Hello, I am having troubles with FreePBX and have a question, so I will try and explain as best as I can the problems I am having.
I am currently running FreePBX 2.3.1.7 and a homebrew Asterisk box on the same network. My firewall I am currently using does not use port-forwarding.
Connecting my phones to the homebrew, I am able to have clear conversations internally, to and from an external phone, and phone out and back in with no problems.
With my FreePBX box, I have clear conversations from internal calls and that’s it. When I call to and from an external line, when it does connect, I can talk clearly from my internal phones to the external one, but when there’s audio coming from the external phone, I can hear 7 seconds of clear real-time audio followed by silence for 30 seconds, after which I can start hearing real-time audio for another 7 seconds. When I call out from an internal phone back in, I can’t hear any audio at all from the phones.
I do not have control over the firewall and I can’t makes changes before I have definitive proof that it won’t affect the other asterisk boxes on the network and that it’s necessary. If the other Asterisk boxes don’t require port forwarding, why is it necessary for this one?
My question is whether or not port-forwarding is absolutely necessary to have FreePBX working and why would other flavors of Asterisk not require port forwarding. If port forwarding is something that is necessary for FreePBX, would there be any changes to the other Asterisk set up if the ports it uses are being forwarded to the FreePBX box?
I will post what I can from these servers.
FreePBX settings:
Outgoing Settings
Trunk Name: Trunk-Out
PEER Details:
allow=ulaw
canreinvite=no
context=from-pstn
disallow=all
dtmfmode=rfc2833
fromdomain=unlimitel.ca
host=sip02.unlimitel.ca
insecure=very
nat=yes
port=5060
qualify=no
secret=vvvvvvv
type=peer
username=YYYYYYYYY
Incoming Settings
USER Context: Trunk-In
USER Details:
allow=ulaw
canreinvite=no
context=from-pstn
dtmfmode=rfc2833
host=sip02.unlimitel.ca
insecure=invite
progressinband=no
relaxdtmf=yes
rfc2833compensate=yes
secret=vvvvvvv
type=user
username=YYYYYYYYY
Register String:
YYYYYYYYY:[email protected]/YYYYYYYYY
sip_nat.conf
externip=192.168.xxx.xxx
localnet=192.168.1.0/255.255.255.0
externrefresh=10
sip.conf
; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn’t, try adding “nat=1” to each peer definition to
; solve translation problems.
[general]
#include sip_general_additional.conf
bindaddr=0.0.0.0
srvlookup=no
disallow=all
allow=ulaw
context = from-pstn; Send unknown SIP callers to this context
tos=0x68
; Reported as required for Asterisk 1.4
notifyringing=yes
notifyhold=yes
limitonpeers=yes
; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_general_custom.conf
#include sip_nat.conf
#include sip_registrations_custom.conf
#include sip_registrations.conf
#include sip_custom.conf
#include sip_additional.conf
#include sip_custom_post.conf
rtp.conf
;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
rtpstart=10001
rtpend=20000
Homebrew:
sip.conf
; File created by the Warp web interface version 1.05
[general]
context=sipdefault
realm=asterisk
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
maxexpiry=120
minexpiry=60
defaultexpiry=120
disallow=all
allow=ulaw
relaxdtmf=yes
mohsuggest=default
mohinterpret=default
useragent=Warp v1.05
dtmfmode = rfc2833
videosupport=no
callevents=yes
rtptimeout=60
rtpholdtimeout=300
#include <sip_registrations.conf>
canreinvite=no
jbenable = yes
jbforce = no
jbmaxsize = 200
jbresyncthreshold = 1000
jbimpl = fixed
[authentication]
#include <sip_daphone.conf>
sip_registrations.conf
register => XXXXXXXXX:[email protected]/XXXXXXXXX
sip_daphone.conf
; File created by Unlimitel WEB Interface
; Version 1.05
;
[200]
type=friend
username=200
secret=200
port=5060
nat=yes
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
context=frominternal200
canreinvite=no
insecure=very
musicclass=default
musiconhold=default
callerid=“Aastra-200” <200>
[201]
type=friend
username=201
secret=201
port=5060
nat=yes
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
context=frominternal201
canreinvite=no
insecure=very
musicclass=default
musiconhold=default
callerid=“Linksys-201” <201>
[Unlimitel1]
type=peer
username=XXXXXXXXX
secret=secret
port=5060
nat=yes
canreinvite=no
qualify=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
insecure=very
context=from-pstn
rtp.conf
;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=10000
rtpend=20000
;
; Whether to enable or disable UDP checksums on RTP traffic
;
;rtpchecksums=no
;
; The amount of time a DTMF digit with no ‘end’ marker should be
; allowed to continue (in ‘samples’, 1/8000 of a second)
;
;dtmftimeout=3000
; rtcpinterval = 5000 ; Milliseconds between rtcp reports
;(min 500, max 60000, default 5000)
sip_nat.conf is not present on homebrew