Open a command prompt under windows, turn OFF windows defender firewall, and at the command prompt issue “tftp -i x.x.x.x (where x.x.x.x is the IP of the tftp server) GET provisioningfilename” You can also just specify the filename of any file in the contents of your /tftpboot folder
#2 will tell you if your tftp server on your Debian install is actually handing out provisioning files
Modify /etc/default/tftpd-hpa and in tftpoptions add --verbose 7 This will increase verbosity of the tftp server. Now power cycle the phone and do a tail /var/log/syslog on the debian tftp server. You should see the phone’s IP attempting to obtain it’s provisioning file from the server and getting it successfully.
That should get you going on troubleshooting this.
Hello, I am having an issue with my S500 IP Phone where when I upload a new wallpaper with any dimensions it tells me “Invalid file name!”. I have tried naming it anything I possibly can. I have tried to upload the image as a .png, and a .jpeg/.jpg. I have even tried to upload the image from a Windows 10 PC, a Windows 11 PC, and my MacBook Pro and nothing has worked. There was a previous image on the phone before I factory reset it (I got it from work) but since that I cannot upload a new image. Thank you for any help you can give.
or if Windows is not part of your thinking, use nmap (netmap) from any host on the lan or lsof (list open files of a network flavor) on the PBX itself
nmap -an -sU 192.168.1.3
lsof -i:69
In the absence of tftpd , you will need to install a daemon, have inetd/xinetd control it and have it point to the base directory of where your provisioning files are (verbosity is of course good), but I would take the effort to change the protocol on the phones to use https as they would be more future proof and less likely to cost you money
Hello, I need help for the topic in the subject. Twilio rejects all follow me calls to the external phone number, because they are initiated via the outbound call trunk with original caller id. (So far my understanding).
I’m new to that topic, I found that @MrXirtam here External Transfer via SIP Refer described his solution, but I’m a bit lost here. Which configurations do I need to do, to achieve call transfer to a external twilio number?
arg, I never got this forum understood, he asked me to paste something but I messed it and pasted the wrong thing LOL, he’s on IRC and got it sorted out he will thank you when he gets back to his work
This was resolved, by changing the certificate in use - > Asterisk SIP settings → sip settings [chan_pjsip] → TLS/SSL/SRTP → certificate manager - Changed the self signed cert to the named one being used for the dashboard and which was in use previously.
Could any of the settings mentioned have been changed by an asterisk upgrade?
Asking as this is probably the 3rd or fourth undocumented setting changed while sangoma support had VPN access to the PBX. Do they have internal reporting for what techs modify?
The last remaining item which support said was related to the DPMA module is from phone_utils during the digium phone module - dpma extension spawn or log spam? Have not seen phone_utils have this error log since yesterday, so maybe now that the cert is in play and the phones have been factory reset this will not occur.
[2024-09-14 14:50:18] ERROR[2232] phone_utils.c: Format ‘mac=%s;auth_flags=%d;data=%s;data_len=%d;ran=%s;contact=%s;uri=%s;cfgd=%d;ua=%s;client_cert=%s;origination_network=%s’ truncated. Wrote ‘2096’ with ‘3191’ remaining [2024-09-14 14:50:18] ERROR[2232] phone_utils.c: Format ‘mac=%s;auth_flags=%d;data=%s;data_len=%d;ran=%s;contact=%s;uri=%s;cfgd=%d;ua=%s;client_cert=%s;origination_network=%s’ truncated. Wrote ‘2096’ with ‘3192’ remaining
[2024-09-14 14:50:18] VERBOSE[16300][C-0000000c] pbx.c: Executing [digium_phone_module@dpma_pjsip_message_context:1] Set(“Message/ast_msg_queue”, “MESSAGE(custom_data)=mark_all_outbound”) in new stack
[2024-09-14 14:50:18] VERBOSE[16300][C-0000000c] pbx.c: Executing [digium_phone_module@dpma_pjsip_message_context:2] Set(“Message/ast_msg_queue”, “TMP_RESPONSE_URI=pjsip:192.168.1.12:36787;transport=tls”) in new stack
[2024-09-14 14:50:18] VERBOSE[16300][C-0000000c] pbx.c: Executing [digium_phone_module@dpma_pjsip_message_context:3] Set(“Message/ast_msg_queue”, “MESSAGE_DATA(Request-URI)=”) in new stack
[2024-09-14 14:50:18] VERBOSE[16300][C-0000000c] pbx.c: Executing [digium_phone_module@dpma_pjsip_message_context:4] Set(“Message/ast_msg_queue”, “MESSAGE_DATA(X-Digium-AppServer-Response-URI)=”) in new stack
[2024-09-14 14:50:18] VERBOSE[16300][C-0000000c] pbx.c: Executing [digium_phone_module@dpma_pjsip_message_context:5] Set(“Message/ast_msg_queue”, “MESSAGE_DATA(X-Digium-AppServer-Response-FullContact)=”) in new stack
[2024-09-14 14:50:18] VERBOSE[16300][C-0000000c] pbx.c: Executing [digium_phone_module@dpma_pjsip_message_context:6] MessageSend(“Message/ast_msg_queue”, “pjsip:192.168.1.12:36787;transport=tls,proxy”) in new stack
[2024-09-14 14:50:18] VERBOSE[16300][C-0000000c] pbx.c: Executing [digium_phone_module@dpma_pjsip_message_context:7] Hangup(“Message/ast_msg_queue”, “”) in new stack
[2024-09-14 14:50:18] VERBOSE[16300][C-0000000c] pbx.c: Spawn extension (dpma_pjsip_message_context, digium_phone_module, 7) exited non-zero on ‘Message/ast_msg_queue’
Hello,
One of our customers wants to play MP3 files on IP Phone speakers. I think this could be possible with the multicast paging feature, but I cannot find any Windows software that multicasts MP3 files from PCs to IP Phones.
Do you have any suggestions for this?
My basis is FreePBX 15.0.37.5 running on Debian 10 using chan_sip. I am trying to mirgate to FeePBX 17.0.19.9 and chan_pjsip.
Upon restoring my former configuration, I get an error indicating that the framework was tampered with:
Module: “FreePBX Framework”, File: “/usr/share/asterisk/agi-bin/phpagi-asmanager.php missing”
Module: “FreePBX Framework”, File: “/usr/share/asterisk/agi-bin/phpagi.php missing”
What I tried did not help:
With and without --force:
fwconsole ma downloadinstall framework --force
fwconsole chwon
fwconsole reload
As this is on a VM, I did
backup the import hoping that the chan_sip conversion and deviations in modules would then be out of the way,
reverted the VM to the point before first importing the configuration
imported again
Unfortunately, the result was the same.
Hi Jose, I have wrote up a how to setup freepbx 16 & Asterisk16 as a 64 bit system. As I only need sip & rstp. The tricky bit was getting the Obdc connectors working.
I understand how frustrating this must be with the FreePBX upgrade. It’s a tricky situation when key files go missing after a restore. Have you tried manually checking the file paths or permissions for the missing files (phpagi-asmanager.php and phpagi.php)? Sometimes re-uploading the framework module or reviewing the upgrade documentation for specific steps can help resolve such issues.
It reminds me of the lesson from Surah Al-Asr about the importance of patience and perseverance through challenges. I hope you find a solution soon!
The issue sees to be that the system I am using up to now is an informal install on Debian (similar to this Debian 10 - Asterisk 16 - FreePBX 15 - PHP 7.3 · GitHub) already, but without Asterisk packages drawn from a repository. If this is used as a backup source, the backup seems to contain absolute paths, such as “/usr/share/asterisk”. The target system using FreePBX 17 on Debian 12 does not have anything under “/usr/share/asterisk”. Why that leads to the message that a module would have been tampered with, is beyond my knowledge. I would have thought that the restore should use variables and relative paths, not absolute paths.
I did convert my FreePBX 15 installation to pjsip. That did work. From that, I did try a regular backup plus a limited backup based on only 12 modules hoping to catch key settings, users, extensions, ring groups, routes and trunks while remaining unsure about voicemail. Even the limited backup would suffer from the same issue as above.
Hence, I think that using a lower Asterisk version on FreePBX 17 will not solve the problem.
Unless someone with a good understanding for the backup/restore function can tell me how to avoid the absolute path issue, I would probably create the configuration again manually on FreePBX 17.
