FreePBX 22.214.171.124 / Current Asterisk Version: 16.15.1 / all modules up to date
I have an issue with Cisco 7911G phones. I realize these are ancient phones and some would recommend tossing them in the trash. Be that as it may, but I am a nostalgic old fogey.
Some months ago I upgraded from FreePBX 14 to FreePBX 15. Although I had been running FreePBX 14 on a physical computer, some time before I upgraded to 15, I moved 14 over to a Virtualbox virtual machine for the convenience. If I remember correctly, I used the Backup & Restore module in FreePBX 14 to make a backup, downloaded the FreePBX 15 distro, and then restored the FreePBX backup files to FreePBX 15. As far as I recall, the process worked great, and there were no obvious glitches. I have a number of older IP phones, Cisco Linksys SPAs, Yealinks, Aastras, Cisco 8861s, Cisco 7911s, 7912s, a 7945, a 7975, all of which worked flawlessly (to my mind) in FreePBX 14. Note that as many of the phones as possible are or have been converted to chan_pjsip, the Cisco 7911G and 7912 extensions are chan_sip (I recall that there are problems with the latter phones with chan_pjsip). When I brought these out of storage after upgrading to FreePBX 15, they all registered, and worked great.
All that is, except the Cisco 7911G phones, of which I have five, extensions 1181, 1182, 1183, 1184, and 1185.
The first problem I noticed after upgrading to FreePBX 15 was that the phones would register, and I could see from the Reports==>Asterisk Info==>Peers that, for example, ext. 1183 was registered:
1183/EXT_NUMBER 172.16.0.83 D No No A 5060 OK (1914 ms)
and I could successfully dial ext. 1183 from another extension, say 7525, when I dial ext. 7525 from ext. 1183 I get Allison Smith’s voice saying “Your call cannot be completed as dialed. Please check the number and dial again.”
Here is the corresponding entry from the /var/asterisk/log/full
[2021-03-28 10:56:05] WARNING chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 102 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
SET SIP DEBUG ON: pastebin for more detail in the logs https://pastebin.com/iAbe7BVZ
Note that I can dial ext. 7525 from ext. 4501 and from ext. 4501 to ext. 7525. Note both ext. 4501 and 7245 are chan_pjsip, while as I mentioned ext. 1183 is chan_sip.