In my day job, we use SD Cards on some communication servers. We get cards that are “high rewrite” ($100 a pop) cards and have to replace them after about a year of use.
We tried consumer-class cards in our “high IO server”, and they lasted about a week. The worst lasted about three days, the best about two weeks.
I use that experience as my benchmark - it’s not an indictment of the Pi, it’s a problem inherent in the SD-Card media.
Set it high enough that every “line button” in the phone can connect. If you phones have three line buttons, set the number to 6.
Phone apps, wipes the config file and sends a reboot. This is not something with the phone.
I’ll get an example in a bit.
You need to look at /var/log/asterisk/full to troubleshoot this. CDR is just Call Details, which isn’t going to help you at all.
@dicko it’s ucp but not the process it’s someone actively looking at calls in ucp
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Unless these Fanvil guys are real idiots, transmission is probably ok. So on an internal call between Cisco and Fanvil, what you hear on the Cisco is likely realistic.
Some thoughts on possibly salvaging the existing phones:
Configure a Fanvil to register to Messagenet directly. (Remove the Messagenet trunk on the PBX so it doesn’t hijack the calls.) Check incoming call quality. If it’s much better, my theory about sequence numbers is probably correct; there may be a way to get Asterisk to “clean up” the incoming RTP so the Fanvil will render it reasonably well.
You may want to investigate the packet loss in more detail. If caused by your router or LAN, it may be easy to fix. If there are errors on your fiber line, you may be able to get the ISP to fix it.
I tried a ping test to voip.vivavox.it (from Paris) and didn’t see a single packet lost:
Ping statistics for 83.211.227.21:
Packets: Sent = 2771, Received = 2771, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 18ms, Maximum = 37ms, Average = 19msCheck the phone’s dial plan.
Two Extensions logged in to two different phones. Phone Apps does this to the config files.
[root@pbx tftpboot]# grep 5782 *.cfg
00085D3BBC1E.cfg:sip line1 auth name: "5782"
00085D3BBC1E.cfg:sip line1 user name: "5782"
00085D3BBC1E.cfg:sip line1 display name: "5782"
00085D3BBC1E.cfg:sip line1 screen name: "5782"
00085D3BBC1E.cfg:action uri registered: "http://pbx.domain.com:82/sync.php?user=5782"
00085D3BBC1E.cfg:services script: "http://pbx.domain.com:82/applications.php/restapps/main?user=5782"
00085D3BBC1E.cfg:xml application URI: "http://pbx.domain.com:82/applications.php/restapps/main?user=5782"
[root@pbx tftpboot]# grep 6115 *.cfg
00085D308FC6.cfg:sip line1 auth name: "6115"
00085D308FC6.cfg:sip line1 user name: "6115"
00085D308FC6.cfg:sip line1 display name: "6115"
00085D308FC6.cfg:sip line1 screen name: "6115"
00085D308FC6.cfg:action uri registered: "http://pbx.domain.com:82/sync.php?user=6115"
00085D308FC6.cfg:services script: "http://pbx.domain.com:82/applications.php/restapps/main?user=6115"
00085D308FC6.cfg:xml application URI: "http://pbx.domain.com:82/applications.php/restapps/main?user=6115"
[root@pbx tftpboot]#
Max Contacts is set to two for both extensions.
When you do that, the Other tab shows this.
But they have disappeared from Endpoint Manager
Now I walk to 6115 and long in as 5782. Poof, there is no more config with 6115 as expected, but there is also only one config with 5782. Not expected.
[root@pbx tftpboot]# grep 6115 *.cfg
[root@pbx tftpboot]# grep 5782 *.cfg
00085D308FC6.cfg:sip line1 auth name: "5782"
00085D308FC6.cfg:sip line1 user name: "5782"
00085D308FC6.cfg:sip line1 display name: "5782"
00085D308FC6.cfg:sip line1 screen name: "5782"
00085D308FC6.cfg:action uri registered: "http://pbx.domain.com:82/sync.php?user=5782"
00085D308FC6.cfg:services script: "http://pbx.domain.com:82/applications.php/restapps/main?user=5782"
00085D308FC6.cfg:xml application URI: "http://pbx.domain.com:82/applications.php/restapps/main?user=5782"
[root@pbx tftpboot]#
Looks like that could be your issue as well:
Interestingly, this time, the existing phone with 5782 did not reboot. So it is still registered, but there is not longer a config tied to that MAC address.
Setting up the SIP version of the 7*** series phones is a known Pain In The A$$.
Setting up Chan-SCCP-B and using the phone in Skinny mode is likewise a PITA.
The SIP version of the phone’s load software uses all of the CPU and memory, so some functions of the phones (like the horizontal buttons) are just awful. The SIP version of the software just isn’t really very friendly. The phones are also EOL, so getting support for them is problematic.
The SCCP version of the phone’s load software is “skinnier”, but it pushes a lot of the functionality you’d expect to be “in the phone” into the PBX.
I use 79xx series phone all the time, but I’m also a user of Chan-SCCP-B and wrote the FreePBX installation that are on GitHub. If you want to try to install them in Skinny mode, I can help you a little bit, but neither method is good for a beginning phone manager.
I’ve had this, but it was because I deleted the destination out from under the system. It sucked finding the problem.
I tried to fix the issue, but ended up tanking the entire PBX. Decision was made to put in on VMWare, and upgrade to version 14 at the same time.
Thank you for your hints though, much appreciated!
Greg
Logged in on a third device and still nothing rebooted, but as expected only two of the device ring when called.
Possibly, there is an issue with codec negotiation. In the phone web interface:
Admin Login > advanced > Voice >Ext_n. Under Audio Configuration, try setting DTMF Tx Method to InBand. If no luck, try AVT.
This is where that “ring group” might come in handy, although I don’t know if it would actually do what you are wanting.
If you set up a ring group with all of the extensions you want to send to calls to in it, you can use the normal “from-internal” context to send to calls to the outbound (local) trunks, which will then use the resiliency you are looking for. Set the outbound route up so that it chooses your local trunks and only sends call to your extensions.
Just stay with known solid SIP phones. Yealink, Sangoma, Snom, Polycom, Grandstream.
You asked for recommendations and I gave them.
Yealink & Sangoma.
I’m fairly certain no one at that location uses the ucp or even knows about it. They do use fop2.
UCP and UCP node server are disabled. I ran service fop2 stop and then started it again. I still see the mysql process popping up with high computer % but it doesn’t seem to stick at 130% forever like it used to.
While I disabled UCP and UCP node, did I need to stop a ucp process before disabling it?
