have you checked UCP → Voicemail settings? Maybe VM users can be record own language a messages ?
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Change Voicemail language
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Issues on Inbound Call V17
You have to dig this message, Same message sngrep showing to you.
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Change Voicemail language
Sadly that woud not work i am trying to change the language when i dail *97 so that woud be englisch and the Rest of the Extensions woud stay in german
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Issues on Inbound Call V17
It seems like configuration level issue than an infra level issue. Provider have confirmed that their infrastructure is working well and we are able to make outgoing call without any problems.
When I dial a number it jumps from CID to different DID and CDR also makes 3-4 entries. I don’t understand why call to ****2244 have jumped to ****2222
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Missedcallnotify on UNVAILABLE
I will give that a try. Thnaks a lot.
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Issues on Inbound Call V17
Without a console log from Asterisk I can only guess, but it honestly looks like it’s looping back to the provider, back to you, back to the provider, back to you.
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Issues on Inbound Call V17
That is true but I don’t reason on why is that happening. I am also having backup setup of Freepbx 16 where everything was working well and I don’t see this observation on freepbx 16. Things that have changed is Freepbx 16 → 17 and Asterisk 18->20. Driver is same chan_sip.
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Issues on Inbound Call V17
Providing an Asterisk log[1] would likely yield more insight.
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FreePBX 17 - PhoneBook
Hello all. I’m new to this forum and newbie @FreePBX. I work with 3CX but want to change. I just install a new FreePBX system.
Extensions are OK and Incoming and Outgoing calls works fine.
It’s stil a good start 
I’ve some Yealink phones and I wan to create a directory/contacts list with a synchronisation phone ↔ FrePBX. I create in Admin > Contacts manager > External some contacts. I want to see them in the phones.
I read that I need to create an XML File to put in /var/www/html/contacts and after that in the phone is accessible by http://MyFreePBXIP/contacts/yealink.xml or http://MyFreePBXIP:8089/applications/rest/contacts/yealink, can you juste help me for this part please ?
Thanks for your advices.
Have a nice day,
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FreePBX 17 - PhoneBook
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Issues on Inbound Call V17
You have somehow created a loop. You need to show actual setup of the Inbound Route and the Extension along with your Outbound Routes.
Provider = 10.0.xx.2
PBX = 10.0.xx.106
You will notice the incoming calls comes in and right after the PBX sends the 100 Trying to the provider for the incoming, the PBX then sends the call right back to the provider which then loops back into the system it looks like.
Every message in this output is either from 10.0.xx.2 or 10.0.xx.106. Never once is another device/endpoint involved in this call.
Here’s the first INVITE sent back to the provider after the initial 100 Trying is sent:
Note how the PBX is sending the call that should be for the extension back to the provider.
2025/03/17 10:08:43.482256 10.0.xx.106:5060 → 10.0.xx.2:5060
INVITE sip:****2244@10.0.xx.2 SIP/2.0
Via: SIP/2.0/UDP 10.0.xx.106:5060;branch=z9hG4bK3cf17b66;rport
Max-Forwards: 70
From: sip:****2222@10.0.xx.2;tag=as5537d8b2
To: sip:****2244@10.0.xx.2
Contact: sip:****2222@10.0.xx.106:5060
Call-ID: 5aee03ec2700bc0331ed61ed67808f04@10.0.xx.2
CSeq: 102 INVITE
Here’s the provider giving a 100 Trying to the INVITE
2025/03/17 10:08:43.502839 10.0.xx.2:5060 → 10.0.xx.106:5060
SIP/2.0 100 Trying
From: sip:****2222@10.0.xx.2;tag=as5537d8b2
To: sip:****2244@10.0.xx.2
Call-ID: 5aee03ec2700bc0331ed61ed67808f04@10.0.xx.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.xx.106:5060;rport=5060;branch=z9hG4bK3cf17b66
Contact: sip:****2244@10.0.xx.2:5060;transport=UDP
Content-Length: 0
Here’s the provider sending that same call back into the PBX
2025/03/17 10:08:43.530559 10.0.xx.2:5060 → 10.0.xx.106:5060
INVITE sip:2244@10.0.xx.106:5060;transport=UDP SIP/2.0
From: "0792222"sip:79****2222@10.0.xx.2:5060;transport=UDP;tag=BN1884504541-1-1742206123-924972956
To: "79****2244"sip:79****2244@10.0.xx.106:5060;transport=UDP
Call-ID: 3a22e172-737d2013-f5fed-7efb42fe8000-6eb330a-13c9-764
CSeq: 1751 INVITE
Via: SIP/2.0/UDP 10.0.xx.2:5060;branch=z9hG4bK-ae05be1-f5fed-3c0eb932-7efb6c2dbee8
Route: sip:10.0.xx.106:5060;transport=UDP;lr
X-SessionId: 1884504541
User-Agent: BN4000-3.9.1-575
Supported: timer,100rel
Max-Forwards: 67
X-ZTEAG: 0001
Contact: sip:79****2222@10.0.xx.2:5060;transport=UDP
Here’s the PBX sending a 100 Trying to that new inbound call
2025/03/17 10:08:43.531952 10.0.xx.106:5060 → 10.0.xx.2:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.xx.2:5060;branch=z9hG4bK-ae05be1-f5fed-3c0eb932-7efb6c2dbee8;received=10.0.xx.2;rport=5060
From: "0792222"sip:79****2222@10.0.xx.2:5060;transport=UDP;tag=BN1884504541-1-1742206123-924972956
To: "792244"sip:79****2244@10.0.xx.106:5060;transport=UDP
Call-ID: 3a22e172-737d2013-f5fed-7efb42fe8000-6eb330a-13c9-764
CSeq: 1751 INVITE
Server: FPBX-17.0.19.24(20.10.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:****2244@10.0.xx.106:5060
Content-Length: 0
And here is the PBX sending the call back to the provider, again.
2025/03/17 10:08:43.597310 10.0.xx.106:5060 → 10.0.xx.2:5060
INVITE sip:****2244@10.0.xx.2 SIP/2.0
Via: SIP/2.0/UDP 10.0.xx.106:5060;branch=z9hG4bK499a3f36;rport
Max-Forwards: 70
From: sip:****2222@10.0.xx.2;tag=as1e215d42
To: sip:****2244@10.0.xx.2
Contact: sip:****2222@10.0.xx.106:5060
Call-ID: 00d58a0161c347420ab82fa3409a4698@10.0.xx.2
CSeq: 102 INVITE
User-Agent: FPBX-17.0.19.24(20.10.0)
Date: Mon, 17 Mar 2025 10:08:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 321
Rinse, repeat until 503 error is thrown.
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FreePBX 17 - PhoneBook
Hello jfinstrom. Thanks for your reply. I read this topic, but it’s not clear for me.
What I have to do so ? the speadking about freepbx.conf. Do you have any tutorial about this ? I don’t see anything in the web about this. I just want to know how can I connect a phone withe my “external contacts” directory. what is the link I have to use in my phone ?
Thanks for your help.
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Sangoma Connect module disabled because of vulnerability
This morning one of our customers called us because the Sangoma softphones were down.
The Sangoma Connect module was disabled because of a vulnerability. I just had to update it to fix the problem.
This is a major issue because this customer primarly use Sangoma Talk and all the phones were down for him. Is there a way to prevent this or to force an update as soon as the module is disabled when a vulnerability is found ?
I have changed the update sheduler for daily checks instead of weekly check to help a little.
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Sangoma Connect module disabled because of vulnerability
Good call. Never turn on auto updates on a production system!
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FreePBX Support - Incoming Call Failures [title edit by mod]
I am going a little crazy. Incoming calls are not getting through on VOIP Innovations because my server wants them to authenticate. I am waiting for FreePBX support who seemed mostly clueless. But it is worse I called them using the emergency PIN 4 hours later no callback. Where is the 1 hour SLA. Once I called them they will look into it and call me back.
On PJSIP settings I have username before IP (which I suspect might be the problem) but if I put IP before username my phones can’t authenticate so I’m not sure what you want me to do here.
Is there somebody who can provide professional grade support. I expected it for $1000 from Sangoma but they don’t seem to be living up to their end of the deal.
Sorry if I sound very frustrated. I am…
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FreePBX Support - Incoming Call Failures [title edit by mod]
This is a bit confusing. What does changing the VI trunk make your phones no longer register? they aren’t related.
As for the incoming calls, you need to have the VI IPs in the Match field of the trunk.
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FreePBX Support - Incoming Call Failures [title edit by mod]
Disable authentication on inbound for trunk.
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FreePBX Support - Incoming Call Failures [title edit by mod]
Tom,
I am talking about the setting in Advanced SIP Settings–PJSIP.
I have the VI IPs in the match filed. But I’m getting the following (according to VOIP Innovations).
Checking my logs for calls to your DID 7325551234, I found this most recent call:
Date/Time Source Destination SIP Release
2025-03-17 06:54:37.000 +972545551234 +17325551234 SIP Challenge Timeout 504 T
In this case, when we are sending these calls to you, your system is expecting us to send authentication information. The EPG that is routing calls to your IP, Company-Regular, is no tan authentication trunk nor is this feature currently active for your account. Because of this, a 504 SIP Challenge Timeout is being returned to us from your network.
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FreePBX Support - Incoming Call Failures [title edit by mod]
Already long disabled.
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FreePBX Support - Incoming Call Failures [title edit by mod]
That doesn’t mean an authentication issue. It means a response wasn’t received in a certain amount of time. Show these configuration settings you are using. Mask any passwords or sensitive info.
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