@lgraham wrote:
FreePBX 13.0.195.1 + EndPoint Manager
PolyCom VVX250 (4 lines)(The VVX250 isn’t an option in EPM so I use the VVX300 template)
We have a helpdesk group whose extensions are also included in a Ring Group. Is it possible to have one of the line keys reflect when the call is coming in via the Ring Group as opposed to the extension directly?
Posts: 3
Participants: 3
@ghurty wrote:
I have a freepbx server hosted on vultr with two yealink phones. Sometimes the phones cant answer the calls and you have to press answer more then once. I have STUN set up. I used to use openvpn on the yealink phones, but the new phones dont support the script I had. Any suggestions?
Thank you
Posts: 6
Participants: 3
@jyanta wrote:
Hi Guys,
This is a very premature message and I plan to have logs tomorrow when I can reduplicate the issue.
When I try to call my DID from a Cell, I sometimes get “The number is not in service, etc”.If I call immediately back, it flows through to the IVR properly. (Or if I make an Outbound call on my SIP).
Seems like it fails once, then it reestablishes a connection and it’s good to go.
I know this can be a variety of issues however Im hoping someone has a few suggestions to inspect.
I am using PFSense, Ports are Opened as required, DHCP to FreePBX with a DHCP Reservation in place. Incoming/Outgoing have no issues once something gets re-established (Seems that from earlier testing everything is fine when the Asterisk log stays “Setting global variable ‘SIPDOMAIN’ to ‘10.10.10.14’” which is my FreePBX IP.
Hopefully someone has some suggestions until I can post some logs.
FreePBX 14.0.5.2 - Running Fully up to date.
Posts: 1
Participants: 1
@striker495 wrote:
I am working on setting up a few GXP2170 to use open VPN using the VPN client built into FreePBX. When looking at the setting on the 2170 for VPN Network>openVPN, I see you have the choice is Ciphers Methods. But nowhere in the GUI of the set up of the VPN do I see anywhere to set this. what am I missing?
Posts: 1
Participants: 1
@SAHEISLER wrote:
Has anyone ever used “Magic Telecom” as a DID source? My PBX shows their trunks are connected but the test DID does not ring in. Any ideas?
Posts: 1
Participants: 1
@sopheak wrote:
Dear everyone,
I have sip trunk with provider is up both site, the out going call it working find but incoming call it got sip/2.0 603 Decline. anything help?
Posts: 2
Participants: 1
@matthewljensen wrote:
I started messing around with the database commands from the CLI, and I wanted to try changing the followme number in an extension from there. It seems to work, meaning that calls get redirected to the new number, simply by running something like:
database put AMPUSER 201/followme/grplist 123456789#But I read in this post: CLI commands, that this isn’t the best way to do this and that it may cause problems. And that it’s better to change it in the MySQL database, and then run a fwconsole reload.
Is this still the case, and if so, what are the repercussions of changing them directly in the asterisk database rather than in the MySQL database?
Posts: 4
Participants: 3
@Axel_23 wrote:
Hello
I try to install chan-sep_master for using 5 Cisco IP-Phones 7970 an 7975. First was installing a fresh version of freepbx with the image from the webside. Let run all the updates and deinstall the commercial modul (homeoffice using). So now i get a a freepbx 14.0.5.2 with Asterisk 15.5.
I looks like a CentOS 7.6. base. Everything its installed in a VM on a XCP-NG server.I am following this instruktion for chan-sccp and chan-sccp_master. but configuration to make is not flying …
github…chan-sccp/chan-sccp/wiki/Setup-FreePBX
load the source with git
cd /usr/src
git clone …/chan-sccp/chan-sccp chan-sccp_master
cd chan-sccp_master
git checkout master
./configureresult was
checking pkg-config asterisk... not found checking Search Path: /usr /usr/local /opt... not-found not-found not-found configure: Please install either the asterisk-devel package. configure: Or run ./configure --with-asterisk=PATH with PATH pointing to the directory where you installed asterisk configure: error: Cannot find pbx libraries - these are required.I am trying ./ ./configure --with-asterisk=PATH
./configure --with-asterisk=/usr/lib64/asterisk/
./configure --with-asterisk=/usr/lib64/asterisk/modules
with the no resultswhere is my mistake oder where is the rigth path ?
thx Axel
PS: Sorry it is my first use of you community
new user cant send links… there must be someone had the same problem
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Participants: 2
@lief4321 wrote:
Hi, I’m looking for some advice.
I have two systems, system1 and system2
system1 contains all user extensions and is the main install for the organization. (PBXAct (FreePBX))
system2 is essentially an old Digium Switchvox 65 that has a 4 port DHADI FXO board installed. (FreePBX)
The POTS service provider refuses to transfer any ownership of the DIDs and the customer would not like to buy an FXO box (I was suggesting a Vega but they are pretty expensive).
I would like to somehow route all external calls from system1 through system 2 straight through the FXO ports. System2 doesn’t contain any extensions so system2 would be purely used for incoming and outgoing external calls (more or so a medium).
What is the best way to accomplish this?
I have tried setting up a regular extension on system2 and then adding that SIP ext onto system1 as a trunk but the problem I face is with the CID. I naturally thought that I would just use the external number as the CID in the trunk settings but when I do that system2 shows that the call is anonymous (see picture 1). The call does not complete. When i change the caller ID in the trunk settings to the extension number used for the trunk extension of system2 (1000), system1 doesn’t route the call to system2 and I get a busy circuit message from system1.
The second thing I tried was doing an IAX2 trunk but I got stumped when I created the trunk and the status was stuck at “Unknown and Unmonitored” in the “asterisk info”.
In thinking that there has got to be an easy way to do this but I just can’t seem to figure it out. Googling around doesn’t seem to help either because of the uniqueness of the scenario.
Any help is much appreciated.
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Participants: 3
@busster8 wrote:
Have a system that has been running for quite a while. Running 2.11.0.43
When monitoring the system, is see all of the sip devices go unavailable, then unreachable, then unavailable, then idle, and finally reachable.
It appears that this happens approximately every 15 minutes, and the entire process resolves itself quickly.
No one complains of service interruption, but this is not a busy system. No one complains of internet loss. Phones and computers use one of two POE network switches.
I would suspect a network switch issue, but then both switches would have to be an issue.
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Participants: 2
@sentinelace wrote:
I updated my tftp option 66 to a new pbx IP using the http://HTTP Username:HTTP Password@FQDN:83
For some reason even factory resetting the phone, it still gets the old TFTP. I only have one DHCP server. Is there a cache somewhere? The DHCP server is 2012
Posts: 1
Participants: 1
@sentinelace wrote:
I have one number that will not accept any of the DTMF options, I tried info, auto, rfc2833 and none work for incoming. I called the sip provider and they said they cannot reproduce the issue. No matter what carrier you come from, it doesn’t work. I know I can enable DTMF for the logs, but I’m not sure what to look for as an example.
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Participants: 1
@gelcom wrote:
As I can’t find any post more recent then 2014 I think this is worth a try…
Is it possible to use any known FXO analog telephone card with FreePBX as a VM on ESXI in passthrough mode or any other way?
I have a powerfull XEON server laying arround and I’d like to virtualize my FreePBX box and as I already have the hardware I have no interest in buying an additional ATA to do the FXO to Asterisk job.
kind regards
Posts: 1
Participants: 1
@rob521 wrote:
I have 2 external handsets. Suddenly the port numbers are being remapped from 5060 to 57529 and 57530.
Both handsets are not unreachable since this started happening.
Yesterday they were both registering on 5060 and working perfectly.Could any suggest how or why this is happening, and where I can make changes to these port numbers so they can register back on 5060.
Much appreciated.
Posts: 1
Participants: 1
@mvogel4949 wrote:
Single hour or multiple hour events seem to be working but if I put in a 2d event in my google calendar - say Dec 22-23 when the FreePBX calendar syncs it displays Dec 21-23. I was thinking it was a cosmetic issue and it was correct on the back end but when I implement the calendar into a TC it sees today the 21st as a match even though in the google calendar it is not there.
Posts: 1
Participants: 1
@kb9mfd wrote:
This is a new install and I have everything setup just like I always would and I tried to get a phone working before I have to install it on Wednesday.
The phones will grab their config, and then they will not register. I have turned on debug in the CLI and here is what I am getting -
[2018-12-21 20:27:15] DEBUG[2543]: res_pjsip/pjsip_distributor.c:383 find_dialog: Could not find matching transaction for Request msg REGISTER/cseq=1 (rdata0x7f44780084b8)
[2018-12-21 20:27:15] DEBUG[2543]: res_pjsip/pjsip_distributor.c:461 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-00000063 to use for Request msg REGISTER/cseq=1 (rdata0x7f44780084b8)
[2018-12-21 20:27:15] DEBUG[2540]: threadpool.c:517 grow: Increasing threadpool SIP’s size by 5
[2018-12-21 20:27:15] DEBUG[24932]: res_pjsip_endpoint_identifier_ip.c:223 common_identify: No identify sections to match against
[2018-12-21 20:27:15] DEBUG[24932]: res_pjsip_endpoint_identifier_user.c:148 username_identify: Attempting identify by From username ‘2000’ domain ‘198.27.60.210’
[2018-12-21 20:27:15] DEBUG[24932]: res_pjsip_endpoint_identifier_user.c:160 username_identify: Identified by From username ‘2000’ domain ‘198.27.60.210’
[2018-12-21 20:27:15] WARNING[24932]: res_pjsip_registrar.c:989 registrar_on_rx_request: Endpoint ‘2000’ has no configured AORs
[2018-12-21 20:27:15] DEBUG[2751]: manager.c:5990 match_filter: Examining AMI event:
Event: FailedACL
Privilege: security,all
EventTV: 2018-12-21T20:27:15.049+0000
Severity: Error
Service: PJSIP
EventVersion: 1
AccountID: 2000
SessionID: 5dc95212990bd96@192.168.0.21
LocalAddress: IPV4/UDP/198.27.60.210/5060
RemoteAddress: IPV4/UDP/24.159.225.222/1024
ACLName: registrar_attempt_without_configured_aorsI have tried to rebuild the station, rebuilt EPM, I tried even changing it to chan_sip, no luck. I have gone over Asterisk SIP settings and that is correct also. I am at a loss. I need this asap as its going in the day after Christmas right away in the morning.
I also opened a ticket with Sangoma, but maybe someone here has a idea what that error means - " Endpoint ‘2000’ has no configured AORs" - Google only returns issues with “anonymous” stations, this is a configured station, and yes I have tried multiple stations and none of them will work.
Funny thing is Phone Apps works, so I do not believe this is a connectivity issue as much as a config issue but I cannot find what. Thanks!
Posts: 1
Participants: 1
@Amitiyer wrote:
Hello,
I have a DID number configured in my Freepbx, Now i have like 30 extensions. Now what i want to do is when a user calls the DID number, he can dial his extension number and then the call would go directly to his voicemail so he can listen to his voicemail box.
Can any one please let me know how can i achieve it ? also please let me know what and where should i change.
Thank you.
Posts: 2
Participants: 2
@asifahmed009 wrote:
Getting these errors continuously in CLI. Everything is working fine but why and from where these logs are generate kindly help me.
– Executing [s@ivr-22:12] WaitExten(“Local/6385@from-queue-00010a9a;2”, “4,”) in new stack
– Timeout on Local/6385@from-queue-00010a9a;2, going to ‘t’
[2018-12-22 12:17:24] WARNING[31518][C-0000c18e]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected ‘+’, expecting $end; Input:
+1
^
[2018-12-22 12:17:24] WARNING[31518][C-0000c18e]: ast_expr2.fl:474 ast_yyerror: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
– Executing [t@ivr-22:1] Set(“Local/6385@from-queue-00010a9a;2”, “TIMEOUT_LOOPCOUNT=”) in new stack
[2018-12-22 12:17:24] WARNING[31518][C-0000c18e]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected ‘>’, expecting $end; Input:1
^
[2018-12-22 12:17:24] WARNING[31518][C-0000c18e]: ast_expr2.fl:474 ast_yyerror: If you have questions, please refer to https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
– Executing [t@ivr-22:2] GotoIf(“Local/6385@from-queue-00010a9a;2”, “?final”) in new stack
– Executing [t@ivr-22:3] Set(“Local/6385@from-queue-00010a9a;2”, “IVR_MSG=custom/timeout”) in new stack
– Executing [t@ivr-22:4] Goto(“Local/6385@from-queue-00010a9a;2”, “s,start”) in new stack
– Goto (ivr-22,s,10)
– Executing [s@ivr-22:10] Set(“Local/6385@from-queue-00010a9a;2”, “TIMEOUT(digit)=3”) in new stack
– Digit timeout set to 3.000
– Executing [s@ivr-22:11] ExecIf(“Local/6385@from-queue-00010a9a;2”, “1?Background(custom/timeout)”) in new stack
– <Local/6385@from-queue-00010a9a;2> Playing ‘custom/timeout.slin’ (language ‘en’)Using
FreePBX 2.11.0
Asterisk 11.25.0 0.el7.centos
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Participants: 1
@rtarson wrote:
I have my sip trunking working perfectly fine for outbound calls. I communication is solid for dialing out. When I dial in though the call ends abruptly/as soon as the call is dialed.
Currently I also have a very simple inbound call routing since I just begun my journey lol. I have all defaults except for DID set to: +1[My Number with Area Code]
Set Destination is set to: Extension & my Extension.I have my main sip_chan which is the one with the CID configured this is the inbound setting:
Main Twilio trunk:
host=54.172.60.0
type=peer
context=from-trunkThen I have other sip_chan conf have inbound settings for the multiple Twilio host IPs they have. I have not entered anything other then name of sip and the inbound configuration:
Twilio Trunk1:
host=54.172.60.1
type=peer
context=from-trunkTwilio Trunk2:
host=54.172.60.2
type=peer
context=from-trunkTwilio Trunk3:
host=54.172.60.3
type=peer_
context=from-trunkI get calls find once I enable both:
- Allow Anonymous Inbound SIP Calls
- Allow SIP Guests
Posts: 2
Participants: 1
@Bradbpw wrote:
I have a backup setup to backup all my call recordings each night. The issue with this is that I either need to change the setting each day or I need to backup a lot of recordings multiple times. My current setting backups up the directory
__ASTSPOOLDIR__/monitor/2018/12. At this point, I am changing the month directory each month. So, December 1st I get recordings from December 1st. But by December 5th I’m getting recordings from December 1st, 2nd, 3rd, 4th, and 5th in one backup.Is there a way to make this incremental? Either by using a wildcard in the directory (
__ASTSPOOLDIR__/monitor/year/month/day) or by adding a command that tells the system to only backup new files that have not been previously backed up?
Posts: 2
Participants: 2
new user cant send links… there must be someone had the same problem 