@smokey7722 wrote:

I've been using FreePBX now with no real issues for >1 year with a Cisco 9971. The only changes have been updates to keeping the system patched. Yesterday I had to reboot the system for the first time in months and while incoming calls still work, outgoing now fails with the following error:

[2017-07-16 08:26:49] WARNING[859][C-0000000a]: chan_sip.c:25808 handle_request_invite: Received SIP INVITE with unsupported required extension: required: unsupported:sdp-anat

I did find this topic https://community.freepbx.org/t/invite-with-unsupported-required-extension/20614 however I don't see any Allow fields in the extension (at least in the web UI) to modify. I'm hoping I am missing something obvious but I figured it would be good to make a post as others are much better than I am at this and may be able to direct me faster.

At the moment the server is fully up to date and indicating it is FreePBX 13.0.192.9.

Thanks!

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@subs wrote:

I have got some older SIP extensions (running chan_sip) which have recently stopped working. Really struggling to get to the bottom of this - partly because I have no idea what logs or where to look.

I was just going to not bother and upgrade the phones, but I have one modified ATA which integrates a doorbell to the phone system and that has stopped working too.

Can anyone suggest what log files I need to check and/or if there have been any changes recently which might have cause this?

UPDATED info
I have managed to get one of the devices working by doing a restore from a backup. Great I thought, now I can bring up the config screens and do a line-by-line check of the config and get the other phone (which I was adding to the network) to also work.

I checked line-by-line
- The chan-SIP config files only changing the 'secret' and the extension number
- The config screens on the devices again carefully ensuring only the secret and the extension numbers were different
Still no luck (and yes I did save and apply the updates)

So, I thought I'd make sure the new device was correctly configured by logging it onto the working extension. Unplugged the working extension and changed the password and secret to match on the 'new' additional handset. Again, no success.

Now the obvious conclusion is that the phone is faulty - EXCEPT - I set it up first go on a 3CX pbx immediately when I could not get freePBX to work. Registers fine ... I went back and triple checked again - line-by-line. Still no joy.

So now I am really confused. The log files (which I have now found) are only marginally helpful. I have to go out but I will post a further update on my return.

Second update
I have now returned from my meeting. I wondered as I traveled if the issue was with the IP address being blocked, so I have now changed the IP address using the DHCP (fixed) allocation tied to the MAC using an IP which has never been used to my knowledge and STILL no success.

I have checked the various log files ... file2ban, freepbx_security.log, freepbx.log and full. When I try rebooting the phone, nothing shows up with even the correct timestamp let alone anything related to this extension.

Really really puzzled.

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@255 wrote:

It seems that the RMS service is down. Cannot configure it and want to disable it. I have uninstalled it from my PBX and now it thinks that the servers are down. The configuration portal is broken

https://cloud.sangoma-rms.com/

and is sending me a email every minute....

Any ides?

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@cursor wrote:

I have a brand new installation of Freepbx 13.0.120.3 fully updated. I works but I keep getting segfaults at random intervals. I see the following message in /var/log/messages:

Jul 17 18:29:45 pbxbogota kernel: asterisk[4470]: segfault at 188 ip 00007ff73e12d23f sp 00007ff6c2d4e5c0 error 4 in libasteriskpj.so.2[7ff73e0bb000+180000]
Jul 17 18:31:47 pbxbogota kernel: asterisk[5564]: segfault at 188 ip 00007febaf75a23f sp 00007feb344355c0 error 4 in libasteriskpj.so.2[7febaf6e8000+180000]
Jul 17 18:32:49 pbxbogota kernel: asterisk[5677]: segfault at 188 ip 00007f2cc462f23f sp 00007f2c493b75c0 error 4 in libasteriskpj.so.2[7f2cc45bd000+180000]
Jul 17 18:40:36 pbxbogota kernel: asterisk[5840]: segfault at 188 ip 00007fc4531f323f sp 00007fc3cbf7d5c0 error 4 in libasteriskpj.so.2[7fc453181000+180000]

The closest issue I could find was that the digium module to provision phones was causing a crash and they recommended to remove the asterisk package via yum. I simply ran the "asterisk-version-switch" command and selected version 13 again. That downloaded all asterisk 13 packages again and erased the res_digiumphone package. Sadly this did not fix the problem as I am still getting segfaults.

The Freepbx distro is running on a virtual machine. Any recommendations on where to look for the problem?

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@alexm2017 wrote:

Dear all,

I have installed the last stable version of Freepbx 64bit.
The whole setup is working fine but i have an issue.

I have tested bot 3cx and xlite for this scenario.
When i am receiving a call but i don't answer on my softphone, and i click close the call, it appears on softphone that the call has ended, and after a few seconds its ringing again until the call initiatiator is closing the call.

If i answer the call on my softphone, i can close it with no issue.

I don't have any firewall in between the computer and pbx
I disabled firewall on windows computer.

I am using tcpdump on freepbx to trace packets.

Does anyone had this issue before.?

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@mannebk wrote:

Hi Folks,

I run the FreePBX 14.0.1.1. with Asterisk 14. I did install the BETA back then.

Last week end I did changed the config to "record all cals" by modifiing the extensions via bulk hanlder to force the recording in all cases.

I did the same to my trunks. force recording.

Now I have a funny thing. We make a call to say a service hotline, or a company where we dail the operator. Now we will be conected to the person we want to talk to. When this person pics up the fone, the call is ended. Spot on. repeatably.

I have no clue where I shold start troubleshooting this one.

I didnt find any helpfull infos in google in combination with the recording setting.

would the output from a call with asterisk -Rvvvvvvvvvv be a starter?

Thanks
Cheers Manne

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@Dan2731 wrote:

Hello,

This is a little complicated, so I will do my best to state this issue clearly. When there is a new voicemail, the file name increments in number. Your first voicemail will be "msg0001.wav", your second will be "msg0002.wav" and so on.

Here is what my problem is. I have a customer who we have set a max message limit of 150 messages. We also remove voicemails from the system that are older than 30 days. The issue I'm seeing is that the counter does not reset. The message number continues to go up until it is named "msg0150.wav". Once it hits that number, every new voicemail will be named "msg0150.wav". We have voicemail to email setup and for users who have a lot of voicemails, this becomes a problem because they all have the same name.

Is their a simply way to have the counter reset it self occasionally? Or parse the current date into the file name to make it unique?

Any thoughts or suggestions are much appreciated.

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@LesD wrote:

We have always used Aastra 6757i phones for many years running firmware 2.6.0.2019.

I have just discovered that the 6757i is no more and it seems that the 6737i is virtually identical as far as specification goes.

Can anyone advise as to comparability between the two? Could we mix 57i and 37i phones?

Many thanks.

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@sillsd4622 wrote:

I have an issue that I'm trying to figure out. If I use my 800 number as the caller ID on our phone then some random phone numbers give us busy signals or a message that says the call can't go through. If I take that 800 number off the outbound CID the call goes through with no issues. When I opened a ticket with Les.net they basically said to change the outbound CID for a toll free number to a non-toll free and re test. Taking off the toll free 800 number as mentioned above and it works. The last message on the tickets said that "The 2080 carrier that the call is going out is our carrier of last resort, as destinations that appear as 2080 have the highest cost of termination." What are they trying to tell me or does anyone have a thought as to what I can do if anything in the FreePBX system to get these numbers to go through without issue ? Any help is greatly appreciated.

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@arielcoloma wrote:

Hello, is there a way or a feature to setup a dynamic agent to be logged in and logged out of a queue automatically with no action from the agent?
We have agents from different countries and we'd like them to be logged in to the queue at certain time and logged out after office hours their local time.
Also, it would be nice to get an alert that an expected agent in the queue is not logged in.
Please help.

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@FPlay wrote:

Hi,

After playing the extension list is not returning to the default IVR menu.

1.jpg1838x937 98 KB

IVR Entries
9 -> Play recording -> Extension List

The extension list was recorded and uploaded to "System Records"

The log showing: (phone numbers were changed to from = 0987654321 to=1234567890)

[2017-07-18 15:30:22] VERBOSE[2960][C-000030f3] netsock2.c: Using SIP RTP TOS bits 184
[2017-07-18 15:30:22] VERBOSE[2960][C-000030f3] netsock2.c: Using SIP RTP CoS mark 5
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:1] Set("SIP/VI OUT-000011ee", "__DIRECTION=INBOUND") in new stack
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:2] Gosub("SIP/VI OUT-000011ee", "app-blacklist-check,s,1()") in new stack
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@app-blacklist-check:1] GotoIf("SIP/VI OUT-000011ee", "0?blacklisted") in new stack
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@app-blacklist-check:2] Set("SIP/VI OUT-000011ee", "CALLED_BLACKLIST=1") in new stack
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@app-blacklist-check:3] Return("SIP/VI OUT-000011ee", "") in new stack
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:3] Set("SIP/VI OUT-000011ee", "_FROMDID=1234567890") in new stack
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:4] Set("SIP/VI OUT-000011ee", "CDR(did)=1234567890") in new stack
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:5] ExecIf("SIP/VI OUT-000011ee", "0 ?Set(CALLERID(name)=10987654321)") in new stack
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:6] Set("SIP/VI OUT-000011ee", "__MOHCLASS=") in new stack
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:7] Set("SIP/VI OUT-000011ee", "_REVERSALREJECT=FALSE") in new stack
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:8] GotoIf("SIP/VI OUT-000011ee", "1?post-reverse-charge") in new stack
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx_builtins.c: Goto (from-pstn,1234567890,10)
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:10] NoOp("SIP/VI OUT-000011ee", "") in new stack
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:11] Set("SIP/VI OUT-000011ee", "_CALLINGNAMEPRESSV=allowed_not_screened") in new stack
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:12] Set("SIP/VI OUT-000011ee", "_CALLINGNUMPRESSV=allowed_not_screened") in new stack
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:13] Set("SIP/VI OUT-000011ee", "CALLERID(name-pres)=allowed_not_screened") in new stack
[2017-07-18 15:30:22] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:14] Set("SIP/VI OUT-000011ee", "CALLERID(num-pres)=allowed_not_screened") in new stack
[2017-07-18 15:30:23] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:15] NoOp("SIP/VI OUT-000011ee", "CallerID Entry Point") in new stack
[2017-07-18 15:30:23] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:16] Set("SIP/VI OUT-000011ee", "_CRMDIRECTION=INBOUND") in new stack
[2017-07-18 15:30:23] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:17] Set("SIP/VI OUT-000011ee", "_CRMSOURCE=10987654321") in new stack
[2017-07-18 15:30:23] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:18] Set("SIP/VI OUT-000011ee", "_CRMLINKEDID=1500409822.34879") in new stack
[2017-07-18 15:30:23] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:19] ExecIf("SIP/VI OUT-000011ee", "1?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
[2017-07-18 15:30:23] VERBOSE[20780][C-000030f3] pbx.c: Executing [1234567890@from-pstn:20] Goto("SIP/VI OUT-000011ee", "ivr-3,s,1") in new stack
[2017-07-18 15:30:23] VERBOSE[20780][C-000030f3] pbx_builtins.c: Goto (ivr-3,s,1)
[2017-07-18 15:30:23] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@ivr-3:1] Set("SIP/VI OUT-000011ee", "INVALID_LOOPCOUNT=0") in new stack
[2017-07-18 15:30:23] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@ivr-3:2] Set("SIP/VI OUT-000011ee", "IVRCONTEXT_ivr-3=") in new stack
[2017-07-18 15:30:23] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@ivr-3:3] Set("SIP/VI OUT-000011ee", "IVRCONTEXT=ivr-3") in new stack
[2017-07-18 15:30:23] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@ivr-3:4] Set("SIP/VI OUT-000011ee", "_IVRRETVM=RETURN") in new stack
[2017-07-18 15:30:23] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@ivr-3:5] GotoIf("SIP/VI OUT-000011ee", "0?skip") in new stack
[2017-07-18 15:30:23] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@ivr-3:6] Answer("SIP/VI OUT-000011ee", "") in new stack
[2017-07-18 15:30:23] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@ivr-3:7] Wait("SIP/VI OUT-000011ee", "1") in new stack
[2017-07-18 15:30:24] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@ivr-3:8] Set("SIP/VI OUT-000011ee", "IVR_MSG=custom/Greeting") in new stack
[2017-07-18 15:30:24] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@ivr-3:9] Set("SIP/VI OUT-000011ee", "TIMEOUT(digit)=3") in new stack
[2017-07-18 15:30:24] VERBOSE[20780][C-000030f3] func_timeout.c: Digit timeout set to 3.000
[2017-07-18 15:30:24] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@ivr-3:10] ExecIf("SIP/VI OUT-000011ee", "1?Background(custom/Greeting)") in new stack
[2017-07-18 15:30:24] VERBOSE[20780][C-000030f3] file.c: Playing 'custom/Greeting.slin' (language 'en')
[2017-07-18 15:30:28] VERBOSE[20780][C-000030f3] pbx.c: Executing [9@ivr-3:1] GotoIf("SIP/VI OUT-000011ee", "1?play-system-recording,71,1:,return,1") in new stack
[2017-07-18 15:30:28] VERBOSE[20780][C-000030f3] pbx_builtins.c: Goto (play-system-recording,71,1)
[2017-07-18 15:30:28] VERBOSE[20780][C-000030f3] pbx.c: Executing [71@play-system-recording:1] Answer("SIP/VI OUT-000011ee", "") in new stack
[2017-07-18 15:30:28] VERBOSE[20780][C-000030f3] pbx.c: Executing [71@play-system-recording:2] Playback("SIP/VI OUT-000011ee", "custom/FPBXDirectoryList") in new stack
[2017-07-18 15:30:28] VERBOSE[20780][C-000030f3] file.c: Playing 'custom/FPBXDirectoryList.slin' (language 'en')
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [proxy@dpma_message_context:1] Set("Message/ast_msg_queue", "MESSAGE(custom_data)=mark_all_outbound") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [proxy@dpma_message_context:2] Set("Message/ast_msg_queue", "MESSAGE_DATA(X-Digium-AppServer-Response-URI)=sip:192.168.1.59:5060") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [proxy@dpma_message_context:3] Set("Message/ast_msg_queue", "MESSAGE_DATA(X-Digium-AppServer-Response-FullContact)=") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [proxy@dpma_message_context:4] MessageSend("Message/ast_msg_queue", "digium_phone:blah") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [proxy@dpma_message_context:5] Hangup("Message/ast_msg_queue", "") in new stack
[2017-07-18 15:30:31] VERBOSE[3303] config.c: Parsing '/var/spool/asterisk/voicemail/default/1010/INBOX/msg0000.txt': Found
[2017-07-18 15:30:31] VERBOSE[3303] config.c: Parsing '/var/spool/asterisk/voicemail/default/1010/INBOX/msg0001.txt': Found
[2017-07-18 15:30:31] VERBOSE[3303] config.c: Parsing '/var/spool/asterisk/voicemail/default/1010/INBOX/msg0002.txt': Found
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Spawn extension (dpma_message_context, proxy, 5) exited non-zero on 'Message/ast_msg_queue'
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:1] Set("Message/ast_msg_queue", "MESSAGE(custom_data)=mark_all_outbound") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:2] Set("Message/ast_msg_queue", "TMP_RESPONSE_URI=sip:192.168.1.59:5060") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:3] Set("Message/ast_msg_queue", "MESSAGE_DATA(Request-URI)=") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:4] Set("Message/ast_msg_queue", "MESSAGE_DATA(X-Digium-AppServer-Response-URI)=") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:5] Set("Message/ast_msg_queue", "MESSAGE_DATA(X-Digium-AppServer-Response-FullContact)=") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:6] MessageSend("Message/ast_msg_queue", "sip:192.168.1.59:5060,proxy") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:7] Hangup("Message/ast_msg_queue", "") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Spawn extension (dpma_message_context, digium_phone_module, 7) exited non-zero on 'Message/ast_msg_queue'
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:1] Set("Message/ast_msg_queue", "MESSAGE(custom_data)=mark_all_outbound") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:2] Set("Message/ast_msg_queue", "TMP_RESPONSE_URI=sip:192.168.1.59:5060") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:3] Set("Message/ast_msg_queue", "MESSAGE_DATA(Request-URI)=") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:4] Set("Message/ast_msg_queue", "MESSAGE_DATA(X-Digium-AppServer-Response-URI)=") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:5] Set("Message/ast_msg_queue", "MESSAGE_DATA(X-Digium-AppServer-Response-FullContact)=") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:6] MessageSend("Message/ast_msg_queue", "sip:192.168.1.59:5060,proxy") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:7] Hangup("Message/ast_msg_queue", "") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Spawn extension (dpma_message_context, digium_phone_module, 7) exited non-zero on 'Message/ast_msg_queue'
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:1] Set("Message/ast_msg_queue", "MESSAGE(custom_data)=mark_all_outbound") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:2] Set("Message/ast_msg_queue", "TMP_RESPONSE_URI=sip:192.168.1.59:5060") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:3] Set("Message/ast_msg_queue", "MESSAGE_DATA(Request-URI)=") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:4] Set("Message/ast_msg_queue", "MESSAGE_DATA(X-Digium-AppServer-Response-URI)=") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:5] Set("Message/ast_msg_queue", "MESSAGE_DATA(X-Digium-AppServer-Response-FullContact)=") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:6] MessageSend("Message/ast_msg_queue", "sip:192.168.1.59:5060,proxy") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Executing [digium_phone_module@dpma_message_context:7] Hangup("Message/ast_msg_queue", "") in new stack
[2017-07-18 15:30:31] VERBOSE[2910][C-00000000] pbx.c: Spawn extension (dpma_message_context, digium_phone_module, 7) exited non-zero on 'Message/ast_msg_queue'
[2017-07-18 15:31:46] VERBOSE[20780][C-000030f3] pbx.c: Executing [71@play-system-recording:3] Hangup("SIP/VI OUT-000011ee", "") in new stack
[2017-07-18 15:31:46] VERBOSE[20780][C-000030f3] pbx.c: Spawn extension (play-system-recording, 71, 3) exited non-zero on 'SIP/VI OUT-000011ee'
[2017-07-18 15:31:46] VERBOSE[20780][C-000030f3] app_stack.c: SIP/VI OUT-000011ee Internal Gosub(crm-hangup,s,1) start
[2017-07-18 15:31:46] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@crm-hangup:1] NoOp("SIP/VI OUT-000011ee", "Sending Hangup to CRM") in new stack
[2017-07-18 15:31:46] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@crm-hangup:2] NoOp("SIP/VI OUT-000011ee", "HANGUP CAUSE: 16") in new stack
[2017-07-18 15:31:46] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@crm-hangup:3] ExecIf("SIP/VI OUT-000011ee", "0?Set(_CRMVOICEMAIL=)") in new stack
[2017-07-18 15:31:46] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@crm-hangup:4] NoOp("SIP/VI OUT-000011ee", "MASTER CHANNEL: 1500409822.34879 = 1500409822.34879") in new stack
[2017-07-18 15:31:46] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@crm-hangup:5] GotoIf("SIP/VI OUT-000011ee", "0?return") in new stack
[2017-07-18 15:31:46] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@crm-hangup:6] Set("SIP/VI OUT-000011ee", "_CRMHANGUP=1") in new stack
[2017-07-18 15:31:46] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@crm-hangup:7] AGI("SIP/VI OUT-000011ee", "sangomacrm.agi") in new stack
[2017-07-18 15:31:46] VERBOSE[20780][C-000030f3] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2017-07-18 15:31:46] VERBOSE[20780][C-000030f3] res_agi.c: AGI Script sangomacrm.agi completed, returning 0
[2017-07-18 15:31:46] VERBOSE[20780][C-000030f3] pbx.c: Executing [s@crm-hangup:8] Return("SIP/VI OUT-000011ee", "") in new stack
[2017-07-18 15:31:46] VERBOSE[20780][C-000030f3] app_stack.c: Spawn extension (play-system-recording, 71, 3) exited non-zero on 'SIP/VI OUT-000011ee'
[2017-07-18 15:31:46] VERBOSE[20780][C-000030f3] app_stack.c: SIP/VI OUT-000011ee Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

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@dmanolis79 wrote:

Going to install a new pbxact, do I Download the new FreePBx sng7 and then run the command to activate as a PBxact or do I go with the old route 10.13.-66 ISO

Thanks in advance

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@jessy5765 wrote:

Is the Wiki down for others right now? I am trying to lookup how to work with something and cant access it.

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@alejoprivas wrote:

i want to call Stasis() whenever any of my local extension are called, so far i understand i have to modify the dial plan, but since freepbx generates its own dial plan i have to use extension_custom.conf

so far i have used extension_conft to add this

[from-internal-custom]

include => channel-dump
[channel-dump]

exten => 66,1,stasis(Channel-dump).

but it only works for a non created extension 66 and if i try to use freepbx created local extensions it overrides it and wont make calls.

so is there a way to call Stasis(channel-dump) for all extensions, so when i receive a call, it gets to the asterisk ari and the call keeps going?

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@daveric wrote:

Hi guys. I have problems because my provider requires a minute of time since the router synchronizes until I can connect to the trunk. Otherwise the trunk will always be rejected. As I can tell Asterisk to wait 1 minute to connect to the trunk. Thank you!!

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@MiMar wrote:

Hi

I'm having a problem when call forwarding extensions to outside numbers using the Follow Me feature . Internal calls are forwarded to the outside number fine. External callers receive the msg "The person at ext X is unavailable". I think this is related to the passing of CLID, but I'm not sure how to resolve it. If I forward to an outside number using a ring group with the Fixed CID set, it works fine. My provided states that I need to use P-Asserted-Identity in the call flow, and I have that set in Settings>Advanced>SIP sendrpid. Im running FreePBX 13.0.192.9.

Can anyone point me in the right direction please?

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@dirweb wrote:

Hi all
i have a little problem:

If I make a phone call and after 1 second I do another, the recipient does not hear me. I have to wait 2-3 minutes and try and work again. How is this happening? How can I fix it?

Maybe i need to add some setting on my freepbx? (FreePBX 13.0.192.9)
Thank you

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@connextechs wrote:

Hey guys, I wanted to consult with you about backing up the FreePBX instance.

In a case where the FreePBX is installed in the cloud VPS server, do I really need to use the FreePBX back up when I can just have automatic daily back ups of the whole VPS server enabled, and they would back up the whole thing on a daily basis?

What do you guys think?

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@faisalkhan wrote:

Hi all,

I need to Setup a VPN Server to my AWS freepbx 13 host and Connect to another Freepbx 13 distro via vpn client.

What should I do to connect the other pbx server with AWS server.

I need to make vpn tunneling.

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@tuta wrote:

Hello,

I have FreePBX 12 and asterisk 11 system working without any problems. Recently we attempted to upgrade asterisk to version 13 without any changes to configuration, and while everything works fine, there is one small problem with CDR records generated after DISA calls. Users call to access phone# (2895558888) , type password and dial destination (036555000). Some of this calls are chargeable, so there is a requirement to run "billing" report every month.
3 records are created for each call. In asterisk 11, this is created (only fields of interest in report are dumped)

+---------------------+-------------+-------------+--------------------+-------------+---------+
| calldate            | src         | dst         | dcontext           | disposition | billsec |
+---------------------+-------------+-------------+--------------------+-------------+---------+
| 2017-07-18 22:31:32 | 4165558888  | 2895558888  | from-trunk-sip-001 | ANSWERED    |      13 |
| 2017-07-18 22:31:45 | 4165558888  | 036555000   | from-internal      | ANSWERED    |       1 |
| 2017-07-18 22:31:45 | 4165558888  | 036555000   | disa-dial          | ANSWERED    |       7 |
+---------------------+-------------+-------------+--------------------+-------------+---------+

3rd record is one of my interest, easy to find with dcontext = 'disa-dial', and billsec is actual length of the call.

With asterisk 13, following is created

+---------------------+-------------+-----------+---------------+-------------+---------+
| calldate            | src         | dst       | dcontext      | disposition | billsec |
+---------------------+-------------+-----------+---------------+-------------+---------+
| 2017-07-18 22:18:23 |             | 036555000 | from-internal | ANSWERED    |       0 |
| 2017-07-18 22:18:23 | 4165558888  | 036555000 | disa-dial     | ANSWERED    |       4 |
| 2017-07-18 22:18:27 | 4165558888  | 036555000 | disa-dial     | ANSWERED    |      15 |
+---------------------+-------------+-----------+---------------+-------------+---------+

As you can see, there is a big difference, and main problem for us is duplicate 'disa-dial' record, second one is actual billing record, while first is always 4 seconds.

Is there anything in settings of asterisk 13 (or Freepbx) to bring CDR records content same or similar with asterisk 11? We can not migrate to 13 before this problem is fixed.

Thanks

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