voip*CLI> sip show peer 334
voip*CLI>
* Name : 334
Secret : <Set>
MD5Secret : <Not set>
Context : from-internal
Subscr.Cont. : <Not set>
Language : fr
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox : 334@default
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 50
Dynamic : Yes
Callerid : "device" <334>
MaxCallBR : 384 kbps
Expire : 128
Insecure : no
Nat : Always
ACL : No
T38 pt UDPTL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Video Support: No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 74.56.64.35 Port 60157
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: 334
SIP Options : (none)
Codecs : 0xc (ulaw|alaw)
Codec Order : (ulaw:20,alaw:20)
Auto-Framing: No
Status : OK (61 ms)
Useragent : Aastra 57iCT/2.2.0.166
Reg. Contact : sip:334@74.56.64.35:51620;transport=udp;srcadr=192.168.0.101:5060
sip_nat.conf
nat=yes
externip=69.70.114.174
localnet=10.0.0.0/255.255.255.0
sip.conf
[general]
#include sip_general_additional.conf
language=fr
bindport = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 10.0.0.9 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68
; Reported as required for Asterisk 1.4
notifyringing=yes
notifyhold=yes
limitonpeers=yes
; enable and force the sip jitterbuffer. If these settings are desired
; they should be set in the sip_general_custom.conf file as this file
; will get overwritten during reloads and upgrades.
;
; jbenable=yes
; jbforce=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
My phone sip configuration
[334]
type=friend
secret=xxxxxx
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=334@default
host=dynamic
dtmfmode=rfc2833
disallow=
dial=SIP/334
context=from-internal
canreinvite=no
callgroup=
callerid=device <334>
call-limit=4
allow=
accountcode=
call-limit=50