after some years of flawless operations I was forced to move my Asterisk VM a new hardware platform and additionally moving from vSphere to Proxmox. My old v16 Asterisk VM came up but the service stopped immediately after starting. So I was suspecting some missing features due to compiling on an other hardware and decided to move up tp 20.9.3.
After successfully compiling and starting I used my old Configs, assuming the upgrade will be smooth… But undfortunately there are saome errors I have no idea how to fix after some hours of using Google and “try and error”.
So, here they are:
Oct 1 17:14:53] WARNING[7351]: tcptls.c:656 ast_tcptls_client_start_timeout: Unable to connect SIP socket to 192.168.10.2:49802: Connection refused
[Oct 1 17:15:21] WARNING[7361]: chan_sip.c:17443 check_auth: username mismatch, have <10>, digest has <11>
[Oct 1 17:15:21] NOTICE[7361]: chan_sip.c:29060 handle_request_register: Registration from ‘sip:10@192.168.10.207’ failed for ‘192.168.10.1:51885’ - Username/auth name mismatch
As I am using a Cisco CP8851 with the “usecallmanager” Patch, what was working like a charm in 16, something seems to behave not the way it was in 16.
So I kindly ask the community for help as I’m running out of ideas and the WAF (wife acceptance factor) is slowly turning negative,
Are you able to get any phone to work ? (It does seem isolated to this particular model, but still, it is a whole new setup, so maybe there’s something else messing up packets eg. cert issues, firewall, etc.)
You might want to co-ordinate with this user who recently got a 8811 working (but needs help on a 6941):
Our system is a fresh install in the version 20.7.0. When i read his errors i would say the config form asterisk ore the config from the phone has the false username or password. I would show in the asterisk config and set than the password in the phone config an load it new on the phone.
I have a CP-6941 3 feet from my desk as I am typing this that is working fine on FreePBX 17 and Asterisk 20 with the usecallmanager patch.
You probably jumped the gun a bit on this migration. We are roughly a month away from the release of Asterisk 22 which is the next LTS version of Asterisk. The author of the usecallmanager patch has committed to releasing patches for the LTS versions of Asterisk and no others so I’ve been waiting to see what he does when 22 finishes it’s release candidates and goes live. I’ve already emailed him the fix he needs to make to his patch for Asterisk 22 + the interlink1d maintained chan_sip driver, so I’m expecting all of this to happen right after 22 goes live.
Anyway, remember that the Cisco phones seem to have issues with dropping SIP packets on UDP. I’ve configured them to work on both UDP and TCP as a test and seen no problems when the phones are local - but strange problems with UDP when the phones are 100 miles away behind a gateway2gateway VPN so I have adhered to the recommendation to keep them on TCP.
The XML conf files for all the Cisco phones are dicy. The phones are perfectly happy reading in 3/4 of the XML config then running into a directive they don’t understand and then scotching the remainder of the config file, leaving the phone looking like it’s configured yet constantly trying to register, etc. And, the phones will behave differently under different firmware versions since Cisco will add new directives into later firmware. Many of the XML config files bouncing around in odd corners of the Internet for Cisco phones will work with one firmware version for the phone but not another.
Once 22 comes out I’m going to post a complete soup-to-nuts set of instructions to bring up a FreePBX 17 system with Asterisk 22 and the Usecallmanager patch along with tested functional XML configs for as many different Cisco phones as I have in inventory. The 6941 is one of those.
I also would encourage you to look on Ebay for CP-8845s. You can get them used for around $25 or less. And if you chuck out all your non-videophones, Simens/SPA/etc. and put in videophones everywhere in the house, the “coolness factor” alone might increase the WAF…lol (even though, of course, the video part won’t work on the PSTN)
!