@JamesMcG wrote:

Hi all,
I am trying to do a simple test with 2 peer sip devices (Zoiper sip phones) with audio alaw and uIlaw codecs. I would like to set up the call with asterisk (freebpx) using pjsip_chan extensions and send the media peer 2 peer. I have this working with 2 chan_sip extensions but for the life of me I cannot get this working even though this, according to the manual, is the default state for pjsip extensions. I am not connected to any router and the call is connected but the audio goes through the asterisk pbx.

Has anyone got a pjsip.conf that I could reference that allows direct media.

I welcome any help on this.
James

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@rmatteson wrote:

I have a few users reporting that their calls are disconnecting (and one that seems to be having the issue more often than others).
I haven’t been able to make heads or tails of what is causing the disconnect.

Here’s the Asterisk log for a recent call.

https://pastebin.com/SjpXKZZq

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@lcn wrote:

Hello

I run FreePBX v14 on Raspberry and i set 2 trunks.

• 1 GSM line issued from a USB dongle
• 1 SIP trunk issued from a provider.

At the moment all incomming calls and redirected to Misc Destination to a mobile phone.
The dial pattern for the GSM Dongle is set as follow :
no prepend ; prefix = 0 ; match pattern = .

All works fine all incomming calls are properly redirected to my mobile phone.

Now, I would that some specific extensions ( 1 ; 4 ; 6 ) going to use a the use my second SIP trunk to place calls.

How could i set the second Outbound route to use that trunck ?
I tried to add the according extensions caller ID into Dial pattern of the SIP outbound route, but calls are still placed with the GSM outbound route.
I wonder if the wildcard . is not the way to proceed …

Many thanks for your help,

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@faisalkhan wrote:

Hi Guys,

I have been compromised.
I have a distro PBX 14 setup. I have twilio account setup for outbound calling.
Auto refill is enabled on the account.
I have noticed successful calls on switzerland High-Risk Toll Fraud destination.
when I want to check the CDRs and Log files in FreePBX, I can’t find any of the record for those dates like on 26th and 27th I have no record of any
of the log files: CDRS, FreepbxLogs, security logs, fail2ban or any of the logs for those dates.

Now I want to trace those call logs and check how system was breached and we were compromised.

Are there any suggestions to start from.

One Major concern is twilio has those high-risk Toll Fraud destinations by default blocked, then how calls passed to these destinations.

some of the numbers :

1. 41740888875
2. 41740888016
3. 41740888015
4. 41740888875

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@mroth wrote:

My FreePBX is running well but when some mobile phones are calling from external there is no audio. When I call with my mobile phone there is audio. I have no idea what’s the mistake…
Here is the log of the call with no audio:
Hope someone have an tip.
Oct 30 20:02:35] dialparties.agi: Starting New Dialparties.agi
[Oct 30 20:02:35] dialparties.agi: Caller ID name is ‘+4915232734567’ number is ‘+4915232734567’
[Oct 30 20:02:35] dialparties.agi: CW Ignore is:
[Oct 30 20:02:35] dialparties.agi: CF Ignore is:
[Oct 30 20:02:35] dialparties.agi: CW IN_USE/BUSY is: 1
[Oct 30 20:02:35] dialparties.agi: Methodology of ring is ‘ringall’
[Oct 30 20:02:35] – dialparties.agi: Added extension 32 to extension map
[Oct 30 20:02:35] – dialparties.agi: Added extension 33 to extension map
[Oct 30 20:02:35] – dialparties.agi: Extension 32 cf is disabled
[Oct 30 20:02:35] – dialparties.agi: Extension 33 cf is disabled
[Oct 30 20:02:35] – dialparties.agi: Extension 32 do not disturb is disabled
[Oct 30 20:02:35] – dialparties.agi: Extension 33 do not disturb is disabled
[Oct 30 20:02:35] > dialparties.agi: extnum 32 has: cw: 1; hascfb: 0 [] hascfu: 0 []
[Oct 30 20:02:35] – dialparties.agi: dbset CALLTRACE/32 to +4915232734567
[Oct 30 20:02:35] > dialparties.agi: extnum 33 has: cw: 1; hascfb: 0 [] hascfu: 0 []
[Oct 30 20:02:35] – dialparties.agi: dbset CALLTRACE/33 to +4915232734567
[Oct 30 20:02:35] – dialparties.agi: Filtered ARG3: 32-33
[Oct 30 20:02:35] > dialparties.agi: NODEST: 333 adding M(auto-blkvm) to dialopts: TtrM(auto-blkvm)
[Oct 30 20:02:35] > dialparties.agi: NODEST: 333 blkvm enabled macro already in dialopts: TtrM(auto-blkvm)
[Oct 30 20:02:35] – <PJSIP/Telekom_567890-00000091>AGI Script dialparties.agi completed, returning 0
[Oct 30 20:02:35] – Executing [s@macro-dial:9] NoOp(“PJSIP/Telekom_567890-00000091”, “Returned from dialparties with groups to dial”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:10] NoOp(“PJSIP/Telekom_567890-00000091”, "ringall array ") in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:11] Set(“PJSIP/Telekom_567890-00000091”, “__FMGL_DIAL=”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:12] Set(“PJSIP/Telekom_567890-00000091”, “LOOPCNT=2”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:13] Set(“PJSIP/Telekom_567890-00000091”, “ITER=1”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:14] Set(“PJSIP/Telekom_567890-00000091”, “__EXTTOCALL=32”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:15] NoOp(“PJSIP/Telekom_567890-00000091”, “Working with 32”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:16] ExecIf(“PJSIP/Telekom_567890-00000091”, “0?Set(HASH(__SIPHEADERS,Alert-Info)=Normal;volume=)”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:17] ExecIf(“PJSIP/Telekom_567890-00000091”, “0?Set(HASH(__SIPHEADERS,Alert-Info)=Normal;volume=)”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:18] Set(“PJSIP/Telekom_567890-00000091”, “ITER=2”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:19] GotoIf(“PJSIP/Telekom_567890-00000091”, “1?ndloopbegin”) in new stack
[Oct 30 20:02:35] – Goto (macro-dial,s,14)
[Oct 30 20:02:35] – Executing [s@macro-dial:14] Set(“PJSIP/Telekom_567890-00000091”, “__EXTTOCALL=33”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:15] NoOp(“PJSIP/Telekom_567890-00000091”, “Working with 33”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:16] ExecIf(“PJSIP/Telekom_567890-00000091”, “0?Set(HASH(__SIPHEADERS,Alert-Info)=Normal;volume=)”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:17] ExecIf(“PJSIP/Telekom_567890-00000091”, “0?Set(HASH(__SIPHEADERS,Alert-Info)=Normal;volume=)”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:18] Set(“PJSIP/Telekom_567890-00000091”, “ITER=3”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:19] GotoIf(“PJSIP/Telekom_567890-00000091”, “0?ndloopbegin”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:20] Macro(“PJSIP/Telekom_567890-00000091”, “dial-ringall-predial-hook,”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial-ringall-predial-hook:1] MacroExit(“PJSIP/Telekom_567890-00000091”, “”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:21] ExecIf(“PJSIP/Telekom_567890-00000091”, “0?Set(ds=SIP/32&SIP/33,20,trM(auto-blkvm)g)”) in new stack
[Oct 30 20:02:35] – Executing [s@macro-dial:22] Dial(“PJSIP/Telekom_567890-00000091”, “SIP/32&SIP/33,20,trM(auto-blkvm)b(func-apply-sipheaders^s^1),”) in new stack
[Oct 30 20:02:35] == Using SIP VIDEO TOS bits 136
[Oct 30 20:02:35] == Using SIP VIDEO CoS mark 6
[Oct 30 20:02:35] == Using SIP RTP TOS bits 184
[Oct 30 20:02:35] == Using SIP RTP CoS mark 5
[Oct 30 20:02:35] == Using SIP VIDEO TOS bits 136
[Oct 30 20:02:35] == Using SIP VIDEO CoS mark 6
[Oct 30 20:02:35] == Using SIP RTP TOS bits 184
[Oct 30 20:02:35] == Using SIP RTP CoS mark 5
[Oct 30 20:02:35] – SIP/32-000000e2 Internal Gosub(func-apply-sipheaders,s,1) start
[Oct 30 20:02:35] – Executing [s@func-apply-sipheaders:1] NoOp(“SIP/32-000000e2”, “Applying SIP Headers to channel SIP/32-000000e2”) in new stack
[Oct 30 20:02:35] – Executing [s@func-apply-sipheaders:2] Set(“SIP/32-000000e2”, “TECH=SIP”) in new stack
[Oct 30 20:02:35] – Executing [s@func-apply-sipheaders:3] Set(“SIP/32-000000e2”, “SIPHEADERKEYS=”) in new stack
[Oct 30 20:02:35] – Executing [s@func-apply-sipheaders:4] While(“SIP/32-000000e2”, “0”) in new stack
[Oct 30 20:02:35] – Jumping to priority 11
[Oct 30 20:02:35] – Executing [s@func-apply-sipheaders:12] Return(“SIP/32-000000e2”, “”) in new stack
[Oct 30 20:02:35] == Spawn extension (from-internal, 333, 1) exited non-zero on ‘SIP/32-000000e2’
[Oct 30 20:02:35] – SIP/32-000000e2 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
[Oct 30 20:02:35] – SIP/33-000000e3 Internal Gosub(func-apply-sipheaders,s,1) start
[Oct 30 20:02:35] – Executing [s@func-apply-sipheaders:1] NoOp(“SIP/33-000000e3”, “Applying SIP Headers to channel SIP/33-000000e3”) in new stack
[Oct 30 20:02:35] – Executing [s@func-apply-sipheaders:2] Set(“SIP/33-000000e3”, “TECH=SIP”) in new stack
[Oct 30 20:02:35] – Executing [s@func-apply-sipheaders:3] Set(“SIP/33-000000e3”, “SIPHEADERKEYS=”) in new stack
[Oct 30 20:02:35] – Executing [s@func-apply-sipheaders:4] While(“SIP/33-000000e3”, “0”) in new stack
[Oct 30 20:02:35] – Jumping to priority 11
[Oct 30 20:02:35] – Executing [s@func-apply-sipheaders:12] Return(“SIP/33-000000e3”, “”) in new stack
[Oct 30 20:02:35] == Spawn extension (from-internal, 333, 1) exited non-zero on ‘SIP/33-000000e3’
[Oct 30 20:02:35] – SIP/33-000000e3 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
[Oct 30 20:02:35] – Called SIP/32
[Oct 30 20:02:35] – Called SIP/33
[Oct 30 20:02:35] – SIP/33-000000e3 connected line has changed. Saving it until answer for PJSIP/Telekom_567890-00000091
[Oct 30 20:02:35] – SIP/32-000000e2 connected line has changed. Saving it until answer for PJSIP/Telekom_567890-00000091
[Oct 30 20:02:37] – SIP/32-000000e2 is ringing
[Oct 30 20:02:39] – SIP/33-000000e3 is ringing
[2019-10-30 20:02:39] NOTICE[1758]: chan_sip.c:24652 handle_response_peerpoke: Peer ‘32’ is now Lagged. (2568ms / 2000ms)
[2019-10-30 20:02:39] NOTICE[1758]: chan_sip.c:24652 handle_response_peerpoke: Peer ‘33’ is now Lagged. (2118ms / 2000ms)
[Oct 30 20:02:39] > 0x6fbb0ce8 – Strict RTP learning after remote address set to: 192.168.178.99:5000
[Oct 30 20:02:39] – SIP/32-000000e2 connected line has changed. Saving it until answer for PJSIP/Telekom_567890-00000091
[Oct 30 20:02:39] – SIP/32-000000e2 answered PJSIP/Telekom_567890-00000091
[Oct 30 20:02:39] – Executing [s@macro-auto-blkvm:1] Set(“SIP/32-000000e2”, “__MACRO_RESULT=”) in new stack
[Oct 30 20:02:39] – Executing [s@macro-auto-blkvm:2] Set(“SIP/32-000000e2”, “CFIGNORE=”) in new stack
[Oct 30 20:02:39] – Executing [s@macro-auto-blkvm:3] Set(“SIP/32-000000e2”, “MASTER_CHANNEL(CFIGNORE)=”) in new stack
[Oct 30 20:02:39] – Executing [s@macro-auto-blkvm:4] Set(“SIP/32-000000e2”, “FORWARD_CONTEXT=from-internal”) in new stack
[Oct 30 20:02:39] – Executing [s@macro-auto-blkvm:5] Set(“SIP/32-000000e2”, “MASTER_CHANNEL(FORWARD_CONTEXT)=from-internal”) in new stack
[Oct 30 20:02:39] – Executing [s@macro-auto-blkvm:6] Macro(“SIP/32-000000e2”, “blkvm-clr,”) in new stack
[Oct 30 20:02:39] – Executing [s@macro-blkvm-clr:1] Set(“SIP/32-000000e2”, “SHARED(BLKVM,PJSIP/Telekom_567890-00000091)=”) in new stack
[Oct 30 20:02:39] – Executing [s@macro-blkvm-clr:2] Set(“SIP/32-000000e2”, “GOSUB_RETVAL=”) in new stack
[Oct 30 20:02:39] – Executing [s@macro-blkvm-clr:3] MacroExit(“SIP/32-000000e2”, “”) in new stack
[Oct 30 20:02:39] – Executing [s@macro-auto-blkvm:7] ExecIf(“SIP/32-000000e2”, “0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=32)”) in new stack
[Oct 30 20:02:39] – Executing [s@macro-auto-blkvm:8] ExecIf(“SIP/32-000000e2”, “0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Telefon Jessica)”) in new stack
[Oct 30 20:02:39] – Channel SIP/32-000000e2 joined ‘simple_bridge’ basic-bridge <78c31943-fe10-4e77-bc2e-afc4c6a485b0>
[Oct 30 20:02:39] > 0x6fbb0ce8 – Strict RTP switching to RTP target address 192.168.178.99:5000 as source
[Oct 30 20:02:39] – Channel PJSIP/Telekom_567890-00000091 joined ‘simple_bridge’ basic-bridge <78c31943-fe10-4e77-bc2e-afc4c6a485b0>
[Oct 30 20:02:39] > 0x6fc1c2e8 – Strict RTP switching to RTP target address 192.168.31.1:7074 as source
[Oct 30 20:02:40] > 0x6fc1c2e8 – Strict RTP learning complete - Locking on source address 192.168.31.1:7074
[Oct 30 20:02:44] > 0x6fbb0ce8 – Strict RTP learning complete - Locking on source address 192.168.178.99:5000
[2019-10-30 20:02:49] NOTICE[1758]: chan_sip.c:24652 handle_response_peerpoke: Peer ‘32’ is now Reachable. (35ms / 2000ms)
[2019-10-30 20:02:49] NOTICE[1758]: chan_sip.c:24652 handle_response_peerpoke: Peer ‘33’ is now Reachable. (35ms / 2000ms)
[Oct 30 20:02:50] – Channel SIP/32-000000e2 left ‘simple_bridge’ basic-bridge <78c31943-fe10-4e77-bc2e-afc4c6a485b0>
[Oct 30 20:02:50] – Channel PJSIP/Telekom_567890-00000091 left ‘simple_bridge’ basic-bridge <78c31943-fe10-4e77-bc2e-afc4c6a485b0>
[Oct 30 20:02:50] == Spawn extension (macro-dial, s, 22) exited non-zero on ‘PJSIP/Telekom_567890-00000091’ in macro ‘dial’
[Oct 30 20:02:50] == Spawn extension (ext-group, 333, 14) exited non-zero on ‘PJSIP/Telekom_567890-00000091’
[Oct 30 20:02:50] – Executing [h@ext-group:1] Macro(“PJSIP/Telekom_567890-00000091”, “hangupcall,”) in new stack
[Oct 30 20:02:50] – Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/Telekom_567890-00000091”, “1?theend”) in new stack
[Oct 30 20:02:50] – Goto (macro-hangupcall,s,3)

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@munozj wrote:

I’m working with a company that has about 1000 polycom’s with a mix of hardware versions. We are needing to upgrade the firmware but not all the models will accept the upgraded firmware? I was able to see that when they provision against freepbx over http, the current firmware version is included in the log.

HTTP/1.1" 200 42463 “-” “FileTransport PolycomVVX-VVX_400-UA/5.5.1.15880 Type/Application”
x.x.x.x - - [30/Oct/2019:20:47:46 +0000] "GET /0004f2xxxxxx-5326.cfg

is there any easy way to be able to find out the hardware version? It doesn’t even show up on the web gui of the devices in the current firmware.

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@GeekBoy wrote:

I see this has been a common issue for a while now, but with version 14, it seems the UCP was changed.

Version 13, there was an enable account in User Management, but on this new version I am not seeing that. I went on and changed the user password, and tried to log in, but all I get is:

Invalid Login Credentials

I don’t see anything on this in the documentation

Also, the “Submit & Send Email to User” does not send any email out. I think even for FreePBX 13, I never saw this function.

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@eman43078 wrote:

Hello All. I have a hosted freepbx server (FreePBX 15.0.16.22) Asterisk 13.27.1 built by mockbuild(from command line) and I’m trying to configure the baseline file in EPM for polycom to include a STUN server. Polycom documentation states that this setting is found in the firewall-nat.cfg but freebpx does not show that as a one of the files to configure. A little guidance would be appreciated.

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@daxm wrote:

I’m not sure how to troubleshoot this but my automatic nightly backups that send to an FTP server have failed since upgrading to 15. Any suggestions on how I can troubleshoot this?

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@asclafit wrote:

Hi Everyone and thanks for your help!

I’ve a Freepbx installed with asterisk in our office. Everything works great but when we have 3 income calls or more it’s impossible to talk with a client and we need to forcely to hangup one line to talk properly…
Another problem we have it’s randomly Freepbx hangup up all the lines and when you tries to call the server answer “All circuits are busy rigth now, call again later”.

SIPconfiguration: Out:
type=peer
secret=XXXXX
port=5095
outboundproxy=XXXXXXX:5095
Nombre=Vodafone
nat=yes
insecure=port,invite
host=ims.vodafone.es
fromuser=+34XXXXXX
fromdomain=ims.vodafone.es:5095
dtmfmode=auto
disallow=all
auth=XXXXXX@ims.vodafone.es
allow=alaw,ulaw,gsm

Incoming:
username=+34XXXXXXX
type=peer
secret=XXXXX
qualify=no
port=5095
outboundproxyport=5095
outboundproxy=XXXXXXXX:5095
nat=yes
insecure=port,invite
host=ims.vodafone.es
fromdomain=ims.vodafone.es:5095
dtmfmode=auto
disallow=all
context=from-trunk
canreinvite=no
allow=alaw,ulaw,gsm

I will apreciate very your help, we have a very big problem with this 2 issues and I can’t find a solution…

Many Thanks!!!

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@andy_woolford wrote:

I am using FreePBX 15.0.16 (distro).
Today I am receiving warnings in the Module Admin, when trying to Check Online. Dashboard tells me there are upgrades waiting. However I get:

Warning: Cannot connect to online repository(s) (http://mirror1.freepbx.org,http://mirror2.freepbx.org). Online modules are not available

If I wget http://mirror1.freepbx.org, then it will download an index.htm file.
If I wget http://mirror2.freepbx.org it cannot find the server.

Using the CLI:

~# fwconsole ma upgradeall
No repos specified, using: [standard,commercial,unsupported,extended] from last GUI settings

Module(s) requiring upgrades: backup, core, filestore, framework, sysadmin, zulu
Upgrading module 'framework' from 15.0.16 to 15.0.16.18
Downloading module 'framework'
Processing framework
Verifying local module download...Redownloading
The following error(s) occured:
 - File Integrity failed for /var/www/html/admin/modules/_cache/framework-15.0.16.18.tgz.gpg - aborting (sha1 did not match)

Also:

~# ping 162.253.134.144
PING 162.253.134.144 (162.253.134.144) 56(84) bytes of data.
^C
--- 162.253.134.144 ping statistics ---
8 packets transmitted, 0 received, 100% packet loss, time 7000ms

~# ping 199.102.239.170
PING 199.102.239.170 (199.102.239.170) 56(84) bytes of data.
64 bytes from 199.102.239.170: icmp_seq=1 ttl=49 time=90.6 ms
64 bytes from 199.102.239.170: icmp_seq=2 ttl=49 time=90.3 ms
64 bytes from 199.102.239.170: icmp_seq=3 ttl=49 time=90.2 ms
^C
--- 199.102.239.170 ping statistics ---
3 packets transmitted, 3 received, 0% packet loss, time 2001ms
rtt min/avg/max/mdev = 90.281/90.446/90.677/0.297 ms

I note the following entries in the iptables -L: (amongst many others redacted)

Chain fpbxhosts (1 references)
target     prot opt source               destination
zone-trusted  all  --  199.102.239.170      anywhere
zone-trusted  all  --  static.162.253.134.144.cyberlynk.net  anywhere

I am not familliar with the syntax used in the static entry, which lists the IP addess that fails, but this might be a red herring since the first IP address does work. Clearly this is not a DNS issue because I’m pinging the resolved IP addresses, not the URL.

Coincidentally (or not?) I have a separate notice which cautions me that my disk space is low:

Storage space is getting high on the following drives of your system:

/dev/mapper/SangomaVG-root is 75% full

However, according to the dashboard there is 25% left and I very much doubt the files are that big.

I would be grateful for any help in diagnosing this issue.

Kind regards

Andy

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@dailas wrote:

Hi there. Some time ago i saw that Original Mailbox Field is no more shown in FreePBX 14 Distro. Now i got to know that i can’t find it in 13 also. Was it completely removed or any chance to turn it on back again? The txt file still has origmailbox= inside for both versions. Is there any chance to see it via UCP GUI?

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@car2ner wrote:

After updating FreePBX to version 14.0.13.6, I began to observe the following warnings in the console during an outgoing call through the SIP trunk:

Executing [s@macro-outbound-callerid:23] ExecIf(“SIP/1234-000008b8”, “1?Set(CALLERID(all)=4952222222)”) in new stack
[2019-10-25 09:53:24] WARNING[5658][C-000003e8]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected ‘=’, expecting $end; Input:
= 1 & 10 = 0
^

At the same time, calls are going fine, but still I would like to get rid of these warnings in the console.

Tell me how to fix it?

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@6taylor wrote:

Hello. I have a little issue with Asterisk 13 con CentOS 7. See, I have configured my TRUNK and OUTBOUND ROUTES, and cannot make calls through the trunk to the other end and viceversa.

I leave with configuration from both ends to please help me out if I’m missing something.

Images from TRUNK 1

1.jpg1367×724 154 KB

2.jpg1335×648 172 KB

3.jpg1056×503 73.2 KB

4.jpg1329×495 72.5 KB

Images from OUTBOUBND ROUTES.

5.jpg1370×663 127 KB

6.jpg1297×485 74.7 KB

On the other end IP address is different and of course, IP from TRUNK 2 is on my peer details from TRUNK 1, and IP from TRUNK 1 is on peer details from TRUNK 2.

On every fields number “8” changes for a number “5” and viceversa on the TRUNK 2.

When Going to CLI, I get this command to check my trunk is conencted to another end, but cannot make calls.

freepbx*CLI> iax2 show peers
Name/Username Host Mask Port Status Description
TO-MAQUINARIAS/ 172.16.34.20 (S) 255.255.255.255 4569 (T) OK (1 ms)
1 iax2 peers [1 online, 0 offline, 0 unmonitored]

Don’t know what else to do. Please help.

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@sentinelace wrote:

I am using the bria mobile app. I need my desk phone and app to ring at the same time since it’s the same extension. I have max contacts set to 5 but only my app rings. Am I missing something?

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@jerryriggin wrote:

Running module updates on 10.13.66-22 from CLI I just got

[Doctrine\DBAL\DBALException]
Unknown database type bit requested, Doctrine\DBAL\Platforms\MySqlPlatform may not support it.

Here is the error in the GUI when I try to update Sound Languages 13.0.26 to 13.0.27:

Doctrine \ DBAL \ DBALException
HELP
Unknown database type bit requested, Doctrine\DBAL\Platforms\MySqlPlatform may not support it.

/var/www/html/admin/libraries/Composer/vendor/doctrine/dbal/lib/Doctrine/DBAL/Platforms/AbstractPlatform.php
if ($this->doctrineTypeMapping === null) {
$this->initializeAllDoctrineTypeMappings();
}

    $dbType = strtolower($dbType);

    if (!isset($this->doctrineTypeMapping[$dbType])) {
        throw new \Doctrine\DBAL\DBALException("Unknown database type ".$dbType." requested, " . get_class($this) . " may not support it.");
    }

I get this on pageingpro, endpoint, soundlang and a number of other modules. I don’t see bit data types in any of those tables. What could cause this error?

I can’t do a reload now because " Unable to locate the FreePBX BMO Class 'Soundlang’A required module might be disabled or uninstalled. " which it is "Disabled; Pending Upgrade to 13.0.27 "

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@bnewton wrote:

I have a PBXact installation where I mounted a USB stick to store periodic backups. It was working okay for a while, but recently I checked on it and the stick is no longer mounted and will not mount. It doesn’t error when trying to mount, just fails silently, like something is immediately unmounting it.

I was mouting on /var/spool/asterisk/backup/USB_Backup and writing a weekly backup.

This fails:
mount -o uid=995,gid=995 /dev/sdb1 /var/spool/asterisk/backup/USB_Backup

Also had this in fstab which also no longer works:
UUID=USB20FD /var/spool/asterisk/backup/USB_Backup auto nofail,uid=995 0 0

Is there some change in a recently release that unmounts things? Or maybe it no longer likes fat?

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@mkleine wrote:

Greetings, I’ve got a new FreePBX 14 installation with a Grandstream HT813 FXS/FXO SIP Server connected to a POTS line. Incoming and outgoing calls are working, but they hangup after 30 seconds. This appears to be the failing segment of the log. 172.31.5.164 is the HT813 SIP Server. The FreePBX server is 172.31.4.3 on a /22 (255.255.252.0) network.

Can anyone shed any light on this subject as to why the “RTCP from 172.31.5.164:5015: Failed first packet validity check” is happening?

Thanks, Mark

[2019-10-31 22:04:21] DEBUG[3421] res_pjsip/pjsip_distributor.c: Searching for serializer associated with dialog dlg0x7fb3741213b8 for Response msg 200/BYE/cseq=21026 (rdata0x7fb3ac12a9c8)
[2019-10-31 22:04:21] DEBUG[3421] res_pjsip/pjsip_distributor.c: Found serializer pjsip/distributor-0000003b associated with dialog dlg0x7fb3741213b8
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: Source of transaction state change is RX_MSG
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: Received response
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: Response is 200 OK
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: Received response
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: Response is 200 OK
[2019-10-31 22:04:21] DEBUG[3059] res_pjsip_session.c: BYE received final response code 200
[2019-10-31 22:04:21] DEBUG[28227][C-0000005c] res_rtp_asterisk.c: Got RTCP report of 12 bytes from 172.31.5.164:5015
[2019-10-31 22:04:21] DEBUG[28227][C-0000005c] res_rtp_asterisk.c: 0x7fb3742a2de0 – RTCP from 172.31.5.164:5015: Failed first packet validity check
[2019-10-31 22:04:21] DEBUG[19332] manager.c: Examining AMI event:
Event: HangupRequest
Privilege: call,all
Channel: PJSIP/6000-0000015a
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 14055551212
CallerIDName: NAME HERE
ConnectedLineNum:
ConnectedLineName:
Language: en
AccountCode:
Context: app-pbdirectory
Exten: pbdirectory
Priority: 4
Uniqueid: 9999999999.999
Linkedid: 9999999999.999
Cause: 18

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@Altocloud1 wrote:

I’m having a very odd issue and hope someone can help out! I’m trying to setup a Twilio trunk and when making an outbound call there is %40 and text between my phone number and @ pstn.twilio.com. I have never seen this before. Any thoughts on where I can look?

To:sip:+1XXXXXXXXXX%40Trunk_Outgoing_Twilio@XXXXX.pstn.twilio.com;tag=53545548_6772d868_acaa008f-e493-4126-a447-53b246f78d65
Via: SIP/2.0/UDP X.X.X.X:5060;received=X.X.X.X;branch=z9hG4bK59006338;rport=5060
Server: Twilio
X-Twilio-Error: 32101 The called number is not correctly formatted.
Content-Length: 0

Here’s my trunk config:

trunkconfig1.png969×512 30.2 KB

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@haris013 wrote:

Hello! Newbie here. It is my first time setting up a Freepbx server so i have a few questions.

I have setted up a trunk sip from my provider, created an extension, then an inbound route that leads to the extension and an outbound route for dialling out.

I have only 1 number from my provider and at the moment i don’t need any other internal phones.

I am using a softphone application (zoiper) at my smartphone in order to receive and dial phone calls. While I am outside my office, i am using VPN in order to still receive phone calls.

When my GSM signal is low or my phone runs out of battery, is there any way to track down missed calls while my softphone app is unregistered? How do you recomend me to handle the missing calls when there is no end point(softphone or any IP phone) connected at the freepbx server?

thanks!

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