@oliv2831 wrote:
Hello,
I would appreciate the ability to change one extension’s user name using FreePBX GraphQL or REST API.
Is it possible to do this ?
Doc [1] does not mention this possibility.[1] https://wiki.freepbx.org/display/FPG/API
Best regards
Posts: 1
Participants: 1
@ashcortech wrote:
Over the weekend I received an alert from my provider saying we’d hit the limit on daily international calling cost. I purposely set this very low so no major disaster.
I do need to track down how this happened though. The sip provider shows several calls over about an hour period. I can only really find one call in the FreePBX CDR’s though and expanding that call log reads like stereo instructions to me.
Can anyone offer some guidance on how to trace down from the FreePBX/Asterisk side who did what?
Posts: 1
Participants: 1
@jch2os wrote:
I got an email from my phone company about fraudulent calls. I don’t expose the system to the internet so they are dialing in and then dialing another number over the phone. I’ve had it before where they guessed a SIP extension password and made calls that way. But that isn’t the case here since I don’t forward any ports from the firewall. I don’t use DISA. Any thoughts by looking at the logs?
Posts: 10
Participants: 6
@KELSIT wrote:
Over the weekend, I turned off the server and the phones to perform a rack swap. When the phones and server were turned back on, outbound calls can be made, but inbound calls go straight to voicemail. The server is a FreePBX 60 system and the phones are Sangoma 505 and 3 of those have extenders.
The graph on the dashboard shows that all the users are offline, but show as online when they make an outbound call:
After seeing that, I checked the
Chan_PJSip Endpointssection under Reports -> Asterisk Info -> chan_pjsip Info and confirmed that all the extensions show as unavailable. So since none of the phones in the ring group are available, the external calls go straight to voicemail. Trying to call from one phone to another also goes to that extension’s voicemail.I’ve tried restarting the phones and the server, nothing changes. There are no new firewall rules since this started and I can ping the phones while logged in via ssh. Checking the asterisk log, the only thing that jumps out at me is this:
[2020-01-13 15:14:57] NOTICE[18819] res_pjsip_exten_state.c: Endpoint ‘117’ state subscription failed: Extension ‘*97’ does not exist in context ‘from-internal’ or has no associated hint
[2020-01-13 15:14:57] NOTICE[18819] res_pjsip_exten_state.c: Endpoint ‘117’ state subscription failed: Extension ‘*88’ does not exist in context ‘from-internal’ or has no associated hintI get a similar message for each extension, but, I’m not sure if it is relevant.
Posts: 4
Participants: 1
@DarkSide wrote:
Hello, guys.
I want to set up two features:
- Restrict internal calls for some pool of phones (for example 1001 cant ring to 1002 and 1002 cant ring to 1001 from 11pm to 7am, but they can call to external numbers all the time)
- Hangup calls that last more than two minutes for some pool of phones (1001 and 1002 can`t talk more than 2 timnues, 1003 can talk as long as he wants it)
The question is: can i do that without buying commercial modules or writing some custom extensions/contexts ?
Thank you
Posts: 2
Participants: 2
@AmidouFlorian92 wrote:
Hi all
I’m trying to setup a freepbx server on VMWARE 15 but I got one problem. I have read the previous posts but they don’t seem to resolve my problem.
I want to configure two networks on my server in VMWARE : one for the updates of my server and the other to be able to access locally to my server.
So i put the vnet0 on NAT and I’m able to make updates but when I want to put a static address on the other interface(vnet0) I lost the internet connection.
My freepbx server must also be able to connect to a gsm gateway on a external network.
I’ve probably missed something but I don’t know what.
Is someone could help me?
Thanks
Posts: 1
Participants: 1
@PitzKey wrote:
In the past (more than a year ago) FreePBX used to have by default ‘Allow Transport Reload’ set to No.
It used to cause audio issues, since the (external?) contact address(es) wasn’t written to transports unless it was reloaded.
The FreePBX devs changed the default to Yes.
Recently I see community members encouraging users to disable it. So what’s the deal here? Did something break recently that causes these issues and the workaround is not to reload transports?
If these issues were always there, why was it changed back then to have the default of Yes? Considering all these issues which can be caused by having it set to Yes. And how come we see only now a wave of these issues being reported?
P.S. this is not a rant. Rather I’d like to understand what’s happening here. Sorry if I came across wrong.
Posts: 3
Participants: 2
@jfinstrom wrote:
The only way to be secure is to incinerate your server and put the remains in a mix of molten unobtainium. Even then it is a maybe at best.
It is rare that I see a original compromise post. In fact security researchers over the last few years have done well with responsible disclosure. That means in general a software release is put out typically a few weeks before the actual exploit. Most security issues in a perfect world can be prevented through simply updating.
I saw a post where code signing did it’s job. People slam it because they feel it pointless. Well like any other measure it IS pointless until it isn’t.
Unfortunately most issues come from something a code signature can’t catch and that is humans. Many issues are simply a matter of human error and bad configuration.
Does your PBX need to be on the public network (internet)? In most cases the answer is no. Removing direct internet access to your PBX will go a long way. With VPN’s and other technologies the answer to this is almost ALWAYS no.
Does your PBX need to be on the general intranet? Many threats are actually internal. Some employee decides to play with your PBX it doesn’t matter how far it is disconnected from the normal internet. Many internal threats are worse than external threats.
Passwords. Are your passwords appropriately secure. Most sip passwords never need to be typed by a human so they should be obnoxous. This includes trunks, endpoints etc. If your SIP passwords don’t feel like nuclear launch codes FIX THEM!
This is not something you can fix in FreePBX but for providers your sip username should be alphanumeric. In FreePBX your sip username will always be numeric so it is even more important to use a strong password. An attacker scanning your extensions will probably know your extension range is between 100 and 9999 for FreePBX.
Voicemail: Don’t use the extension, 1234, 1111, 0000 as a voicemail pin.
Set up intrusion detection and have it send meaningful notifications.
Lets tl;dr this. Keep updated, use good passwords, keep your pbx isolated from your LAN and the internet whenever possible (spoiler alert, it is always possible).
If you don’t know how to do any of this please learn or hire someone.
Posts: 1
Participants: 1
@GrandPoobah1 wrote:
Newbie to the Asterisk Server Community - Unable to register multiple subscriber lines ( FXS ) via a GW using the same source IP address. Once the server appears to bind the GW IP and successfully registers the 1st line, it rejects the registration attempts of the additional lines with 403s.
Can someone shed a lil’ light on how to resolve this issue?
Thanks in advance
Posts: 2
Participants: 2
@bigmillz wrote:
I found a 5 year old post about this but really no resolution.
Is it possible to have multiple CID prefixes retained?
My reasoning: we just added a 2nd main number to track advertising conversions. I’d like a prefix for calls coming in through that number to show up. We’re small enough that I’m leaving it to my team to report good/bad conversions to me.
Posts: 3
Participants: 2
@CantComplain wrote:
Hello Everyone,
Installed FreePBX 15. All modules are up to date. After updating, the Queue, Ring Group, Set Caller ID and Queue Priority disappeared under the GUI.
Appreciate any help :).
Posts: 4
Participants: 2
@tezza4x4 wrote:
I’m having a problem with my ISP SIP Trunk (IInet). For a while now, it hasn’t been registering with my ISP and as such incoming calls were not working at all. Outbound calls were fine. So i did a yum update, and this “improved” the situation and the trunk is now registering. Outbound calls working fine as before, but now Inbound calls are ringing on my Cisco IP Phone extension, but as soon as I try to answer they disconnect. Log of the call ringing and disconnecting below. Error seems to be related to trying to transcode G.729. Trouble is I’m not trying to use G.729 on either the trunk (have configured it for only G.711 ulaw and alaw) nor the IP Phone the call is terminating on. I have disallowed all codecs on the extension config, and only allowed ulaw and alaw. The inbound invite from the ISP is also below. The trace on the extension side shows G.711 ulaw being successfully negotiated.
Any ideas what could be wrong.
Asterisk Log:
[2020-01-14 12:38:30] VERBOSE[16963][C-0000001b] app_dial.c: Connected line update to PJSIP/iinet-00000038 prevented.
[2020-01-14 12:38:30] VERBOSE[16963][C-0000001b] app_dial.c: PJSIP/101-00000039 is ringing
[2020-01-14 12:38:30] VERBOSE[16963][C-0000001b] app_dial.c: PJSIP/101-00000039 is ringing
[2020-01-14 12:38:33] VERBOSE[16963][C-0000001b] app_dial.c: PJSIP/101-00000039 answered PJSIP/iinet-00000038
[2020-01-14 12:38:33] VERBOSE[16964][C-0000001b] bridge_channel.c: Channel PJSIP/101-00000039 joined ‘simple_bridge’ basic-bridge
[2020-01-14 12:38:33] VERBOSE[16963][C-0000001b] bridge_channel.c: Channel PJSIP/iinet-00000038 joined ‘simple_bridge’ basic-bridge
[2020-01-14 12:38:33] WARNING[16963][C-0000001b] translate.c: No translator path: (ending codec is not valid)
[2020-01-14 12:38:33] WARNING[16963][C-0000001b] translate.c: No translator path: (ending codec is not valid)
[2020-01-14 12:38:33] WARNING[16964][C-0000001b] channel.c: Unable to find a codec translation path: (g729) -> (alaw)
[2020-01-14 12:38:33] VERBOSE[16964][C-0000001b] bridge_channel.c: Channel PJSIP/101-00000039 left ‘simple_bridge’ basic-bridge
[2020-01-14 12:38:33] VERBOSE[16963][C-0000001b] bridge_channel.c: Channel PJSIP/iinet-00000038 left ‘simple_bridge’ basic-bridge
[2020-01-14 12:38:33] WARNING[16963][C-0000001b] channel.c: Unable to find a codec translation path: (g729) -> (alaw)
[2020-01-14 12:38:33] VERBOSE[16963][C-0000001b] app_macro.c: Spawn extension (macro-dial-one, s, 54) exited non-zero on ‘PJSIP/iinet-00000038’ in macro ‘dial-one’
[2020-01-14 12:38:33] VERBOSE[16963][C-0000001b] app_macro.c: Spawn extension (macro-exten-vm, s, 26) exited non-zero on ‘PJSIP/iinet-00000038’ in macro ‘exten-vm’
[2020-01-14 12:38:33] VERBOSE[16963][C-0000001b] pbx.c: Spawn extension (ext-local, 101, 2) exited non-zero on ‘PJSIP/iinet-00000038’
[2020-01-14 12:38:33] VERBOSE[16963][C-0000001b] pbx.c: Executing [h@ext-local:1] Macro(“PJSIP/iinet-00000038”, “hangupcall,”) in new stack
[2020-01-14 12:38:33] VERBOSE[16963][C-0000001b] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/iinet-00000038”, “1?theend”) in new stack
[2020-01-14 12:38:33] VERBOSE[16963][C-0000001b] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2020-01-14 12:38:33] VERBOSE[16963][C-0000001b] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/iinet-00000038”, “0?Set(CDR(recordingfile)=)”) in new stack
[2020-01-14 12:38:33] VERBOSE[16963][C-0000001b] pbx.c: Executing [s@macro-hangupcall:4] Hangup(“PJSIP/iinet-00000038”, “”) in new stack
[2020-01-14 12:38:33] VERBOSE[16963][C-0000001b] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘PJSIP/iinet-00000038’ in macro ‘hangupcall’Inbound Invite from ISP:
No. Time Source Destination Protocol Length Info
480 31.903656 203.55.231.200 192.168.100.100 SIP/SDP 878 Request: INVITE sip:0731224895@192.168.100.100:5060 |Frame 480: 878 bytes on wire (7024 bits), 878 bytes captured (7024 bits) on interface \Device\NPF_{E52C8EC1-5642-4A42-BEA3-3A10B7D186F4}, id 0
Ethernet II, Src: Cisco_34:ba:05 (00:24:c4:34:ba:05), Dst: HewlettP_7f:bc:ee (30:e1:71:7f:bc:ee)
Internet Protocol Version 4, Src: 203.55.231.200, Dst: 192.168.100.100
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:0731224895@192.168.100.100:5060 SIP/2.0
Method: INVITE
Request-URI: sip:0731224895@192.168.100.100:5060
Request-URI User Part: 0731224895
Request-URI Host Part: 192.168.100.100
Request-URI Host Port: 5060
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 203.55.231.200:5060;branch=z9hG4bKd0ej18102gr7csis0jo0.1
From: sip:0412644815@10.11.1.1;user=phone;tag=876634584-1578960212713-
SIP from address: sip:0412644815@10.11.1.1;user=phone
SIP from tag: 876634584-1578960212713-
To: "Not Known"sip:0731224895@iinetphone.iinet.net.au
SIP Display info: “Not Known”
SIP to address: sip:0731224895@iinetphone.iinet.net.au
Call-ID: BW0003327131401202145142210@10.11.1.1
[Generated Call-ID: BW0003327131401202145142210@10.11.1.1]
CSeq: 279865717 INVITE
Sequence Number: 279865717
Method: INVITE
Contact: sip:0412644815@203.55.231.200:5060;transport=udp
Contact URI: sip:0412644815@203.55.231.200:5060;transport=udp
Supported: 100rel
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY
Recv-Info: x-broadworks-client-session-info
Accept: application/media_control+xml,application/sdp,multipart/mixed
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 176
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): BroadWorks 111915857 1 IN IP4 203.55.231.203
Session Name (s): -
Connection Information ©: IN IP4 203.55.231.203
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 43050 RTP/AVP 8 0 18 101
Media Type: audio
Media Port: 43050
Media Protocol: RTP/AVP
Media Format: ITU-T G.711 PCMA
Media Format: ITU-T G.711 PCMU
Media Format: ITU-T G.729
Media Format: DynamicRTP-Type-101
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Format: 101
MIME Type: telephone-event
Sample Rate: 8000
Media Attribute (a): fmtp:101 0-15
Media Attribute Fieldname: fmtp
Media Format: 101 [telephone-event]
Media format specific parameters: 0-15
[Generated Call-ID: BW0003327131401202145142210@10.11.1.1]200OK from FreePBX:
No. Time Source Destination Protocol Length Info
498 34.143994 192.168.100.100 203.55.231.200 SIP/SDP 957 Status: 200 OK |Frame 498: 957 bytes on wire (7656 bits), 957 bytes captured (7656 bits) on interface \Device\NPF_{E52C8EC1-5642-4A42-BEA3-3A10B7D186F4}, id 0
Ethernet II, Src: HewlettP_7f:bc:ee (30:e1:71:7f:bc:ee), Dst: Cisco_34:ba:05 (00:24:c4:34:ba:05)
Internet Protocol Version 4, Src: 192.168.100.100, Dst: 203.55.231.200
User Datagram Protocol, Src Port: 5060, Dst Port: 5060
Session Initiation Protocol (200)
Status-Line: SIP/2.0 200 OK
Status-Code: 200
[Resent Packet: False]
[Request Frame: 480]
[Response Time (ms): 2241]
Message Header
Via: SIP/2.0/UDP 203.55.231.200:5060;rport=5060;received=203.55.231.200;branch=z9hG4bKd0ej18102gr7csis0jo0.1
Call-ID: BW0003327131401202145142210@10.11.1.1
[Generated Call-ID: BW0003327131401202145142210@10.11.1.1]
From: sip:0412644815@10.11.1.1;user=phone;tag=876634584-1578960212713-
SIP from address: sip:0412644815@10.11.1.1;user=phone
SIP from tag: 876634584-1578960212713-
To: “Not Known” sip:0731224895@iinetphone.iinet.net.au;tag=9fb48522-6188-4229-86d7-24f2fc64f4ac
SIP Display info: “Not Known”
SIP to address: sip:0731224895@iinetphone.iinet.net.au
SIP to tag: 9fb48522-6188-4229-86d7-24f2fc64f4ac
CSeq: 279865717 INVITE
Sequence Number: 279865717
Method: INVITE
Server: FPBX-13.0.197.21(13.29.2)
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Contact: sip:192.168.100.100:5060
Contact URI: sip:192.168.100.100:5060
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 257
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 111915857 3 IN IP4 192.168.100.100
Session Name (s): Asterisk
Connection Information ©: IN IP4 192.168.100.100
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 15914 RTP/AVP 8 0 101
Media Type: audio
Media Port: 15914
Media Protocol: RTP/AVP
Media Format: ITU-T G.711 PCMA
Media Format: ITU-T G.711 PCMU
Media Format: DynamicRTP-Type-101
Media Attribute (a): rtpmap:8 PCMA/8000
Media Attribute Fieldname: rtpmap
Media Format: 8
MIME Type: PCMA
Sample Rate: 8000
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute Fieldname: rtpmap
Media Format: 0
MIME Type: PCMU
Sample Rate: 8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Format: 101
MIME Type: telephone-event
Sample Rate: 8000
Media Attribute (a): fmtp:101 0-16
Media Attribute Fieldname: fmtp
Media Format: 101 [telephone-event]
Media format specific parameters: 0-16
Media Attribute (a): ptime:20
Media Attribute Fieldname: ptime
Media Attribute Value: 20
Media Attribute (a): maxptime:150
Media Attribute Fieldname: maxptime
Media Attribute Value: 150
Media Attribute (a): sendrecv
[Generated Call-ID: BW0003327131401202145142210@10.11.1.1]
Posts: 1
Participants: 1
@Nanotech wrote:
Hi everyone, I’m new to FreePBX but i’ve watched tutorials and i’ve been able to set everything I want except for one thing. Time Conditions. I’ve create a time group with our Business hours (9am - 6pm Monday-Friday) and a Time Condition that if it matches, it goes to a ring group, If not, it goes to the IVR.
My inbound route is setup to the time conditions. The problem i’m having is that all the calls go to the IVR and never to the ring group when this is activated Any help is appreciated. Thanks in advance!
Posts: 1
Participants: 1
@davidschmidt88 wrote:
Hi there, posting here on the forums instead of reddit!
Maybe somebody can help me with a log?
Commercial plugin vm notify. But sometimes it does not even reach the voicemail.
Worst issue if something does not happen 100% of the time!
I just discovered sngrep which looks like an awesome tool. Will try later on with that as well!
But if anybody sees something obvious here it would speed up things possibly!
Best
David
Posts: 1
Participants: 1
@cagriaksu wrote:
Hello everyone,
Yesterday, after updating our server from 13 to 14, things got messy. I’ve solved most of the problems but couldn’t figure out why I can’t click and extend the modules in the module admin page. I can check for updates, and apply them, but I can’t individually select the modules (they extend downwards when you click on them as you all know)
Does anybody know a reason and solution for this?
Posts: 4
Participants: 2
@Strato83 wrote:
Hi, all.
Using Freepbx 14. In the description of the “Call Forward Ring Time” option in the Advanced tab of SIP extension i see - “If voicemail is disabled and their is not destination specified, it will be forced into Always mode”.
But it seems for me, that this description is not correct.
I have extension 2999, that have CFU set to extension 2998.
Voicemail on the 2999 is disabled, optional destinations is set to “… Voicemail if Enabled”, Ring Time - Default (20 sec in Advanced Options), Call Forward Ring Time - Default (0 in Advanced Options).As i see it, witn this settings the “Call Forward Ring Time” for extension 2999 must be forced to Always, but in fact it is set to 20 sec (same as default Ring Time).
As result, when i call to extension 2999 in rings for 20 seconds, then 2998 rings for 20 seconds, then i get a BUSY signal.
If i manually set “Call Forward Ring Time” for extension 2999 to Always i get the right result - ext. 2999 rings for 20 sec. then ext. 2998 rings until i hangup.
What i’m missing? Why “Call Forward Ring Time” is not forcing in Always mode?
Posts: 1
Participants: 1
@oliv2831 wrote:
Hello,
After reading [1] several times, I’m realizing I don’t exactly understand what this “Add API Application” menu is all about.
It certainly gives a valuable insight on how grants are handed but not much about architectures.Does it allow a FreePBX admin to implement:
a custom web application hosted on some local or remote host (ie not hosted on the same box FreePBX is installed on) and share credentials with FreePBX but not linked with FreePBX (a click within FreePBX web app does not link the this custom web app).
a custom web application hosted on the same box FreePBX is installed on and share credentials with FreePBX
a custom web application that queries FreePBX data
a custom web application queried by FreePBX when some events occur (extension creation, …).
Thoughts ?
[1] https://wiki.freepbx.org/display/FPG/API+Applications
Best regards
Posts: 1
Participants: 1
@dotcom wrote:
Hi All,
We are noticing a lot of these warning messages:
[2020-01-14 12:21:26] WARNING[15203][C-00000187]: chan_sip.c:10395 process_sdp: Declining non-primary audio stream: audio 13224 RTP/AVP 8 0 101 13
[2020-01-14 12:21:33] WARNING[15203][C-00000188]: chan_sip.c:10395 process_sdp: Declining non-primary audio stream: audio 22156 RTP/AVP 8 0 101 13
[2020-01-14 12:21:35] WARNING[15203][C-00000189]: chan_sip.c:10395 process_sdp: Declining non-primary audio stream: audio 12450 RTP/AVP 8 0 101 13We have ALAW as preferred codec on our FreePBX and also on the Sangoma SBC.
Does anyone has an idea with could cause this?
Thanks for your help!
Posts: 2
Participants: 2
@jcadman wrote:
I have FreePBX installed on a raspberry pi which I use for my home. I use 3 x Grandstream 2 Line 1 SIP phones as well as 2 x Mitel 5300 phones with SIP Version 06.03.03.06 which was implemented by TFTP. After setting up the Mitel 5330 phones I realised that there was an issue with the sound recordings on the Mitel 5330 where the sound recordings for voices distort/crackle. This is not the case on some of the sounds but majority of the sounds start to crackle. With my Grandstream phones, the sound recordings do not crackle. When calling in and out using the phones including extensions within the house, the audio is clear. The crackle only appears to be with the system recordings voice.
Posts: 3
Participants: 2
@jsp1864 wrote:
I need some help figuring how language is working with Park and Announce Module. The slot announce is always in english no matter what configuration I use. I tried to force channel language in french and it was still in english.
Is there a menu or a way to force the park and announce in a specific language ?
Thank you for your help
Posts: 1
Participants: 1
