I will see what I can do later, as I would prefer a little different information. For now, it works to pass it.
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How can I pass a variable to the trunk monitor agi script
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How can I pass a variable to the trunk monitor agi script
For anyone curious, I have put the updated version of this on my github.
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How to delete old backups via SSH commands
POMPs_Full_Backup]# 43G ./POMPs_Full_Backup
I can get into that /var/spool/asterisk/backup folder, but i can’t seem to get into the ./POMPs_Full_Backup folder
are the backups in zip or bak files? how can i list those files to use the rm command?
I read this https://documentation.cpanel.net/display/CKB/How+to+Manage+your+Hard+Drive+Space
but i guess i need a more basic step by step to navigate directly to that folder to list the bak files
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FreePBX 15 (PBXact) and UCP question/issue
Just as an additional FYI, I just went through and made sure any real extension has a Userman entry, a couple were missing. That said, it didn’t seem to change anything with my issue of some users seeing one set of options, and the others different (more) options…
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New S505 phone is not displaying incoming caller id
We plugged in a new phone at this client. We have applied the same template as all the other phones. We’ve looked at the extension settings and compared it to a working user with no differences found. But with any incoming calls to this extension it does not display the incoming call id with either an external caller or an internal caller.
Attached is the not working phone and a working phone.
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New S505 phone is not displaying incoming caller id
Can you share the Asterisk CLI (or /var/log/asterisk/full) output that shows up when receiving a test call to that extension? I’d probably look at the extension’s settings to see if there’s some unexpected routing rules going on, but I can’t think of any that would specifically cause this.
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Trouble with Outbound SIP
Our Telco provider sends us calls first over PRIs and if that fails or if they are full, then via SIP. During a PRI outage yesterday we noticed that none of the SIP calls were coming in or out.
I’ve been doing some test calls from the SIP trunks and have found an odd behavior. If I dial my cell phone from the PBX, my cell phone rings and I can answer but my phone on the PBX shows “Trying” on the display. If I hang up the cell phone it rings again, but again the call never completes.
it’s like the signaling starts but never completes.
netsock2.c: Using SIP RTP TOS bits 184
netsock2.c: Using SIP RTP CoS mark 5
app_stack.c: SIP/level3-00010478 Internal Gosub(func-apply-sipheaders,s,1(1)) start
pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf(“SIP/level3-00010478”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
pbx.c: Executing [s@func-apply-sipheaders:2] NoOp(“SIP/level3-00010478”, “Applying SIP Headers to channel SIP/level3-00010478”) in new stack
pbx.c: Executing [s@func-apply-sipheaders:3] Set(“SIP/level3-00010478”, “TECH=SIP”) in new stack
pbx.c: Executing [s@func-apply-sipheaders:4] Set(“SIP/level3-00010478”, “SIPHEADERKEYS=”) in new stack
pbx.c: Executing [s@func-apply-sipheaders:5] While(“SIP/level3-00010478”, “0”) in new stack
app_while.c: Jumping to priority 13
pbx.c: Executing [s@func-apply-sipheaders:14] Return(“SIP/level3-00010478”, “”) in new stack
app_stack.c: Spawn extension (from-trunk-sip-level3, 82145551212, 1) exited non-zero on ‘SIP/level3-00010478’
app_stack.c: SIP/level3-00010478 Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
app_dial.c: Called SIP/level3/2145551212
app_macro.c: Spawn extension (macro-dialout-trunk, s, 33) exited non-zero on ‘PJSIP/5331-00000150’ in macro ‘dialout-trunk’
pbx.c: Spawn extension (from-internal, 82145551212, 6) exited non-zero on ‘PJSIP/5331-00000150’
pbx.c: Executing [h@from-internal:1] Macro(“PJSIP/5331-00000150”, “hangupcall”) in new stack
pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/5331-00000150”, “1?theend”) in new stack
pbx_builtins.c: Goto (macro-hangupcall,s,3)
pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/5331-00000150”, “0?Set(CDR(recordingfile)=)”) in new stack
pbx.c: Executing [s@macro-hangupcall:4] NoOp(“PJSIP/5331-00000150”, "SIP/level3-00010478 montior file= ") in new stack
pbx.c: Executing [s@macro-hangupcall:5] GotoIf(“PJSIP/5331-00000150”, “1?skipagi”) in new stack
pbx_builtins.c: Goto (macro-hangupcall,s,7)
pbx.c: Executing [s@macro-hangupcall:7] Hangup(“PJSIP/5331-00000150”, “”) in new stack
app_macro.c: Spawn extension (macro-hangupcall, s, 7) exited non-zero on ‘PJSIP/5331-00000150’ in macro ‘hangupcall’
pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/5331-00000150’
app_stack.c: PJSIP/5331-00000150 Internal Gosub(crm-hangup,s,1) start
pbx.c: Executing [s@crm-hangup:1] NoOp(“PJSIP/5331-00000150”, “Sending Hangup to CRM”) in new stack
pbx.c: Executing [s@crm-hangup:2] NoOp(“PJSIP/5331-00000150”, “HANGUP CAUSE: 127”) in new stack
pbx.c: Executing [s@crm-hangup:3] ExecIf(“PJSIP/5331-00000150”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
pbx.c: Executing [s@crm-hangup:4] NoOp(“PJSIP/5331-00000150”, “MASTER CHANNEL: 1593033876.183038 = 1593033876.183038”) in new stack
pbx.c: Executing [s@crm-hangup:5] GotoIf(“PJSIP/5331-00000150”, “0?return”) in new stack
pbx.c: Executing [s@crm-hangup:6] Set(“PJSIP/5331-00000150”, “__CRM_HANGUP=1”) in new stack
pbx.c: Executing [s@crm-hangup:7] AGI(“PJSIP/5331-00000150”, “sangomacrm.agi”) in new stack
res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
res_agi.c: <PJSIP/5331-00000150>AGI Script sangomacrm.agi completed, returning 0
pbx.c: Executing [s@crm-hangup:8] Return(“PJSIP/5331-00000150”, “”) in new stack
app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/5331-00000150’
app_stack.c: PJSIP/5331-00000150 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
[2020-06-24 21:25:08] WARNING[16234] chan_sip.c: Retransmission timeout reached on transmission 1fbe12da0b0ada3640b4fdd14674f453@12.156.39.41:5060 for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 31999ms with no response
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Sip client registration ok , debug shows correct action based on dial plan, RTP ports open but no sound
This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.
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XHR response code: 503 XHR responseText: undefined jQuery status: error
This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.
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Trouble with Outbound SIP
Is your PBX behind a NAT device?
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How to delete old backups via SSH commands
The back ups ate up all the disk space and wouldnt allow the device to boot.
perhaps there should be a setting to limit the storage space for back up, not only how many runs. I couldn’t delete any backup files via gui
i had to get tech support to find the commands
df -h then ll then less ./backup
to select the tz file and then the rm command, but i wasnt able to delete the tz file, so i had to delete the sub directory for the back up. not fun.
Also somehow the pbxact device was downloading 12 copies of the new update packages for the last 3 weeks, many copies. so my device was a full as can be with junk and that crashed the boot up process. thankfully asterisk worked well when the rest of the processes couldnt load because of disk space, so phone calls could go on today.
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Zulu mobile ios still rings when set to DND or unavailable
Yes, I do see the extension there. I know that a tech support person from a company that installs the sangoma devices opened up a ticket for this ringing issue.
Also, When I log into zulu, my extension does not show up on fop2 for webrtc device, only ucp will show that the extension is live.
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What are my update paths?
Do any modules currently need to be on Edge for this to work properly, Lorne?
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Asterisk Version Switch from 13 to 16
Now after the upgrade, we are unable to send the mails from GUI.
We have configure Postfix mail client and able to send mails from CLI successfully.
Earlier before the upgrade we could successfully able to send mails from GUI with the exiting setup.
Any pointer for the same
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What are my update paths?
I thought the backup / restore across versions only works for v13 or v14 to v15.
Excerpt from Backup Restore info:
“FreePBX 15 backup also allows you to restore from legacy backups. This means that people can take a backup from an older version of FreePBX (specifically versions 13 and 14) and restore to a FreePBX 15 system.”
This person is trying to go from v12 to v15. Will this work?
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What are my update paths?
There are backup 15 module improvements coming out frequently. At this point, I would recommend upgrading the backup module on the new 15 system to edge before the restore.
[root@pbx ~]# grep "Module: backup Updated" /var/log/asterisk/freepbx.log*
/var/log/asterisk/freepbx.log:[2020-Jun-15 08:13:15] [freepbx.INFO]: Module: backup Updated to version 15.0.10.7 [] []
/var/log/asterisk/freepbx.log:[2020-Jun-16 08:53:14] [freepbx.INFO]: Module: backup Updated to version 15.0.10.8 [] []
/var/log/asterisk/freepbx.log:[2020-Jun-17 06:17:24] [freepbx.INFO]: Module: backup Updated to version 15.0.10.9 [] []
/var/log/asterisk/freepbx.log:[2020-Jun-22 09:07:27] [freepbx.INFO]: Module: backup Updated to version 15.0.10.10 [] []
/var/log/asterisk/freepbx.log:[2020-Jun-23 06:30:21] [freepbx.INFO]: Module: backup Updated to version 15.0.10.11 [] []
/var/log/asterisk/freepbx.log:[2020-Jun-24 06:30:20] [freepbx.INFO]: Module: backup Updated to version 15.0.10.12 [] []↧
What are my update paths?
I would expect it to. I have successfully done restores from earlier versions.
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The #OpenSourceLounge
Bump.
Reminder of tonight’s #OpenSourceLounge session. June 25th at 7 PM EDT
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Sangoma S 206
Hi,
I am experiencing the following problem.
When i ve my phone on-hook and i am lowering the ringer volume lets say to 3.After i am receiving a phone call the ringer volume goes back to tis maximum.
These phones are in a group by the way.
is there any way to control this ?
I am using a sangoma-default template for all sangoma phones i ve in my network.
Also can i control the mic volume because our customers complain that they cannnot hear us.
Thanks
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Restore problems
This topic was automatically closed 365 days after the last reply. New replies are no longer allowed.
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