sudo -s
switches me to root and I can run fwconsole from there without issue. So wierd. I have been running Linux in AWS for YEARS but I am no expert.
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Distro update on AWS
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PJSIP hung channels
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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NightMode pbx
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Raspbx 11-11-2019 and no audio between extension calls in local network
I found the solution.
Actually it was on the iphone/softphone.app side.
The RTP range was from 10000 - 60000 , beyond the 10100 i had in my raspbx.
Once I sync’ed them with 10000 - 10100, everything went fine.
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IAX trunk failing call
from that paste (would be nice if text wasn’t wrapped!), looks like you have a phone using pjsip to talk to freepbx and I assume you are expecting the call to go out an IAX trunk to the “grandstream pbx” extension 1001?
Looks like gets DIALSTATUS: CHANUNAVAIL but is dialing PJSIP/1001, which is obviously not IAX.
So you haven’t said if your IAX trunk is shown as up and working?
And you have an outbound route handling extension 1001 that uses the IAX trunk?
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Seeing calls in CDR that do not appear in /var/log/asterisk/full*
I have resolved my problem though did not “fix” the issue. I did not find anything in the knowledge base or forums regarding this and would still like an answer.
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IAX trunk failing call
Hey @nielsen yes. I have been trying to call from the grandstream pbx via the iax trunk to the freepbx. Now I have changed my network environment, strange is that the iax works with the same configuration. I do not know if I had made a blunder with the network settings. I am going to place the other ip back and check it, because I have to place this in production. I am very afraid now to change the ip. I hope the uax will still work. I could before see that there is a trunk, but the calls failed. I do not know what the cause was
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Get user list and create users from CLI?
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IAX trunk failing call
I’m confused. I assume that you are calling extension 2000 on FreePBX from ext. 1001 on GS. If not, please explain.
Can you call 2000 from another extension on FreePBX without issue? If so, please provide a paste of that log for comparison.
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The #OpenSourceLounge
Another good chat in the evening. Enjoyed seeing the live action Dilbert comic playing out. And great to hear solutions given and upcoming tools for FreePBX.
Look forward to the next event!
@lgaetz
Per your idea:
https://dilbert.com/strip/2020-09-10
Chris
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Seeing calls in CDR that do not appear in /var/log/asterisk/full*
Your posts are very vague. Do you mean that’s the only message in the log file for the past 30 minutes? Or that’s the only message with a particular call ID? Or, something else?
Do calls between extensions get properly logged? Outgoing calls? Incoming calls?
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System plays default voicemail message and not recorded message
So I can now answer my own stupid question…
In my haste to test uploading a new audio file directly I renamed the original busy.WAV to temp.wav. It seems that busy.WAV was a bad file, but not in use.
And as those that know (and now me) temp.wav is used by default as the temporary greeting which will take precedence when it exists. So simply deleting the temp.wav fixed my problem.
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Inbound Calls transfer to extensions but hang up after accepting the call
I have raspbx with freepbx.
I’ve set up a trunk which Im able to make calls.
My intention is to transfer incoming calls (from the trunk) to a specific extension (1001).
The incoming call goes to my extension 1001 but once i accept the call connection fails with his:
841[2020-09-12 00:18:50] VERBOSE[10840][C-00000030] app_dial.c: PJSIP/1002-00000031 is ringing
842[2020-09-12 00:18:53] VERBOSE[10840][C-00000030] app_dial.c: PJSIP/1002-00000031 answered SIP/Yuboto-00000014
843[2020-09-12 00:18:53] VERBOSE[10841][C-00000030] bridge_channel.c: Channel PJSIP/1002-00000031 joined 'simple_bridge' basic-bridge <8f52b680-3d16-42be-ac79-31b1baf04538>
844[2020-09-12 00:18:53] VERBOSE[10840][C-00000030] bridge_channel.c: Channel SIP/Yuboto-00000014 joined 'simple_bridge' basic-bridge <8f52b680-3d16-42be-ac79-31b1baf04538>
845[2020-09-12 00:19:01] VERBOSE[10841][C-00000030] bridge_channel.c: Channel PJSIP/1002-00000031 left 'simple_bridge' basic-bridge <8f52b680-3d16-42be-ac79-31b1baf04538>
846[2020-09-12 00:19:01] VERBOSE[10840][C-00000030] bridge_channel.c: Channel SIP/Yuboto-00000014 left 'simple_bridge' basic-bridge <8f52b680-3d16-42be-ac79-31b1baf04538>
847[2020-09-12 00:19:01] VERBOSE[10840][C-00000030] app_macro.c: Spawn extension (macro-dial-one, s, 55) exited non-zero on 'SIP/Yuboto-00000014' in macro 'dial-one'
848[2020-09-12 00:19:01] VERBOSE[10840][C-00000030] app_macro.c: Spawn extension (macro-exten-vm, s, 14) exited non-zero on 'SIP/Yuboto-00000014' in macro 'exten-vm'
849[2020-09-12 00:19:01] VERBOSE[10840][C-00000030] pbx.c: Spawn extension (from-did-direct, 1002, 3) exited non-zero on 'SIP/Yuboto-00000014'
850[2020-09-12 00:19:01] VERBOSE[10840][C-00000030] pbx.c: Executing [h@from-did-direct:1] Macro("SIP/Yuboto-00000014", "hangupcall,") in new stack
851[2020-09-12 00:19:01] VERBOSE[10840][C-00000030] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/Yuboto-00000014", "1?theend") in new stack
852[2020-09-12 00:19:01] VERBOSE[10840][C-00000030] pbx_builtins.c: Goto (macro-hangupcall,s,3)
853[2020-09-12 00:19:01] VERBOSE[10840][C-00000030] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("SIP/Yuboto-00000014", "0?Set(CDR(recordingfile)=)") in new stack
854[2020-09-12 00:19:01] VERBOSE[10840][C-00000030] pbx.c: Executing [s@macro-hangupcall:4] NoOp("SIP/Yuboto-00000014", "PJSIP/1002-00000031 montior file= ") in new stack
855[2020-09-12 00:19:01] VERBOSE[10840][C-00000030] pbx.c: Executing [s@macro-hangupcall:5] GotoIf("SIP/Yuboto-00000014", "1?skipagi") in new stack
856[2020-09-12 00:19:01] VERBOSE[10840][C-00000030] pbx_builtins.c: Goto (macro-hangupcall,s,7)
857[2020-09-12 00:19:01] VERBOSE[10840][C-00000030] pbx.c: Executing [s@macro-hangupcall:7] Hangup("SIP/Yuboto-00000014", "") in new stack
858[2020-09-12 00:19:01] VERBOSE[10840][C-00000030] app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'SIP/Yuboto-00000014' in macro 'hangupcall'
859[2020-09-12 00:19:01] VERBOSE[10840][C-00000030] pbx.c: Spawn extension (from-did-direct, h, 1) exited non-zero on 'SIP/Yuboto-00000014'
860[2020-09-12 00:19:18] WARNING[1265] chan_sip.c: Retransmission timeout reached on transmission 61e482f73d36a9a32aba247a3feea9e2@xx.xx.xx.xx:5060 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
861Packet timed out after 24640ms with no response
Seems like:
- Incoming call comes through and gets transfered to the extension
- Once i accept the call from my extention a bridge channel is created
- Something happens and everything fall apart.
- Hang up.
What amI missing?
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Inbound Calls transfer to extensions but hang up after accepting the call
In Asterisk SIP Settings, confirm that Local Networks and External Address are correctly set. If you change these, you must restart (not just reload) Asterisk.
If this doesn’t help, at the Asterisk command prompt, type
sip set debug on
pjsip set logger on
make a failing incoming call, paste the Asterisk log for the call at pastebin.freepbx.org and post the link here. Also report: After answering, can the caller hear the called extension? Can the called extension hear the caller?
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Announcing new Dashboard feature
Very good. I had suggested an ability to make notes to save info on events and solutions and I am glad that it is now considered useful. Thanks.
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Bad Audio on conference bridge - Follow-up
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Inbound Calls transfer to extensions but hang up after accepting the call
Setting the external address correctly solved the problem. thank you
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Seeing calls in CDR that do not appear in /var/log/asterisk/full*
Sorry, the later. This is the only item in the log with the Call ID. And there are many calls but no log entries. Some of the calls are answered but all are very short connects. All are to one area code in Iowa.
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Seeing calls in CDR that do not appear in /var/log/asterisk/full*
These are likely fraudulent calls to high cost (rural) providers where the attacker gets a piece of the action. Look up the area code / exchange at
https://www.telcodata.us/search-area-code-exchange-detail
Possibly, the calls were made via an unsecured transfer mechanism and not logged correctly due to a bug. Look at the complete log, from one minute before through one minute after a problematic call. Also, search the log for the problematic numbers.
Or, your system was hacked into and SIP credentials were stolen. The CDR should show the extension the call was made from. Search the log for registrations of that extension.
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Voice Mail To Email HTML Format
If you want to send voicemail to email in HTML format from issabel or asterisk below is a link with all the detail that I fallow
https://www.nibbletec.com/issabel-pbx-voice-mail-to-email-html-format
it works for me with the lab environment describe in the article.
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