I have a major issue. A client has a huge call flow. When 3 or more calls come into the office, the phones will drop the calls. 2 of the CSR’s will be on the phone and a third call will come in causing all the calls, the CSR’s were on to drop. It is a ring group with 4 extensions.
Where do I start looking for issues or adjusting settings.
(greenhorn, on this system for 1 year)
/var/log/asterisk/full would be where I’d start.
Next, I’d try having one CSR call another, then have the third try to call his mom. If all the calls drop in that scenario, check your power supply for the phones (assuming the phones are running on POE from the switch).
what about this error in log:
2018-08-13 12:32:22] ERROR[2329] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data
Bad network card or bad switch.
so its possible when that happens their calls could be dropped???
If it’s a bad switch, then all bets are off. If it’s the network card in the PBX, then absolutely.
this pbx has two nics. One configured, one not. If the first nic in the pbx is bad, can i assign the same static ip as the first nic, to the second one, and then just move the cable from the one nic to the other?
Sure - why not? The system (FreePBX) doesn’t look at the specific Ethernet port, it works with the address. If you just remove the first NIC (or disable it in the BIOS) the second NIC should pick up the slack and go to town.
So safe to configure the second nic with same address through pbx gui and then just switch the cable to the second nic onsite?
In other threads you commented how this is generally an error in the sip config. Which is what it mainly is a result from.
@Red post you trunk config.
inbound
type=friend
dtmfmode=auto
username=zepu_hha
secret=XXXXXXXX
context=from-trunk ; (this could be ext-did or from-pstn as well)
insecure=port,invite
canreinvite=no
host=inbound28.vitelity.net
Outbound
username=zepu_hhaout
type=friend
trustrpid=yes
sendrpid=yes
secret=XXXXXXXXX
host=outbound.vitelity.net
fromuser=zepu_hha
dtmfmode=auto
context=from-trunk
canreinvite=no
You are not registering with them I’m take it? So if you’re just peering with them it should be
Inbound
type=peer
dtmfmode=rfc2833
defaultuser=zepu_hha
secret=XXXXXXXX
context=from-trunk ; (this could be ext-did or from-pstn as well)
insecure=port,invite
directmedia=no
disallow=all
allow=ulaw
nat=no
host=inbound28.vitelity.net
Outbound
defaultuser=zepu_hhaout
type=peer
trustrpid=yes
sendrpid=pai
secret=XXXXXXXXX
host=outbound.vitelity.net
fromuser=zepu_hha
dtmfmode=rfc2833
directmedia=no
disallow=all
allow=ulaw
nat=no
We do have a registry string we use with the trunks as well. Especially on the inbound trunk.
Then just put them back as friend instead of peer. Is this PBX behind NAT?
according to firewall and sip settings, yes it is.
So the error I posted earlier is in response to a 991003 extension using a websocket connection. It also shows the IP address of the laptop. Why would this laptop be using a 99 extension to log into the pbx? Would it have anything to do with UCP module and call recordings that the client utilizes?
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