Clean install of SNG7-PBX-64bit-1805-1 - when I try to upload a 4-5MB MP3 to music on hold, I get told:
File size is too large. Max size is 2097152 bytes
On previous distros I haven’t had this? I have run all updates and it persists.
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New install of SNG7-PBX-64bit-1805-1, file size is too large uploading MoH?
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PAI and CID on 'common' trunks
Do I understand correctly that you want the outbound caller ID in the From header and the account number in P-Asserted-Identity?
If so, configure the trunk so it doesn’t set PAI at all (From will contain the caller ID propagated from the extension). Then, use some custom dialplan to sent PAI:
On a chan_sip trunk, this is easy. In extensions_custom.conf, put some code like:
[macro-dialout-trunk-predial-hook]
exten => s,1,SipAddHeader(P-Asserted-Identity: <12345678>)
exten => s,n,MacroExit()
where 12345678 is the account number. This will of course be more complex if you have multiple trunks with different account numbers and/or trunks from other providers where you don’t want to monkey with the header.
I’ve not done this with pjsip, but there are some gotchas. See @maple solution in Add custom SIP header to PJSIP trunk , which I hope will work for you.
I’d not heard of Spark Voice Connect before and looked at their website. Alas, I found only marketing speak. If you want to try another trunking provider (as a workaround for the present problem, as a backup provider, or perhaps are your main provider if rates and/or quality prove superior to Spark), take a look at AnveoDirect and Voxbeam. Both provide a small credit at signup so you can test without making a payment or providing any financial information. Both allow sending an arbitrary caller ID and don’t require any special headers.
Prices below are in US dollars.
AnveoDirect’s NZ rates on their ‘Prime’ route are $0.00835/min. (landline) and $0.02662/min. (mobile). Voxbeam’s ‘Platinum’ route is $0.01 (Auckland), $0.0116 (other landline), $0.034 (mobile).
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Failure reading sector 0x0 from hd2 after fresh install
Hi,
On a newly bought intel i5 with 16GB and 2 WD RED HDD, i installed the latest freepbx distro.
After the installation I got a message
failure reading sector 0x0 from hd2
press any key to continue
Any suggestions?
Lazaros
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What causing so much security events that fill fail2ban.log
Fresh install of FreePBX 14 distro with around 300 endpoints. All is working fine but I noticed an huge amount of security events that seems coming from the Asterisk Manager. There is around 20 security events per seconds that fill the fail2ban log file. Around 1Gb at the end of the day.
FreePBX: 14.0.3.13
Asterisk: 13.22.0
Here what we see in the /var/log/asterisk/fail2ban
**[2018-10-02 05:02:02] SECURITY[2177] res_security_log.c: SecurityEvent="SuccessfulAuth",EventTV="2018-10-02T05:02:02.312-0400",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="386",SessionID="613daa02-19bf779d-d85ef138@192.18.100.174",LocalAddress="IPV4/UDP/192.18.100.2/5060",RemoteAddress="IPV4/UDP/192.18.100.174/5060",UsingPassword="1"**
**[2018-10-02 05:02:02] SECURITY[2177] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="2018-10-02T05:02:02.329-0400",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="106",SessionID="17db7bdf-e56725ea-53229fc9@192.18.100.194",LocalAddress="IPV4/UDP/192.18.100.2/5060",RemoteAddress="IPV4/UDP/192.18.100.194/5060",Challenge=""**
**[2018-10-02 05:02:02] SECURITY[2177] res_security_log.c: SecurityEvent="SuccessfulAuth",EventTV="2018-10-02T05:02:02.348-0400",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="106",SessionID="17db7bdf-e56725ea-53229fc9@192.18.100.194",LocalAddress="IPV4/UDP/192.18.100.2/5060",RemoteAddress="IPV4/UDP/192.18.100.194/5060",UsingPassword="1"**
**[2018-10-02 05:02:02] SECURITY[2177] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="2018-10-02T05:02:02.370-0400",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="309",SessionID="a46c9148-5e10bd4b-4f9b4492@192.18.100.133",LocalAddress="IPV4/UDP/192.18.100.2/5060",RemoteAddress="IPV4/UDP/192.18.100.133/5060",Challenge=""**
And here a sample of the traffic that is going on (tcpdump -i lo -A -s0 port 5038)
05:14:28.039964 IP localhost.53628 > localhost.5038: Flags [.], ack 74494, win 6146, options [nop,nop,TS val 1557381226 ecr 1557381226], length 0
E..4t.@.@............|..@y.k...<.....(.....
\..j\..j
05:14:28.118549 IP localhost.5038 > localhost.53628: Flags [P.], seq 74494:74808, ack 1, win 342, options [nop,nop,TS val 1557381304 ecr 1557381226], length 314
E..nkf@.@..!...........|...<@y.k...V.b.....
\...\..jEvent: ChallengeSent
Privilege: security,all
EventTV: 2018-10-02T05:14:28.118-0400
Severity: Informational
Service: PJSIP
EventVersion: 1
AccountID: 248
SessionID: 654440dd-d829c004-35872147@192.18.100.81
LocalAddress: IPV4/UDP/192.18.100.2/5060
RemoteAddress: IPV4/UDP/192.18.100.81/5060
Challenge:
05:14:28.118574 IP localhost.53628 > localhost.5038: Flags [.], ack 74808, win 6146, options [nop,nop,TS val 1557381304 ecr 1557381304], length 0
E..4t/@.@............|..@y.k...v.....(.....
\...\...
05:14:28.135823 IP localhost.5038 > localhost.53628: Flags [P.], seq 74808:75128, ack 1, win 342, options [nop,nop,TS val 1557381322 ecr 1557381304], length 320
E..tkg@.@..............|...v@y.k...V.h.....
\...\...Event: SuccessfulAuth
Privilege: security,all
EventTV: 2018-10-02T05:14:28.135-0400
Severity: Informational
Service: PJSIP
EventVersion: 1
AccountID: 248
SessionID: 654440dd-d829c004-35872147@192.18.100.81
LocalAddress: IPV4/UDP/192.18.100.2/5060
RemoteAddress: IPV4/UDP/192.18.100.81/5060
UsingPassword: 1↧
Registration of the pjsip trunk to Sip Server german Telekom rejected
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Zulu plugin dial prefix
Our outbound pattern is 0. - it is required to dial a 0 prefix for outbound calls.
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PAI and CID on 'common' trunks
If this is a Chan_SIP trunk then you do
sendrpid=pai
If this is a PJSIP trunk then you go under the Advanced tab of the trunk settings and set the CallerID options to PAI.
Those setinngs will use PAI for the CallerID
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New install of SNG7-PBX-64bit-1805-1, file size is too large uploading MoH?
Sounds like the PHP upload limit is set to that. Needs to be adjusted. Is this a Distro install?
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Would it be possible to change PABX settings via CLI or changing files manually?
Asterisk by itself, sure. Bit donr the right way, it is fine.
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Scheduled Module Updates Not Occurring
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New install of SNG7-PBX-64bit-1805-1, file size is too large uploading MoH?
Yes, it is. I’ve changed the upload limit - it seems odd this would be set at 2MB by default on a distro install.
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Would it be possible to change PABX settings via CLI or changing files manually?
Hello @dicko I understand you. Thank you very much for your reply.
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Would it be possible to change PABX settings via CLI or changing files manually?
Hello BlazeStudios, Thank you very much for your reply.
Would you like to share your advice for this case?
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Please help - updated modules and now unable to access freepbx GUI
I was told the issue was a few broken modules that needed to be removed and reinstalled.
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Names not updating on phones
We have a client with over 60 extensions using Polycom VVX300’s, a few VVX500’s and 2 Yealink T46’s.
The Polycom Phones configure via TFTP, while the Yealinks configure via HTTP.
We are using FreePBX 14.0.3.19
We had 5 name changes on the VVX300’s
When the rest of the polycom phones dial an extension that had a name change, the old name shows as caller ID as opposed to the new name.
The yealink phones display properly, as does a VVX500 phone I have set up remotely in our office (which configures via FTP).
I had a support technician tell me that all these phones will need to be factory reset and reconfigured.
I tested this on one phone, and this indeed worked for the reset extension which now properly displays the correct name.
My concern is that I will have this issue again when a new employee is hired and the name changes on their extension.
Is this a bug or am I missing something?
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V14 Automatic Updates Not Occurring - Modules or System
I have multiple systems that have system and module updates sent to automatic and none of them are happening. Are there any logfiles I can check to see why the updates aren’t running?
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Would it be possible to change PABX settings via CLI or changing files manually?
@dicko would you hold the same opinion if you used a docker container for each client, if the orchestration (creation and deletion of containers) was automated?
I thought about this possibility, but I came across the following big question:
How to route SIP and RTP traffic to different containers using only a single external IP, securely and steadily?
It would be something like a SIP Proxy or SIP Traffic Router
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V14 Automatic Updates Not Occurring - Modules or System
I found a post by Lorne giving the update logs of the system. It appeared to be choking on a broken module:
[2018-Oct-02 08:49:00] [CRITICAL] (BMO/Notifications.class.php:507) - [NOTIFICATION]-[freepbx]-[modules_broken] - You have a broken module (The following modules are disabled because they are broken:
areminder
With a broken module it seems that all updates stop.
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Transferring calls with the BLF keys proper method?
What is the best method for a receptionist to transfer calls using the BLF keys? They want to let the end user know they have a call. When they put the caller on hold, then push their BLF key, it transfers the calls. We currently have them pick up a line and then hit the BLF to call. Is there an easier way? We want the BLF keys so we can see who is on the phone
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Aastra phones beeping
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