@trixie_no5 wrote:
Do the results of sip set debug on or pjsip set logger on output results to a file? It is difficult to capture the outputs from the command line.
Posts: 3
Participants: 3
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SIP set debug
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Dahdi Install
@v70ff wrote:
This is my 3rd reinstall of freePBX to try to resolve this issue each time using a different version
I can’t seem to figure out this dahdi issue.“DAHDi Doesn’t appear to be running. Click the ‘Restart DAHDi & Asterisk’ button below”
I would click the restart button but it changes nothing?I see no results under analog hardware but if i run dahdi_hardware I can see the card.
dahdi_hardware
pci:0000:02:08.0 wctdm24xxp- d161:8006 Wildcard AEX410Pstrangely if I try any other dahdi commands i get this error
“Unable to open /dev/dahdi/ctl: No such file or directory” or “permission denied”
I’m currently running FreePBX 14.0.1.24
Linux 3.10.0-693.11.1.el7.x86_64
DAHDI Tools Version - 2.11.1Any sort of help would be welcomed. If you’re currently using a system running with dahdi config please share your knowledge or at least your os version and dahdi tools version .
Posts: 1
Participants: 1
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AMI port forwarding?
@mvogel4949 wrote:
Probably a really stupid question, but do I need a specific port on the firewall open for remote access to the AMI? Thanks
Posts: 6
Participants: 3
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How to determine which party hangs up the calls
@faisalkhan wrote:
Hi all,
How can I determine which party hangs up the call.
Posts: 9
Participants: 5
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Call line is busy?
@jpark1205 wrote:
Hello!
Currently, I am working on a project creating the callcenter program. Everything was setup up and the call worked for both inbound and outbound but, after the certificate update the phone call does not work anymore. It displays the error saying its busy on the inspect console. I have no idea what to do next…
Posts: 2
Participants: 2
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Tts module
@AmidouFlorian92 wrote:
Hi
After installing tts module I got this error and I cannot access to freepbx admin GUI
Please help me
- Whoops\Exception\ErrorException->() /var/www/html/admin/modules/texttospeec h/functions.inc.php:25
Posts: 1
Participants: 1
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FreePBX ISO Download link
@Elision wrote:
Hello,
I need to download freepbx 2.11.0.41 version’s ISO.
So please if any have link for this so provide me.Regards,
Elision
Posts: 5
Participants: 3
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Yum install php-mssql
@jeetzz wrote:
Hi
I am getting this error, when i hit “yum install php-mssql” I am will to change the script, but what do i go for in order to connect SQL Server Database 2008 R2…any ideas would be helpful.
[root@freepbx ~]# yum install php-mssql Loaded plugins: fastestmirror, versionlock Loading mirror speeds from cached hostfile Resolving Dependencies --> Running transaction check ---> Package php-mssql.x86_64 0:5.4.16-9.el7 will be installed --> Processing Dependency: php(zend-abi) = 20100525-64 for package: php-mssql-5.4.16- 9.el7.x86_64 --> Processing Dependency: php(api) = 20100412-64 for package: php-mssql-5.4.16- 9.el7.x86_64 --> Processing Dependency: libsybdb.so.5()(64bit) for package: php-mssql-5.4.16-9.el7.x86_64 --> Running transaction check ---> Package freetds.x86_64 0:0.95.81-1.el7 will be installed ---> Package php-common.x86_64 0:5.4.16-46.el7 will be installed --> Processing Dependency: libzip.so.2()(64bit) for package: php-common-5.4.16- 46.el7.x86_64 --> Running transaction check ---> Package libzip.x86_64 0:0.10.1-8.el7 will be installed --> Processing Conflict: php56w-common-5.6.40-1.sng7.x86_64 conflicts php-common < 5.6 --> Finished Dependency Resolution Error: php56w-common conflicts with php-common-5.4.16-46.el7.x86_64 You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigestAppreciate your help.
Jeet
Posts: 1
Participants: 1
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PJSIP update external IP address automatically?
@avayax wrote:
How do I have my external IP address in Freepbx updated automatically when using PJSIP?
I am behind a dual wan situation and have DDNS, where my FQDN updates automatically.I understand that there is the External Address setting under General SIP Settings, where you can press the Detect Network Settings to update the external IP address manually.
If you want that to happen automatically for chansip you can provide a Dynamic Host under Chansip Settings and set a refresh intervall.How do I do the same thing with PJSIP?
There is a External IP Address field under PJSIP settings, is that where I put my FQDN?
Does it update periodically like chansip?
Posts: 10
Participants: 3
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Freepbx and asterisk log files
@AmidouFlorian92 wrote:
Hi
Sorry again for the inconvenience
I’m developping a script to make dynamic announcements on freepbx and I’m searching the directory of 2 log files :
The location of the file a inconming call or for a call in progress
The location of the file to find the history of digits when someone presses when he calls my IVR
Posts: 1
Participants: 1
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Asterisk CLI message
@AmidouFlorian92 wrote:
Hi I have this message on my asterisk CLI. I just want to stop showing this how can I do?
[2019-08-22 13:23:35] NOTICE[11077]: chan_sip.c:28752 handle_request_register: Registration from ‘“1700” sip:1700@199.16.131.19’ failed for ‘77.247.110.201:6388’ - Wrong password
Posts: 1
Participants: 1
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International call to china fails
@snaplink wrote:
We’re able to call about a dozen international countries but calls to China show as failed in the disposition field in CDR reports. Is there a way to dig deeper and see why it failed or where it failed?
We’re using flowroute and China is in the allowed countries list. The dial plan for international appears to be correct as we can call the other dozen or so countries with no problem, it’s only China that fails.
Can this be done from the Web Interface? If not I do have access to the CLI of the server itself but I’m not a high level linux guy so would need some guidance.
Thank you!
Posts: 1
Participants: 1
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Record new custom greeting with OPUS enabled
@jfrey40535 wrote:
I submitted a bug report this evening and am wondering if any other FPBX admins have encountered a similar issue with recording new system recordings with OPUS enabled.
Bug report:
With OPUS enabled, when end users dial the feature code to listen to and change the recording, the old greeting plays, system says to rerecord press *. They press *, it says start recording, then immediately says press 1 to listen 2 to keep. It never allows them to actually record anything.
We disabled OPUS codec and were then able to record a new greeting normally.
Posts: 4
Participants: 2
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Inbound calls
@v70ff wrote:
I have just created a dahdi trunk
I then setup inbound route and set it to ring a specific extension but when i call the extension does not ring.
My extension is setup using chan_sip could it be the case that it should be a dahdi extension?
hopefully the logs helpMy pstn is in analog 1
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-analog:1] NoOp(“DAHDI/1-1”, "Entering from-dahdi with DID == ") in new stack
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-analog:2] Ringing(“DAHDI/1-1”, “”) in new stack
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-analog:3] Set(“DAHDI/1-1”, “DID=s”) in new stack
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-analog:4] NoOp(“DAHDI/1-1”, “DID is now s”) in new stack
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-analog:5] GotoIf(“DAHDI/1-1”, “1?dahdiok:checkzap”) in new stack
VERBOSE[6431][C-00000001] pbx_builtins.c: Goto (from-analog,s,9)
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-analog:9] NoOp(“DAHDI/1-1”, “Is a DAHDi Channel”) in new stack
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-analog:10] Set(“DAHDI/1-1”, “CHAN=1-1”) in new stack
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-analog:11] Set(“DAHDI/1-1”, “CHAN=1”) in new stack
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-analog:12] Macro(“DAHDI/1-1”, “from-dahdi-1,s,1”) in new stack
WARNING[6431][C-00000001] app_macro.c: No such context ‘macro-from-dahdi-1’ for macro ‘from-dahdi-1’. Was called by s@from-analog
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-analog:13] NoOp(“DAHDI/1-1”, “Returned from Macro from-dahdi-1”) in new stack
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-analog:14] Goto(“DAHDI/1-1”, “from-pstn,s,1”) in new stack
VERBOSE[6431][C-00000001] pbx_builtins.c: Goto (from-pstn,s,1)
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-pstn:1] NoOp(“DAHDI/1-1”, “No DID or CID Match”) in new stack
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-pstn:2] Answer(“DAHDI/1-1”, “”) in new stack
WARNING[6431][C-00000001] chan_sip.c: This function can only be used on SIP channels.
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-pstn:3] Log(“DAHDI/1-1”, "WARNING,Friendly Scanner from ") in new stack
WARNING[6431][C-00000001] Ext. s: Friendly Scanner from
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-pstn:4] Wait(“DAHDI/1-1”, “2”) in new stack
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-pstn:5] Playback(“DAHDI/1-1”, “ss-noservice”) in new stack
VERBOSE[6431][C-00000001] file.c: <DAHDI/1-1> Playing ‘ss-noservice.ulaw’ (language ‘en’)
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-pstn:6] SayAlpha(“DAHDI/1-1”, “”) in new stack
VERBOSE[6431][C-00000001] pbx.c: Executing [s@from-pstn:7] Hangup(“DAHDI/1-1”, “”) in new stack
VERBOSE[6431][C-00000001] pbx.c: Spawn extension (from-pstn, s, 7) exited non-zero on ‘DAHDI/1-1’
VERBOSE[6431][C-00000001] pbx.c: Executing [h@from-pstn:1] Macro(“DAHDI/1-1”, “hangupcall,”) in new stackVERBOSE[6431][C-00000001] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“DAHDI/1-1”, “1?theend”) in new stack
VERBOSE[6431][C-00000001] pbx_builtins.c: Goto (macro-hangupcall,s,3)
VERBOSE[6431][C-00000001] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“DAHDI/1-1”, “0?Set(CDR(recordingfile)=)”) in new stack
VERBOSE[6431][C-00000001] pbx.c: Executing [s@macro-hangupcall:4] NoOp(“DAHDI/1-1”, " montior file= ") in new stack
VERBOSE[6431][C-00000001] pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“DAHDI/1-1”, “1?skipagi”) in new stack
VERBOSE[6431][C-00000001] pbx_builtins.c: Goto (macro-hangupcall,s,7)
VERBOSE[6431][C-00000001] pbx.c: Executing [s@macro-hangupcall:7] Hangup(“DAHDI/1-1”, “”) in new stack
VERBOSE[6431][C-00000001] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘DAHDI/1-1’ in macro ‘hangupcall’
VERBOSE[6431][C-00000001] pbx.c: Spawn extension (from-pstn, h, 1) exited non-zero on ‘DAHDI/1-1’
VERBOSE[6431][C-00000001] sig_analog.c: Hanging up on ‘DAHDI/1-1’VERBOSE[6431][C-00000001] chan_dahdi.c: Hungup ‘DAHDI/1-1’
Posts: 1
Participants: 1
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Playing Back Video left in voice mail
@harmfultoast wrote:
I would like to offer recorded video in voicemail as messages. It works when I build my own Asterisk platform running lean. Voice and Video in perfect sync during playback. Run the same process (H.264) of FreePBX and the video just spoon-feeds through frame by frame. Using Grandstream GXV32xx series in both tests.
What was cool in my test was that I used 10 frame splash videos in IVR so that users get audio visual feed back as they navigate through the menus. Want to offer this up in my multi-customer IAX2 service customer community server. Video-in-voicemail seems to be almost taboo in the Asterisk community. Can somebody point me to an expert? Still researching
Thanks, Brian
Posts: 2
Participants: 1
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I changed the call recording location to a SMB folder but no files are being created
@shivansps wrote:
Hi,
sorry i had previusly opened a thread but i went away and is now closed.
I connected to my SMB shared using fstab, the share is mounted to /mnt/recordings, i tested from terminal and i can create and delete files on the smb share from there.But after changing the override call location to “/mnt/recordings” nothing is happening, no files are being created there, but there seem to be no errors as well.
On termical i checked using asrerisk -rvvv and no errors or warnings are show, whats more, i see mix monitor getting the correct folder.
-- Executing [recordcheck@sub-record-check:16] NoOp("SIP/", "Starting recording: in, ") in new stack -- Executing [recordcheck@sub-record-check:17] Set("SIP/", "__CALLFILENAME=in--20190822-194611-1566503171.2845") in new stack -- Executing [recordcheck@sub-record-check:18] MixMonitor("SIP/", "/mnt/recordings/2019/08/22/in-20190822-194611-1566503171.2845.wav,abi(LOCAL_MIXMON_ID),") in new stack -- Executing [recordcheck@sub-record-check:19] Set("SIP/", "__MIXMON_ID=0x7f8810012240") in new stack -- Executing [recordcheck@sub-record-check:20] Set("SIP/", "__RECORD_ID=SIP") in new stack -- Executing [recordcheck@sub-record-check:21] Set("SIP/", "__REC_STATUS=RECORDING") in new stack -- Executing [recordcheck@sub-record-check:22] Set("SIP/", "CDR(recordingfile)=in--20190822-194611-1566503171.2845.wav") in new stack -- Executing [recordcheck@sub-record-check:23] Return("SIP/T", "") in new stackthen at the end
== End MixMonitor Recording SIP/
Posts: 1
Participants: 1
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Cisco 7942g wont register with FreePBX
@nathan_morganIT wrote:
I’ve been trying to get a 7942g working with FreePBX for what seems like months. I have tried different firmware versions, using different SEP(MAC) templates, trying different websites for instructions for the config file, different options in the TFTP software on my PC, and it still won’t.
I have also tried sip debugging in the Asterisk CLI: “asterisk -vvvvvr”, and sometimes it gives me a yellow NOTICE tag and says it’s trying to register from my servers IP to the Phones IP but can’t because “wrong password”. The secret in the Cisco’s config file matches the one for the extenion(s). The phone just gets stuck on “registering” for a few minutes but won’t actually do anything and the line keys have a little x next to the phone icon plus no dial tone.
I am using CHAN_SIP in FreePBX and changed the port in the phones config file to 5160 since that’s what FreePBX CHAN_SIP listens on. Still nothing.
When I factory reset the phone and plug the phone and server into a switch without the actual network(router cable) connected, the phone will boot and the files go through to the phone successfully. Then when it gets to “registering” I unplug the server from the switch+relocate the switch and phone and plug in the actual network cable to give the phone internet. Won’t register. I’ve also tried restarting the phone with the actual network cable originally plugged in and without the server plugged in, won’t register. Should I have the server connected to the actual network too?
I’m quite frustrated and don’t know what else could be going wrong.
Posts: 1
Participants: 1
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CallerID settings
@NorColorNorName wrote:
Hello,
My task is to set a custom CallerID.
I have a webRTC sipML5 softphone, and i want the user can choose there CallerID with the softphone.
But first i just want to change the callerID for all extensions to understand how the callerID can be modify. Right after that i will get the variable the sipML5 send me to modify the callerID
Can someone show me the way?
I already try to change the CallerID in the trunk settings and outbound settings. But no luck the callerid when i call my personnal phone is the same.Any advices will help !
Thanks by advance.
Regards,
NCNN.
Posts: 2
Participants: 1
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Some time after boot up, calls fail & phones won't register
@nickb wrote:
On one of my PBX installations, running 12.7.6-1904-1.sng7 (showing all updates installed in yum and modules - fully updated system), some time after boot (a day, a few days, maybe a week), calls will stop coming through. The asterisk log shows no indication of any call received.
In addition, the SIP endpoints (Grandstream GXP1628) all show “no response” on the screen.
I have restarted asterisk, with no results; and stopped the firewall, again with no results. There are no useful log entries in the asterisk or system logs. It all just simply stops working, and the only action that clears it up (temporarily) is to reboot the system.
I’ve requested logs from my SIP provider to see what they show from their end as well, but I don’t have this information yet.
Any suggestions as to what might be the issue?
Posts: 4
Participants: 2
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Import phonebook command line
@Muso wrote:
Is there a way to import phonebook from csv with command line and schedule the job?
Posts: 2
Participants: 2
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