Internal and External calls dropping randomly!

I just tried to post the log but it was too big.

One of the warnings I see is:

[2016-01-25 14:27:22] WARNING[2177] chan_sip.c: 
Retransmission timeout reached on transmission 
[email protected] for seqno 115 (Critical Request) -- See 
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2016-01-25 14:27:26] NOTICE[2177] chan_sip.c: Peer 'BroadV-1' is now Lagged. (2072ms / 2000ms)

This phone system has ~50 users/phones

It has been running smoothly for over a year now with BroadVoice SIP Trunk service, but the last few days has been dropping both internal and external calls randomly…

I have 2 examples of call drops that I can find in the logs

In Example 1:
Internal_UserA calls Internal_B
A and B talk for 4 seconds (2way audio)
A is talking and gets garbled, then disconnected
B is still on the line, asks if A is there, then hangs up

In example 2:
Internal_UserC calls Outside line.
C goes thru IVR prompts for 40 seconds
Outside Line IVR gets garbled and disconnects.
C hangs up

Possibly related: I updated all the modules in the phone system a few weeks ago.

I tried rebooting the system today, but still having issues.

Does not happen to every call, only sometimes

PBX Firmware:5.211.65-14
PBX Service Pack:1.0.0.0

That will always be a network/firewall/router/provider problem. You will need to dig deeper with sip debugging.

I agree with dicko. Check to see if you have an unusual amount of traffic on the network. If internal calls are experiencing the same issues, I would look for an infected system, or someone using bittorrent or similar on the network. Try speedtest.net or something similar from a few machines and see if your ping time and bandwidth are sufficient. Even ping machine to machine internally. If your network infrastructure is decent, you should get 2-3ms ping times internally. Ping your phone server and see if it responds any differently than another workstation. Good luck!

Thanks for the help guys

Getting 1ms pings on LAN to the phone in question and the phone system …

I talked to the SIP Trunk provider and he said they have had some problems today, which may or may not be related to the outgoing call drops

But even if SIP Trunk provider is having a bad day - that shouldnt effect my internal calls dropping! Hmmm…

did you do the sip debugging thing? It will clue you in to what is not working.

How can I SIP debug if it happens randomly.

Are there SIP Logs?

Heres some more debug info — it looks like a bunch of extensions are losing connectivity including the SIP Trunk to BroadVoice

Maybe the PBX is having its own network issues on its card or server port?

 [2016-01-25 14:27:21] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 141 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6401ms with no response
    [2016-01-25 14:27:22] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 111 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6400ms with no response
    [2016-01-25 14:27:22] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 111 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6400ms with no response
    [2016-01-25 14:27:22] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 115 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6400ms with no response
    [2016-01-25 14:27:26] NOTICE[2177] chan_sip.c: Peer 'BroadV-1' is now Lagged. (2072ms / 2000ms)
    [2016-01-25 14:27:27] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 112 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6400ms with no response
    [2016-01-25 14:27:27] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 112 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6401ms with no response
    [2016-01-25 14:27:27] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission c69[email protected] for seqno 112 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6400ms with no response
    [2016-01-25 14:27:27] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 112 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6400ms with no response
    [2016-01-25 14:27:27] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 112 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6400ms with no response
    [2016-01-25 14:27:27] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 112 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6400ms with no response
    [2016-01-25 14:27:27] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 111 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6399ms with no response
    [2016-01-25 14:27:27] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 112 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6400ms with no response
    [2016-01-25 14:27:27] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 112 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6400ms with no response
    [2016-01-25 14:27:27] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 111 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6400ms with no response
    [2016-01-25 14:27:27] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 111 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6400ms with no response
    [2016-01-25 14:27:28] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 129 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6401ms with no response
    [2016-01-25 14:27:28] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 111 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6401ms with no response
    [2016-01-25 14:27:40] WARNING[2177] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 111 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
    Packet timed out after 6400ms with no response
    [2016-01-25 14:27:40] NOTICE[2177] chan_sip.c: Peer 'BroadV-1' is now UNREACHABLE! Last qualify: 2072
    [2016-01-25 14:27:44] NOTICE[2177] chan_sip.c: Peer '104' is now UNREACHABLE! Last qualify: 9
    [2016-01-25 14:27:44] VERBOSE[2114] chan_sip.c: == Extension Changed 104[ext-local] new state Unavailable for Notify User 102
    [2016-01-25 14:27:44] VERBOSE[2114] chan_sip.c: == Extension Changed 104[ext-local] new state Unavailable for Notify User 101
    [2016-01-25 14:27:44] VERBOSE[2114] chan_sip.c: == Extension Changed 104[ext-local] new state Unavailable for Notify User 112
    [2016-01-25 14:27:44] VERBOSE[2114] chan_sip.c: == Extension Changed 104[ext-local] new state Unavailable for Notify User 108
    [2016-01-25 14:27:44] VERBOSE[2114] chan_sip.c: == Extension Changed 104[ext-local] new state Unavailable for Notify User 106
    [2016-01-25 14:27:44] VERBOSE[2114] chan_sip.c: == Extension Changed 104[ext-local] new state Unavailable for Notify User 104
    [2016-01-25 14:27:44] VERBOSE[2114] chan_sip.c: == Extension Changed 104[ext-local] new state Unavailable for Notify User 115
    [2016-01-25 14:27:44] VERBOSE[2114] chan_sip.c: == Extension Changed 104[ext-local] new state Unavailable for Notify User 117
    [2016-01-25 14:27:44] VERBOSE[2114] chan_sip.c: == Extension Changed 104[ext-local] new state Unavailable for Notify User 123
    [2016-01-25 14:27:44] VERBOSE[2114] chan_sip.c: == Extension Changed 104[ext-local] new state Unavailable for Notify User 100
    [2016-01-25 14:27:44] VERBOSE[2114] chan_sip.c: == Extension Changed 104[ext-local] new state Unavailable for Notify User 114
    [2016-01-25 14:27:44] VERBOSE[2114] chan_sip.c: == Extension Changed 104[ext-local] new state Unavailable for Notify User 111
    [2016-01-25 14:27:44] VERBOSE[2114] chan_sip.c: == Extension Changed 104[ext-local] new state Unavailable for Notify User 125
    [2016-01-25 14:27:44] VERBOSE[2114] chan_sip.c: == Extension Changed 104[ext-local] new state Unavailable for Notify User 116
    [2016-01-25 14:27:44] VERBOSE[2114] chan_sip.c: == Extension Changed 104[ext-local] new state Unavailable for Notify User 113
    [2016-01-25 14:27:45] NOTICE[2177] chan_sip.c: Peer '112' is now Reachable. (10ms / 2000ms)

You have a network problem on your 10.1.1.0 network, replace the hardware or correct the route. to watch :-

tcpdump -i (eth?) -nnvv net 10.1.1.0/(whatever)

Can you run a ping from the phone server to one of your phones and just leave it running? That might tell you if it’s something specific to SIP if the pings are fine while SIP is not responding…

hmmmm … network card issue or switch issue … possibly

I can set up a ping or tcpdump or sip debug and watch it for a few minutes but everything is working fine right now, so I won’t see any problems…

I guess I will have to wait until it happens again to get more info

It happened only 2x today