@claloano wrote:
Going to the sound language menu and trying to install Italian gives me this error:
SQLSTATE[22001]: String data, right truncated: 1406 Data too long for column ‘filename’ at row 1
File:/var/www/html/admin/modules/soundlang/Soundlang.class.php:1125I am a little confused someone can tell me how to solve
FreePBX is new … recently installed
FreePBX version 15.0.16.38 with all updated modules
Posts: 1
Participants: 1
@ghurty wrote:
New asterisk 13.19.1 install
The PBX is in vultr, the phones are registered via STUN.
When the user an inbound call on hold, they get an error that the “call transaction doesnt exist”.
I factory restored the phone, but it didnt help. The phone is a yealink.
The logs dont show anything.Thank you
Posts: 2
Participants: 2
@rymes wrote:
So, we have struggled with this issue for years, to the point that details about it have passed from fact into lore, but we have a queue that will occasionally send a second call to an agent who is already on the phone with another queue call, or perhaps an outbound call.
The extension has call waiting disabled. The Cisco 7960 phone being used has its call_waiting setting disabled. The queue is configured like so (plus agent restrictions is set to “Extensions Only”):
[500] announce-frequency=0 announce-holdtime=no announce-position=no autofill=yes autopause=no autopausebusy=no autopausedelay=0 autopauseunavail=no joinempty=yes leavewhenempty=no maxlen=0 memberdelay=0 min-announce-frequency=15 musicclass=default penaltymemberslimit=0 periodic-announce-frequency=0 queue-callswaiting=silence/1 queue-thereare=silence/1 queue-youarenext=silence/1 reportholdtime=no retry=5 ringinuse=no servicelevel=60 strategy=wrandom timeout=25 timeoutpriority=app timeoutrestart=no weight=0 wrapuptime=30 context= lazymembers=noWhat can I start digging for in the logs to figure out why this happens from time to time? When we have an issue, we often see a single call that is repeatedly ringing agents that are not available or on a call, even while other calls that have been waiting for less time are presented to available agents. I do understand that the autofill setting will mean that calls won’t always be answered first-come, first-served, but when we have an issue, it’s particularly bad (one call holding 8+ minutes, another answered after 10 seconds).
Posts: 1
Participants: 1
@ehstein wrote:
I’m working on a phone system for a small client (apx 20 phones or so) and I’m having a heck of a time getting one particular feature to work.
The client has asked that the receptionist be able to toggle day/night mode - easy. No problem. I setup a time condition based on a time group. All good.
We’re using Grandstream GXP-2170 phones. I configured a softkey as a BLF for *270.
When the button is pressed, a voice announces that the time is activated or deactivated, but the BLF light doesn’t change.
Interestingly, if I go into the time condition and Invert the BLF hint, the light will change to red. But it doesn’t seem to toggle when the state of the time condition changes.
Any thoughts?
Posts: 1
Participants: 1
@User_42 wrote:
Hello I am new to VoIP and SIP so I thought I would try to learn by doing. Unfortunately I’m ripping my hair out with trying to get this to work. I’ve looked at the forums from my sip provider(Twilio), my router(Unifi USG), and FreePBX I have tried absolutely everything I could find.
My problem so far is on outbound calls the soft phone(X-Lite) can hear the PSTN but the PSTN can’t here the soft phone. And on top of that the calls end after 30 seconds, my sip provider(Twilio) says that the “caller”(PBX or soft phone) is ending the call.
Any help is greatly appreciated and if you need more information please ask!
Posts: 1
Participants: 1
@User_42 wrote:
I am not receiving inbound calls from my sip provider(Twilio). Twilio says it’s my sip infrastructure has a problem. I have forwarded the correct ports on my router. I have looked at articles from Twilio and these forums but every thing I try doesn’t work. Any help is greatly appreciated!
If you need more info please ask!
Posts: 1
Participants: 1
@uncleawesome wrote:
Hello.
First of all, I am a complete beginner, and I have no knowledge about this program or how modems and phones work. I collect old computers and I want them online using old modems of different kinds. Right now I am trying with a regular 56k modem,v90 or something like that using winxp.
The server:
Old pentium 4.
DAHDI Version: 2.11.1 Echo Canceller: OSLEC
Asterix version: 13.17.0
FreePBX 13.0.192.16
I have a tdm400p with 4 fxs modules. There is no molex power input on the card. All 4 led’s are green on the card.When I try to dial something on my modem, it just says, “phone line is busy”.
I did not get any dial tone before, then I did stuff, and got “phone is busy”.
But I need to set up PSTN I guess? I tried to google this, but I can’t figure out completely how to set this up with my setup, how to get internet in through lan, and shared out to my modems.Any help, suggestions are really appreciated :
Posts: 1
Participants: 1
@wgbecks wrote:
Hello.
I have a question regarding how I might configure my SEPconf.xml to enable handset sidetone for a Cisco 7945G working on FreePBX 15.0.16.38? I tried adding sideToneLevel to the vendorConfig section of my SEP.cnf.xm,l but no joy, it made no difference.
According to the Cisco documentation, handset/headset sidetone was support started at Firmware Version 9-3.x and my 7945G is at SIP45.9-4-2SR3-1S.
Any help would be appreciated!
Posts: 1
Participants: 1
@gguldens56 wrote:
Using the Freepbx GUI 15.0.16.38 I modified the callerid source for my trunk that was working under pjsip, and when I reloaded the config, the trunk no longer worked. Making an outbound call, I get the following in my /var/asterisk/full.
Checked the obvious to see that I didn’t change anything related to the name of the trunk, and everything is in agreement. Looking in the pjsip.endpoint.conf file the trunk entry looks good. The “outbound_auth=” has the trunk name in it, but there in no section in the file for that stores the actual authentication. Oddly, running the CLI and issuing the "pjsip show registrations does show the trunk as registered.
So using the Freepbx GUI to manage the trunk, why now did the pjsip trunk suddenly stop being found as an endpoint which in turn seems to prevent pjsip from creating a channel for the trunk?
Been troubleshooting for a few days now, this is my home system, so I can keep it down for s bit, but I would like to solve this issue so I can avoid it in the future.
Thanks,
Greg Guldenschuh[2020-01-04 12:51:20] VERBOSE[26486][C-00000003] pbx.c: Executing [s@macro-dialout-trunk:26] Dial(“PJSIP/2201-00000002”, “PJSIP/6783441513@Teli-Inbound,300,Tb(func-apply-sipheaders^s^1,(4))”) in new stack
[2020-01-04 12:51:20] ERROR[24013] chan_pjsip.c: Unable to create PJSIP channel - endpoint ‘Teli-Inbound’ was not found
[2020-01-04 12:51:20] WARNING[26486][C-00000003] app_dial.c: Unable to create channel of type ‘PJSIP’ (cause 3 - No route to destination)
[2020-01-04 12:51:20] VERBOSE[26486][C-00000003] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
Posts: 4
Participants: 3
@tonyroberts wrote:
I wonder if someone can help a newby: I cannot get my installation of FreePBX to add, delete or update its outbound routes.
I am a FreePBX novice but have installed FreePBX 15.0.16.38 / Asterisk 16.6.1 and successfully set up extensions, app phones and a trunk to and from an ATA with FXO. Internal and Inbound calls are fine.
I set up a test outbound route with dial patterns that successfully placed calls to a limited set of test numbers via the ATA. However, when I have tried to extend the range of dial patterns accepted by that route, it would not allow the new number patterns to be sent to the route; calling results in the ‘your call cannot be completed, please check the number’ sequence.
Adding a new outbound route has not worked: the dial patterns for the new route are not being processed and still result in calls not being completed.
Deleting the original outbound route has not worked: the dial patterns contained within it are still being processed even after it has been deleted and no outbound rules are listed in FreePBX.
And yes: I have turned it all off and on again.
Am I doing something wrong or missing a step? Any ideas how I can get it to update the outbound rules?
Posts: 3
Participants: 2
@ALEXTORRES07 wrote:
Hello good afternoon:
I installed a UCM6204, it was installed to replace an old asterisk.
They are two remote sites (“AYB”) and there is communication between both sites through a VPN (IPSec), from any extension (80XX) of site “A” to any extension (90XX) of site “B” I can make a call without problem, but from any extension (90XX) of the site “B” to any extension (80XX) of the site “A” I cannot make the call, it throws me a message that says: “ALL CIRCUITS ARE OCCUPIED, PLEASE TRY MORE LATE”.NOTE:
- On the “A” site I have a FreePBX 12.0.76.4 installed with asterisk version 11.16.0.
- On site “B” is where I have installed UCM6204 (the one I replaced with the old asterisk) with version 1.0.19.27.
It is worth mentioning that I am using in both “IAX” trunk sites for communication between both sites, I performed a test with “SIP” trunks in both sites but it is the same result.
Then I share a little of what the Log of the UCM6204 (site “B”) throws at me when I make the call:
Dec 19 23:54:13 UCM6204 user.err asterisk: [20191219 23:54:13.412] ERROR[08503] netsock2.c:309: getaddrinfo(“nlu1s”, “(null)”, …): Name or service not known
Dec 19 23:54:16 UCM6204 user.err asterisk: [20191219 23:54:16.551] ERROR[11386] res_pjsip_header_funcs.c:490: This function requires a PJSIP channel.
Dec 19 23:54:16 UCM6204 user.err asterisk: [20191219 23:54:16.687] ERROR[11386] res_pjsip_header_funcs.c:490: This function requires a PJSIP channel.
Dec 19 23:54:16 UCM6204 user.err asterisk: [20191219 23:54:16.688] ERROR[11386] res_pjsip_header_funcs.c:490: This function requires a PJSIP channel.
Dec 19 23:54:16 UCM6204 user.err asterisk: [20191219 23:54:16.725] ERROR[08021] res_pjsip_header_funcs.c:446: No headers had been previously added to this session.
Dec 19 23:54:16 UCM6204 user.err asterisk: [20191219 23:54:16.791] ERROR[11386] res_pjsip_header_funcs.c:490: This function requires a PJSIP channel.
Dec 19 23:54:16 UCM6204 user.err asterisk: [20191219 23:54:16.792] ERROR[11386] res_pjsip_header_funcs.c:490: This function requires a PJSIP channel.Dec 27 07:04:02 UCM6204 local1.notice lowmem-killer[6346]: Asterisk VSZ:155220KB RSS:54904KB(limit 826948KB)
Dec 27 07:04:04 UCM6204 user.err asterisk: [20191228 07:04:04.849] ERROR[23510] res_pjsip_header_funcs.c:549: This function requires a PJSIP channel.
Dec 27 07:04:19 UCM6204 user.err asterisk: [20191228 07:04:19.777] ERROR[09524] netsock2.c:309: getaddrinfo(“nlu1s”, “(null)”, …): Name or service not known
Posts: 1
Participants: 1
@Whacka wrote:
I had a linksys router die, I bought an identical router to replace it. On the old router the IP address looked like this 10.0.161.XXX. But on the new router the IP addresses are like this 192.168.XXX.XX
I have used these instructions to attempt to fix it. Static IP for Asterisk computer
It shows up in my router as a connected network device, but I can’t ping it, and the webUI does not work. What should I try here?
Posts: 2
Participants: 2
@gregstahl wrote:
Hello Folks,
I’m running FreePbx 15.0.16 on Asterisk 16.6.
I’m unable to receive calls on my Voyant SIP Trunk UNLESS I have “Allow Annonymous Inbound SIP Calls” set to yes… If set to no, Caller gets “Number you have dialed is not in service”.
Single Voyant SIP Trunk for both Inbound and Outbound. Outbound calls work fine.
Asterisk Verbose shows the following on these inbound calls-
Executing [s@from-sip-external:7] Log(“SIP/corpglobal.net-000000bb”, "WARNING,“Rejecting unknown SIP connection from 137.192.80.49"”) in new stack
Voyant SIP Trunk-
Server is corpglobal.net
Outbond Proxy is reg-gw.sip.global Resolves to 137.192.80.49
Trunk SIP Settings (Outgoing PEER Details)
username=3157400109
type=friend
trustrpid=yes
sendrpd=no
secret=xxxxxxxxxx
qualify=yes
port=5060
outboundproxy=reg-gw.sip.global
nat=yes
keepalive=45
insecure=port,invite
host=corpglobal.net
fromuser=3157400109
fromdomain=corpglobal.net
dtmfmode=rfc2833
disallow=all
directmedia=no
context=from-pstn-toheader
allow=ulaw
Incoming USER details-
type=friend
insecure=invite
context=from-trunk
outboundproxy=reg-gw.sip.global
Register String-
3157400109@corpglobal.net:xxxxxxx@reg-gw.sip.global:5060/3157400109
Please Help, any advice would be greatly appreciated. I don’t want to leave Allow anonymous calls.
Thank you,
Greg
Posts: 2
Participants: 2
@hkjarral wrote:
Hello everyone,
My first post here, I am in a bit of pickle here. I was doing upgrade using this script
https://wiki.freepbx.org/display/PPS/Upgrading+from+FreePBX+10.13.66+to+SNG7
It passed 1st stage and on second stage it went into reboot loop, I tried all recovery but all in vain.
Anyhow long story short, Luckily I had full backup from backup/restore module on another server. So I installed identical setup on new server using the ISO.
The problem I am encountering is when I am trying to restore, everything goes fine but I don’t see anything in GUI. Even logs don’t give any errors,
# cat /var/log/asterisk/backup.log January 4, 2020, 3:59 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Initializing Backup 2 January 4, 2020, 3:59 pm - Backup Lock acquired! January 4, 2020, 3:59 pm - Running pre-backup hooks... January 4, 2020, 3:59 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Backup Lock acquired! January 4, 2020, 3:59 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Running pre-backup hooks... January 4, 2020, 3:59 pm - Adding items... January 4, 2020, 3:59 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Adding items... January 4, 2020, 3:59 pm - Building manifest... January 4, 2020, 3:59 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Building manifest... January 4, 2020, 3:59 pm - Creating backup... January 4, 2020, 3:59 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Creating backup... January 4, 2020, 3:59 pm - Storing backup... January 4, 2020, 3:59 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Storing backup... January 4, 2020, 3:59 pm - Running post-backup hooks... January 4, 2020, 3:59 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Running post-backup hooks... January 4, 2020, 3:59 pm - Backup successfully completed! January 4, 2020, 3:59 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Backup successfully completed! January 4, 2020, 3:59 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: END January 4, 2020, 5:48 pm - Initializing Restore... January 4, 2020, 5:48 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Initializing Restore... January 4, 2020, 5:48 pm - Running pre-restore hooks, if any... January 4, 2020, 5:48 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Running pre-restore hooks, if any... January 4, 2020, 5:48 pm - Restoring files (this may take some time)... January 4, 2020, 5:48 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Restoring files (this may take some time)... January 4, 2020, 8:36 pm - File restore complete! January 4, 2020, 8:36 pm - Restoring astDB... January 4, 2020, 8:36 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: File restore complete! January 4, 2020, 8:36 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Restoring astDB... January 4, 2020, 9:18 pm - Restoring Settings complete January 4, 2020, 9:18 pm - Running post-restore hooks, if any... January 4, 2020, 9:18 pm - Cleaning up... January 4, 2020, 9:18 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Restoring Settings complete January 4, 2020, 9:18 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Running post-restore hooks, if any... January 4, 2020, 9:18 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Cleaning up... January 4, 2020, 9:18 pm - Restore complete! January 4, 2020, 9:18 pm - Reloading... January 4, 2020, 9:18 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Restore complete! January 4, 2020, 9:18 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Reloading... January 4, 2020, 9:19 pm - Done! January 4, 2020, 9:19 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Done! January 4, 2020, 9:19 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: END January 4, 2020, 9:25 pm - Initializing Restore... January 4, 2020, 9:25 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Initializing Restore... January 4, 2020, 9:25 pm - Running pre-restore hooks, if any... January 4, 2020, 9:25 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Running pre-restore hooks, if any... January 4, 2020, 9:25 pm - WARNING! January 4, 2020, 9:25 pm - Web Root restore not detected, not restoring module table January 4, 2020, 9:25 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: WARNING! January 4, 2020, 9:25 pm - You should run "fwconsole moduleadmin upgradeall" to ensure system integrity January 4, 2020, 9:25 pm - Restoring astDB... January 4, 2020, 9:25 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Web Root restore not detected, not restoring module table January 4, 2020, 9:25 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: You should run "fwconsole moduleadmin upgradeall" to ensure system integrity January 4, 2020, 9:25 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Restoring astDB... January 4, 2020, 10:10 pm - Restoring Settings complete January 4, 2020, 10:10 pm - Running post-restore hooks, if any... January 4, 2020, 10:10 pm - Cleaning up... January 4, 2020, 10:10 pm - Restore complete! January 4, 2020, 10:10 pm - Reloading... January 4, 2020, 10:10 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Restoring Settings complete January 4, 2020, 10:10 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Running post-restore hooks, if any... January 4, 2020, 10:10 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Cleaning up... January 4, 2020, 10:10 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Restore complete! January 4, 2020, 10:10 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Reloading... January 4, 2020, 10:11 pm - Done! January 4, 2020, 10:11 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Done! January 4, 2020, 10:11 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: END January 4, 2020, 10:19 pm - Initializing Restore... January 4, 2020, 10:19 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Initializing Restore... January 4, 2020, 10:19 pm - Running pre-restore hooks, if any... January 4, 2020, 10:19 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Running pre-restore hooks, if any... January 4, 2020, 10:19 pm - Restoring files (this may take some time)... January 4, 2020, 10:19 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Restoring files (this may take some time)... January 4, 2020, 11:05 pm - File restore complete! January 4, 2020, 11:05 pm - Restoring astDB... January 4, 2020, 11:05 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: File restore complete! January 4, 2020, 11:05 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Restoring astDB... January 4, 2020, 11:48 pm - Restoring Settings complete January 4, 2020, 11:48 pm - Running post-restore hooks, if any... January 4, 2020, 11:48 pm - Cleaning up... January 4, 2020, 11:48 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Restoring Settings complete January 4, 2020, 11:48 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Running post-restore hooks, if any... January 4, 2020, 11:48 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Cleaning up... January 4, 2020, 11:48 pm - Restore complete! January 4, 2020, 11:48 pm - Reloading... January 4, 2020, 11:48 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Restore complete! January 4, 2020, 11:48 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Reloading... January 4, 2020, 11:48 pm - Done! January 4, 2020, 11:48 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Done! January 4, 2020, 11:48 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: END January 5, 2020, 4:53 pm - Initializing Restore... January 5, 2020, 4:53 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Initializing Restore... January 5, 2020, 4:53 pm - Running pre-restore hooks, if any... January 5, 2020, 4:53 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Running pre-restore hooks, if any... January 5, 2020, 4:53 pm - Restoring files (this may take some time)... January 5, 2020, 4:53 pm - id: bb2ac0b8da1f64a3498af147ba43fc10 data: Restoring files (this may take some time)...I have done all basics, Freepbx version is same, backup and restore version is same, I have done all basic troubleshooting I could do.
This is a production system and I have less than 14 hours left to bring it back up. Please help !!
Why isn’t my backup from older machine restoring correctly on new machine.
Posts: 1
Participants: 1
@mlaihk wrote:
Hi. I am wondering if anyone here have setup Freepbx 15/Asterisk 16 and running PJSIP with ipv6 clients?
I setup custom transports in the pjsip.transports_custom.conf as follows:
[ipv6-udp]
type=transport
protocol=udp
bind=[::]:5060
allow_reload=no
tos=cs3
cos=3the transport shows up when I go into the CLI->pjsip show transports
but my clients keep getting time out and no response errors. IPv4 setups works flawless so it is most likely a PJSIP ipv6 config issue… No connection attempt appear in the logs but the Freepbx GUI is accessible via ipv6, and I can use ipv6 to ssh into the server as well.Any insight you can share is much appreciated.
Posts: 1
Participants: 1
@a0429 wrote:
Hi
Can anyone help me to identify why my outbound CID is not being displayed? Instead it is always the main number of the company:
Extension setings:
Trunk settings:
Logs look like the extension CID is being set:
– Executing [s@macro-outbound-callerid:12] Set(“PJSIP/0000-000003a6”, “USEROUTCID=0911XXXXX”) in new stack
– Executing [s@macro-outbound-callerid:13] Set(“PJSIP/0000-000003a6”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:14] Set(“PJSIP/0000-000003a6”, “TRUNKOUTCID=0911XXXXX”) in new stack
– Executing [s@macro-outbound-callerid:15] GotoIf(“PJSIP/0000-000003a6”, “1?trunkcid”) in new stackThe SIP trunk goes to a Vega geteway which dials out using the calling party id:
Can anybody confirm if I am missing something obvious? When previously all working over ISDN this worked fine and showed the correct ID.
Thanks
Posts: 1
Participants: 1
@tdhendup wrote:
Dear All,
I am quite new to FreePBX. However I have installed the software on one of my servers. In the process, I could not make the use out of the FreePBX box. Therefore, I would like to seek the advises from this forum on the followingWhat all cards/equipment/hardware are required and whom can I approach so that I can make the use of my FreePBX server.
Regards
Posts: 1
Participants: 1
@jameeldroid wrote:
Hi, we are facing very strange issue, i am new to freepabx so i am not sure where to start looking for trouble? The version we are using is FreePBX 14.0.1alpha34.
everything was normal for years and then since last week i have started receive complains for no audio. here is the detail user told me.
when user A calls user B both hear no audio. When the same user A calls user C audio is working fine. and when user C calls user A and user B audio is working fine.
where should i look for trouble? this is really strange for me.
Posts: 1
Participants: 1
@djinn wrote:
New Dell T30 server with Sangoma A200 cards, with 4 FXO card - we actually have two cards each with 4 FXO modules, one set is intended to be a spare. The cards are new as are the FXO Modules. FreePBX is the latest version. After DAHDI setup FreePBX sees the card and FXO ports, and all 4 ports work.
The system is replacing an older PBX. Lines are analog lines coming from an Xfinity Business analog phone box. All 4 lines work perfectly in the old PBX system, and work perfectly if you connect an analog phone directly to the FXS ports on the Xfinity box.
The problem is if we connect the lines to the Sangoma A200 system only 2 of the lines are picked up. It doesn’t matter which port we put them in, only two of the lines receive calls. If I watch the call progress using ‘asterisk -vvvvvr’ I can see the calls come in on the 2 lines that work, the others absolutely nothing happens as if the calls never hit the system.
We’ve replaced the card, and FXO modules, and rerun the setup just in case there is some issue. Fax is disabled in the FreePBX Global Settings and I confirmed they are disabled in wanpipe1.conf.
Xfinity says there is nothing wrong with the box or lines, and won’t offer any further support.
Any idea what else I can do to get these lines working with FreePBX?
Posts: 1
Participants: 1
@ptravel wrote:
Early this morning, FreePBX (latest distro) apparently decided to call itself. The CID was the DID for one of my trunks (an analog line that comes in through a Cisco SPA8080). In looking at the logs, I noticed this line, which puzzles me. Any idea what’s going on?
Set(“PJSIP/9493359290-00000021”, “FROMEXTEN=unknown”)
Posts: 10
Participants: 5
