No
Log congestion: No
* Registered Backends
cdr_manager cdr-custom res_config_sqlite Adaptive ODBC
Tried loading the cdr adaptative while atserisk was running and:
module load cdr_adaptive_odbc.so
Unable to load module cdr_adaptive_odbc.so
Command 'module load cdr_adaptive_odbc.so ' failed.
[2017-09-21 15:52:27] WARNING[18312]: loader.c:1042 load_resource: Module 'cdr_adaptive_odbc.so' already exists.
Fwconsole restarted and:
Loading cdr_csv.so.
== Parsing '/etc/asterisk/cdr.conf': Found
Loading cdr_manager.so.
[2017-09-20 23:27:07] WARNING[5630]: cdr_manager.c:214 load_config: Failed to load configuration file. Module not activated.
== Unregistered 'cdr_manager' CDR backend
Loading cdr_odbc.so.
[2017-09-20 23:27:07] WARNING[5630]: cdr_odbc.c:205 odbc_load_module: cdr_odbc: Unable to load config for ODBC CDR's: cdr_odbc.conf
Loading cel_custom.so.
== Parsing '/etc/asterisk/cel_custom.conf': Found
Added CEL CSV mapping for 0 files.
== cel_custom.so => (Customizable Comma Separated Values CEL Backend)
Loading cel_manager.so.
== Parsing '/etc/asterisk/cel.conf': Found
== Parsing '/etc/asterisk/cel_general_additional.conf': Found
== Parsing '/etc/asterisk/cel_general_custom.conf': Found
== Parsing '/etc/asterisk/cel_custom_post.conf': Found
== cel_manager.so => (Asterisk Manager Interface CEL Backend)
Loading cel_odbc.so.
== Parsing '/etc/asterisk/cel_odbc.conf': Found
== Parsing '/etc/asterisk/cel_odbc_custom.conf': Found
-- Found CEL table cel@asteriskcdrdb.
== cel_odbc.so => (ODBC CEL backend)
Loading cdr_syslog.so.
[2017-09-20 23:27:07] ERROR[5630]: cdr_syslog.c:145 load_config: Unable to load cdr_syslog.conf. Not logging custom CSV CDRs to syslog.
Loading cdr_custom.so.
[2017-09-20 23:27:07] ERROR[5630]: cdr_custom.c:101 load_config: Unable to load cdr_custom.conf. Not logging custom CSV CDRs.
== cdr_custom.so => (Customizable Comma Separated Values CDR Backend)Can you think of anything that might be stopping CDR's from being logged?
Is the Adaptive ODBC the only backend I need for it to log into CDR reports and show up properly in the UCP? My Freepbx 14 server only has the Adaptive enabled.
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Participants: 3
@oscarenzo wrote:
Hello,
Actually I'm working with FreePBX 12.0.76.4 and have configured fail2ban to block the IP addresses when try to access to extension by brute force, after work send email by default, i would like to know if is possible add a custom command when do this, i would to block the ipaddress by iptables also.
Thank you advance, best regards.
Posts: 2
Participants: 2
@dsubs wrote:
Hi all, Freepbx 12 asterisk 11.21
Remote phone with TLS (transport=all, TLS primary) registers with the phone system.
This phone is able to make calls to extensions that have transport=All, UDP primary. But, tls phone to another tls phone fails. Both have transport=all. TLS primary. Also, calls from other "all, UDP primary" extensions to this tls phone fails.SIP error code given is 488 not acceptable here...
any ideas?
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@netchester wrote:
My question is very simple.
How can I install OSS Endpoint Manager on FreePBX freshly install on Centos 7.
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@kwriley87 wrote:
I am using FreePBX 12 on Asterisk 11.
I've gotten a few notifications of 50+ SIP attempts against one of my FreePBX boxes the past couple of days. Initially, I added this IP subnet into the IP Tables rules list to deny further connections, but I received a notification this morning that this same IP address was banned again after 97 SIP attempts, even though this IP address should already be banned. I checked my IP Tables rule list and this IP is listed in there.
Why would Fail2Ban say that it has banned an IP but that IP is still allowed connection attempts to my system?
I'm not sure if this matters but my machine is running SSH on port 20022 instead of the standard port.
Any feedback is appreciated! Thanks!
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Participants: 1
@chrischevy wrote:
Quick question: Can I install FreePBX with my default customizations and use it as a "ghost" to install new systems ?
Is there is a unique ID assigned during installation or is it just at the activation ?I would like to be able to install FreePBX quicker by just copying my initial installation (which is on a virtual hard drive). The activation would be done afterwards.
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Participants: 2
@sentinelace wrote:
When I want to transfer to an extensions number I dial * then extension. I get “please leave a message” and notbthe personal greeting. When I try to use. BLF key I get a recall. How is his supposed to be setup? I am using yealink T29G
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@intravista wrote:
OK, before I start... I've read the forums, and did a little research... So I know that Vonage with FreePBX is not common for several reason... BUT...
I have a gandfatherd vonage plan that is ridicuously cheap & thought I've outgrown the need for 1 single line, and now have a FreePBX install to better meet those needs... I also don't really want to give up my vonage line... And I'd really like to not have 2 phones on my desk.
Mostly, I use the vonage line for outbound calls since it does have unlimted calling.
One thing to note, is that my PBX is hosted offsite.
So my questions are...
A: Can I use something like an fxo to plug the vonage device phone line into, and then set that fxo to connect to my freepbx server? I've seen to cisco fxo's that seem to be for that purpose (ie: SPA3000)?
B: Any risks in doing so? Quality or security?
C: Am I missing something? I don't think I have seen this Idea anywhere else.. So Maybe I'm missing something... I am somewhat inexperienced.
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Participants: 1
@etgllc wrote:
Need some recommendations or suggestions on SIP providers for a single office with 5 DIDs and average 25-30 calls per 8 hour day. First time setting up FreePBX and we're at the stage of ordering SIP service and activating it. I've been reading materials from Twilio, Flowroute, Nextiva, and SIPStation. I think for a small shop like ours, the pay as you go model might be the best. Any recommendations are greatly appreciated.
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Participants: 2
@msarti wrote:
Hi,
I made a fresh installation from FreePBX Distro ISO. The installation is in an internal (private) LAN and have a trunk configurated with outbound and inbound routes.
For now I use a softphone (Ekiga) for test.
I have created an extension 101, of type PJ-SIP, and associate it to inbound route. This works well, softphone is registered and rings when an external call coming.
When I try to make an outbound external call from the softphone, I obtain the following error:[2017-09-24 08:47:00] NOTICE[4101] res_pjsip/pjsip_distributor.c: Request 'INVITE' from '"Ekiga" <sip:msarti@192.168.200.183>' failed for '192.168.200.183:5060' (callid: 8cfb1599-729f-e711-823d-ec8eb5a89425@laptop.sarti.lan) - No matching endpoint found [2017-09-24 08:47:00] NOTICE[4101] res_pjsip/pjsip_distributor.c: Request 'INVITE' from '"Ekiga" <sip:msarti@192.168.200.183>' failed for '192.168.200.183:5060' (callid: 8cfb1599-729f-e711-823d-ec8eb5a89425@laptop.sarti.lan) - Failed to authenticate192.168.200.183 is the actual IP address of softphone, where no firewall is running. Naturally is a LAN IP.
Any ideas?
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Participants: 3
@bob_dt wrote:
I have two locations connected via PfSense (firewall) IpSec VPN tunnel. Location "A" has my FreePBX box behind it and location "B" has two IP telephone extensions behind it.
Everything was working fine until I discovered that my FreePBX box (and firewall) were being attacked by rogue (known blacklisted VOIP) ip addresses. When I determined how to set PfS to accept ONLY connection from my SIP provider then the location "B" extensions had no RTP in or out. I could dial in and out but calls had no voice connection and would time out after 31 seconds.
This is the part that baffles me. PfS is connected via IPSec VPN so no settings on WAN should effect the VPN traffic but, location "B" RTP is being blocked. When I added the location "B" to allow FreePBX to have connection RTP traffic returned.
The PfSense forum is concerned that I might be exposing my FreePBX box or my location "B" extensions to the internet due to the current settings being used.
I am in a difficult position in that this could be a firewall issue, which is not a FreePBX forum issue but without the ability to understand testing and logfile entries of FreePBX I will not be able to figure this out what is wrong, if anything.
My question is what log files do I need to look at to determine the changes in RTP connection? Is there something to "turn on" to help me debug this issue?
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@GPz1100 wrote:
It looks like * codes are passed on when pressed by the remote (calling) party.
Where can I disable this, or configure which * codes can be passed by the remote party?
Note, remote above means party that call into the pbx or I called from the pbx (ie, not local user).
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Participants: 1