@lemming86_au wrote:
Hi All,
I’m trying to set up a PJSIP trunk between two FreePBX servers, but I’m not having much luck.
Does anyone have a barebones config for a working trunk they could share?
Thanks,
Josh
Posts: 3
Participants: 2
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PJSIP trunk between servers
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SQL Table recover
@xentity wrote:
hi there,
may it be possible that someone please dumps his definition of the sql tables:
- ‘asterisk.incoming’
- ‘asterisk.ivr_details’
- ‘asterisk.kvstore_FreePBX_modules_Conferences’
Had some file system issues and the backup was also effected (transferring the backup to another system was an open point… shame on me).
However, I was able to fetch all config files (asterisk is working) and most of the DB tables. But the tables mentioned above were not able to recover.The following commands
mysqldump --no-data -u root asterisk incoming mysqldump --no-data -u root asterisk ivr_details mysqldump --no-data -u root asterisk kvstore_FreePBX_modules_Conferencesshould do the trick on a most recent system.
The content of incoming should be restorable by the content of extensions_additional.conf, is that right? I didn’t use IVR nor Conferences, therefore the content of those should be avoidable, right? But the tables are needed for the module updates.
Thx a lot.
Posts: 2
Participants: 1
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FreePBX web interface behind Cloudflare CDN?
@v4704 wrote:
Hi, I’d like to provide UCP to some of my users but wouldn’t like to expose the server IP address.
Is it going to work well if I provide them with a domain behind Cloudflare for UCP purposes, while I access the admin panel and extensions (port 5060 and so on) directly with the server IP to work around the CDN?
Or if you guys have any other suggestions to accomplish the same thing.
Thanks!
Posts: 1
Participants: 1
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911 Dispatch Location
@MollyMae wrote:
Greetings,
I have an issue and I’m not sure where to address the concern. It’s probably not FreePBX (I believe it’s the SIP provider), but I’m hoping I can confirm that before reaching out.I recently configured E911 CIDs for a company with 5 locations. The CID information (including the address) is passed correctly to the dispatcher, but when I dial 911 it goes to the county dispatch instead of the city dispatch. I confirmed with the county dispatch that based on the address, it should be going to the city.
I use SIPStation, so I will be reaching out to them next unless someone knows what else could be the issue. I don’t know if any config information would help, since I think FreePBX is passing out the Emergency Route correctly, but I’ll gladly provide any information requested.
Thanks,
Molly
Posts: 1
Participants: 1
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Inbound dynamic routes based on phonebook or CID
@justinm001 wrote:
We’re having a massive issue with robocalls and sales calls. Blacklist helps with 5% of these. Is there a way we can have all calls go through a filter and if they’re in a “approved list” it’ll go to a destination, but if not on the “approved list” then it’ll go to another destination? I’m thinking the not approved will go to IVR where they’ll at least go to some prompts before getting to a receptionist who will then transfer calls.
Also would love an easy way to update that list by either pressing something like *33 to “approve” them or quick website to “approve” them.
I know there’s a custom module called dynamic route but looks like not updated in a few years so would rather have a solution that is better managed.
Posts: 2
Participants: 2
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Copyright page loading
@ianarman wrote:
Hello,
My freepbx instance is only displaying the copyright information, when logging in.
0 System Admin 14.0.38.3 Copyright 2019 by Sangoma Technologies Inc., All rights reserved By installing, copying, downloading, distributing, inspecting or using the materials provided herewith, you agree to all of the terms of use as outlined in our End User Agreement which can be found and reviewed at https://www.freepbx.org/legal/
Posts: 1
Participants: 1
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Full HDD FreePBX
@Pletev wrote:
Hello, The system reports to me:
Storage space is getting high on the following drives of your system:
/dev/sda2 is 76% fullUnfortunately I don’t have access to SSH consoles. Can I delete old backups from the web interface?
Thank you for the advice.
Posts: 6
Participants: 4
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Ivr will not answer if ring time is more than 10 seconds
@GaryCameron wrote:
I have 2 phones in a ring group that handle all outside calls. If the ring group does not answer the call an IVR is supposed to answer. If i set the ring time for more than 10 seconds the IVR will not pick up. The phone continues to ring until the call times out. I have compared all the programming with our other server and it matches exactly. That server has ring time set to 25 seconds. I have also compared the programming on all of our Vega 60G gateways and they all match. Looking for a suggestion on what could possibly be the problem. Thanks
Posts: 2
Participants: 2
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Sangoma s705 Pass Caller ID to extension on transfer
@bajramia wrote:
Hi,
I have Sangoma S705 phones i want to pass CallerID to extension on transfer my extension setting is Trust RPID =YES and Send RPID= Send P-Asserted-identity headerwhen i do transfer is not working
thank you
Posts: 1
Participants: 1
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Full HDD FreePBX
@Pletev wrote:
Hello, The system reports to me:
Storage space is getting high on the following drives of your system:
/dev/sda2 is 76% fullUnfortunately I don’t have access to SSH consoles. Can I delete old backups from the web interface?
Thank you for the advice.
Posts: 6
Participants: 4
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Freepbx Failing to load retrieve_conf
@Smurfturf wrote:
I updated the zulu module of a system today and immediately after, it stopped processing calls. I get an error message that I need to run /var/lib/asterisk/bin/retrieve_conf. When I log into the sytem, here’s the output I see:
[root@ ~]# /var/lib/asterisk/bin/retrieve_conf
[FATAL] retreive_conf failed to get engine information and cannot configure up a softwitch with out it. Error: ERROR-UNABLE-TO-PARSEI’ve been able to narrow it down to this piece of code in retrieve_conf:
$engineinfo = engine_getinfo();
if($engineinfo[‘version’] == 0){
fatal(sprintf(_(“retreive_conf failed to get engine information and cannot configure up a softwitch with out it. Error: %s”),$engineinfo[‘engine’]),true);Does anyone have any suggestions on how to get this system running again?
Posts: 2
Participants: 1
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"You have one new message"-Announcement: "One" can't be played
@Quafi wrote:
When calling a mailbox with only one message in it, the system can’t play the “one” in “You have one new message”. This is the case for only the one, every other number I’ve tried works. As soon as there are two or more calls, everything’s fine. I’ve also tried both german (that’s what the system is natively set to) and english, to verify there’s nothing wrong with the language packs. Happens in both.
Here you can see the error message I’m getting:
Any advise on how to fix this? I’m using FreePBX Version 14.0.13.6, System Version 12.7.6-1904-1.sng7. All modules are up do date.
Posts: 1
Participants: 1
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No audio when calling trunk on the same system
@Quafi wrote:
Hey there! I have three trunks configured on my pbx. When calling them from external numbers, they all work perfectly fine. Calling other extensions internally also works flawlessly. But whenever I try to call one of our external numbers from an internal extension there’s no audio. Phones ring normally, you can pick up the call, but on no side can you hear anything. Any idea why this is happening?
Posts: 1
Participants: 1
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Warning ast_yyerror() after last updates
@Strato83 wrote:
Hello.
15.10.2019 i’ve updated several Freepbx 14 installations to the latest module versions. It seems everything is working fine, but after update i’ve started to get warnings in the CLI of all updated PBX. This warning appears only on outbound calls.
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:1] NoOp(“SIP/210-00004c32”, “210”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:2] NoOp(“SIP/210-00004c32”, “”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:3] NoOp(“SIP/210-00004c32”, “off”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:4] ExecIf(“SIP/210-00004c32”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:5] ExecIf(“SIP/210-00004c32”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:6] ExecIf(“SIP/210-00004c32”, “0?Set(REALCALLERIDNUM=210)”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:7] ExecIf(“SIP/210-00004c32”, “0?Set(AMPUSER=210)”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:8] GotoIf(“SIP/210-00004c32”, “1?normcid”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx_builtins.c: Goto (macro-outbound-callerid,s,12)
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:12] Set(“SIP/210-00004c32”, “USEROUTCID=43210”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:13] Set(“SIP/210-00004c32”, “EMERGENCYCID=”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:14] Set(“SIP/210-00004c32”, “TRUNKOUTCID=78332256466”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:15] GotoIf(“SIP/210-00004c32”, “1?trunkcid”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx_builtins.c: Goto (macro-outbound-callerid,s,21)
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:21] ExecIf(“SIP/210-00004c32”, “1?Set(CALLERID(all)=78332256466)”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:22] ExecIf(“SIP/210-00004c32”, “1?Set(CALLERID(all)=43210)”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:23] ExecIf(“SIP/210-00004c32”, “1?Set(CALLERID(all)=78332256466)”) in new stack
[2019-10-15 09:22:50] WARNING[1178][C-0000262a] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected ‘=’, expecting $end; Input:
** = 1 & 0 = 0**
** ^**
[2019-10-15 09:22:50] WARNING[1178][C-0000262a] ast_expr2.fl: If you have questions, please refer to link to asterisk org
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:24] ExecIf(“SIP/210-00004c32”, “?Set(CALLERID(all)=210)”) in new stack
[2019-10-15 09:22:50] WARNING[1178][C-0000262a] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected ‘=’, expecting $end; Input:
** = 1 & 0 = 0**
** ^**
[2019-10-15 09:22:50] WARNING[1178][C-0000262a] ast_expr2.fl: If you have questions, please refer to link to asterisk org
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:25] ExecIf(“SIP/210-00004c32”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:26] ExecIf(“SIP/210-00004c32”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:27] Set(“SIP/210-00004c32”, “CDR(outbound_cnum)=78332256466”) in new stack
[2019-10-15 09:22:50] VERBOSE[1178][C-0000262a] pbx.c: Executing [s@macro-outbound-callerid:28] Set(“SIP/210-00004c32”, “CDR(outbound_cnam)=”) in new stack
Posts: 1
Participants: 1
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Play announcement to callee on outbound call
@Quafi wrote:
Hey there,
I’ll try to briefly explain what I want to accomplish:
Whenever someone calls an external number, I want to route this callee to an announcement as soon as he picks up. They’re supposed to hear a greeting informing them that the call is automatically recorded, then be redirected to a queue with high priority or eventually the destination the call has been originated from, but that sounds even harder to accomplish
Posts: 1
Participants: 1
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IVR trouble
@chasemixon wrote:
So a user called me this morning saying that all my calls are going to a different user, at first I was like sure they are… well that’s persons extension was 151 and the IVR had the other person as 1, 5 so it made sense the person was probably just dialing the 151 too slow and it was going to the other phone right… NOPE I dialed in using my Cell and pressed 151 as soon as the IVR picked up. and it went to the other person… so it seems like Enable Direct Dial in the IVR isn’t working as it used too? Can someone tell me what I can do to resolve this?
Posts: 2
Participants: 1
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Calls Randomly Disconnecting
@rmatteson wrote:
I have several users that are reporting dropped calls. I noticed something in the Call Event Log for those calls which seems to be different than calls that are not dropping, but I’m not sure what it tells me or if it’s significant at all.
Normal calls seem to look like this:
|CHAN_START|from-internal|
|CHAN_START|from-trunk|
|ANSWER|from-trunk|
|ANSWER|macro-dialout-trunk|
|BRIDGE_ENTER|from-trunk|
|BRIDGE_ENTER|macro-dialout-trunk|
|BRIDGE_EXIT|macro-dialout-trunk|
|BRIDGE_EXIT|from-trunk|
|HANGUP|from-trunk|
|CHAN_END|from-trunk|
|HANGUP|restrictedroute-c4ca4238a0b923820dcc509a6f75849b|
|CHAN_END|restrictedroute-c4ca4238a0b923820dcc509a6f75849b|
|LINKEDID_END|restrictedroute-c4ca4238a0b923820dcc509a6f75849b|
Note that both parties seem to BRIDGE_EXIT before the HANGUP and CHAN_END occurs.Here’s what the dropped calls look like:
|CHAN_START|from-internal|
|CHAN_START|from-trunk|
|ANSWER|from-trunk|
|ANSWER|macro-dialout-trunk|
|BRIDGE_ENTER|from-trunk|
|BRIDGE_ENTER|macro-dialout-trunk|
|BRIDGE_EXIT|from-trunk|
|HANGUP|from-trunk|
|CHAN_END|from-trunk|
|BRIDGE_EXIT|macro-dialout-trunk|
|HANGUP|restrictedroute-c4ca4238a0b923820dcc509a6f75849b|
|CHAN_END|restrictedroute-c4ca4238a0b923820dcc509a6f75849b|
|LINKEDID_END|restrictedroute-c4ca4238a0b923820dcc509a6f75849b|
Note that the external side goes through BRIDGE_EXIT, HANGUP, and CHAN_END before the internal side gets to BRIDGE_EXIT.Is this an indication that the call is dropping due to an issue at the SIP Provider (or carrier on the external end) or is this just a concidence that this pattern seems to appear on the dropped calls?
Posts: 1
Participants: 1
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Scheduler and Alerts
@mcottone wrote:
I’m current running FreePBX 15x and have a quick question that seems to be hard to find and answer too. Some I’m reaching out to the community for an answer.
Within the following module:
Admin | Module Admin | Scheduler and AlertsI’m looking for a script to run from the command line, that would set the following settings?
- Email Address —) none or noemail@nodomain.com
- System Identifier —) VoIP Server
- Automatic Module Updates —) Disabled
- Automatic Module Security Updates —) Email Only
- Send Security Emails For Unsigned Modules —) Disabled
Posts: 1
Participants: 1
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I cant place a call after an update
@NetBrowser wrote:
Hello colleagues how are you today…
I hade an installation working and after I have made an update to all the modules I have lost some functionallity.
I had a basic configuration using DIDFORSALE and everything was working fine until i’ve made the update, after i’ve made the update i cant place a call.
I can receive calls but when i try to make a call the line sounds busy and the call is not going through.
If I go to the CDR i can see the call and the message is BUSY, I have tryed adding the + sign and creating all sets of dial patterns like the following.
NXXNXXXXX
1NXXNXXXXXX
+1NXXNXXXXXX
etc…
Posts: 6
Participants: 4
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CDR server
@smwein wrote:
Hi all,
I had my CDR server on remote server. Now I’m trying to change it back to my phone server(the original) server. Please let me know what I need to do.
Thanks
Posts: 4
Participants: 2
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