Asterisk/Trixbox Trunk Sample Configurations
[vp.thinktel.ca]
disallow=all
allow=ulaw
dtmfmode=rfc2833
fromdomain=vp.thinktel.ca
host=159.18.161.67
username=XXXYYYZZZZ # your SIP authentication number
secret=@@@@@@@@ # your SIP authentication password
nat=yes # or nat=no if using publicly accessible IP
insecure=invite,port
type=friend
context=from-trunk
[eico.thinktel.ca]
disallow=all
allow=ulaw
dtmfmode=rfc2833
fromdomain=eico.thinktel.ca
host=208.68.17.144
username=XXXYYYZZZZ # your SIP authentication number
secret=@@@@@@@@ # your SIP authentication password
nat=no # eico is public IP only
insecure=invite,port
type=friend
context=from-trunk
[edm.trk.tprm.ca]
disallow=all
allow=ulaw
dtmfmode=rfc2833
fromdomain=edm.trk.tprm.ca
host=208.68.17.52
username=XXXYYYZZZZ # your SIP authentication number
secret=@@@@@@@@ # your SIP authentication password
nat=no # or nat=yes if behind NAT
insecure=invite,port
type=friend
context=from-trunk
[tor.trk.tprm.ca]
disallow=all
allow=ulaw
dtmfmode=rfc2833
fromdomain=tor.trk.tprm.ca
host=206.80.250.100
username=XXXYYYZZZZ # your SIP authentication number
secret=@@@@@@@@ # your SIP authentication password
nat=no # or nat=yes if behind NAT
insecure=invite,port
type=friend
context=from-trunk
SIP_NAT.conf file must contain these definitions if you are using NAT to ensure proper audio redirection:
internip=192.168.10.55 # use your PBX internal IP address
localnet=192.168.10.0/255.255.255.0 # use your internal subnet range
externip=207.69.25.120 # use your WAN external IP address
SIP.CONF must contain 'srvlookup=yes' in the General context
One way audio on our TrixBox
Running a TrixBox behind a firewall can cause one way audio issues. On a Trixbox, you will need to specify the external IP as well as the internal network
in sip.conf.
ie:
[general]
externhost = asterisk.company.com
externip=123.123.123.123 ; the ip address your ISP gives you
localnet=192.168.1.0/255.255.255.0
nat=yes
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all ; Modify these to meet your codec needs
allow=ulaw
allow=alaw
allow=g729
allow=gsm
Low volume on voicemail
If you have a SIP trunk and are using Asterisk for voicemail. we have found Asterisk records the volume quite low in its voicemail files.One approach is to use a post-voicemail command in the voicemail configuration, which can run Sox on the saved files and amplify them automatically.
A recommended level for amplification of these is -7dB from maximum.
I just get dead air on inbound call to my Asterisk server occasionally.
If your traces are showing the Thinktel switch responding with a 491 - Request Pending, you must set canreinvite=no to stop this.
Asterisk does not always acknowledge this properly before any audio change is redone.
Asterisk & DTMF
Asterisk normally handles RFC2883 in a strange manner:
From the phone:
Audio Audio Audio DigitStart Duration Duration DigitEnd Audio Audio
It waits until it sees 'DigitEnd' before it does anything, then it sends (in a tight for loop):
DigitStart Start Start End End End (takes about 1ms normally)
So, this consumes the DTMF entirely (duration is lost), and it sends it in such a short period that devices that use the other definition of how long to make DTMF (you can either use Duration field, the Asterisk way, or the start playing with the start message, stop at the end way, which our switch does) end up generating the worlds shortest DTMF sequences.
Thus, the side effect of this patch, by slowing the packet rate down to around 80ms for the entire period, is it won't accept further DTMF/audio for those 80ms (Asterisk, that is, it will delay everything by that amount). So, really fast DTMF dialers can miss a keystroke.
We hope this info was a help. If you have any problems or questions, please let us know.
You can open a support ticket or contact us at
1-888-852-2381 for additional assistance.