I setup a brand new sip trunk from teliax gui and made sure all the setting were the same from 1.4 to 1.8 asterisk --
I am able to make calls out and the sip provider is registered
When I call in I get the following error
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
[2011-07-25 13:12:29] NOTICE[4568]: chan_sip.c:21515 handle_request_invite: Failed to authenticate device "TELIAX FAX" ;tag=9SyZQtapBjUga
[2011-07-25 13:12:30] WARNING[4568]: chan_sip.c:3551 retrans_pkt: Retransmission timeout reached on transmission 39569499-319d-122f-32bd-00114336bd5a for seqno 15476303 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
[2011-07-25 13:12:31] NOTICE[4568]: chan_sip.c:21515 handle_request_invite: Failed to authenticate device "TELIAX FAX" ;tag=N6D9cZm08v6gN
voip*CLI>
This is my sip.conf
[general]
bindport=5060
bindaddr=0.0.0.0
rtcachefriends=yes
disallow=all
allow=ulaw
allow=alaw
allowguest=no
trustrpid=yes
sendrpid=yes
callerid=unknown
context=from-outside ; Default context for incoming calls
mohinterpret=default
useragent=Thirdlane
allowsubscribe=yes
notifyhold=yes
notifyringing=yes
callcounter=yes ;;=Asterisk-1.6
;;limitonpeer=yes ;;=Asterisk-1.4
videosupport=yes
t38pt_udptl=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
tos_text=af41 ;;=Asterisk-1.6
cos_sip=3 ;;=Asterisk-1.6
cos_audio=5 ;;=Asterisk-1.6
cos_video=4 ;;=Asterisk-1.6
cos_text=3 ;;=Asterisk-1.6
register => username:password@lax.teliax.net ; teliax
[teliax]
qualify=yes
nat=yes
secret=whatever
;=description=teliax voip
host=lax.teliax.net
username=username
dtmfmode=rfc2833
context=from-outside
type=friend
insecure=very
canreinvite=no
disallow=all
allow=ulaw
This is my Trunks.include
TRUNK_NAME_teliax=teliax
TRUNK_PROTOCOL_teliax=SIP
TRUNK_DESCRIPTION_teliax=teliax voip
TRUNK_DIRECTION_teliax=both
TRUNK_PROVIDER_URL_teliax=http://www.teliax.com
TRUNK_TENANT_teliax=
TRUNK_STATUS_teliax=1
TRUNK_STRIP_teliax=0
TRUNK_PREPEND_teliax=
TRUNK_CLI_STRIP_teliax=0
TRUNK_CLI_PREPEND_teliax=
TRUNK_DIALSTRING_teliax=
TRUNK_NAME_g0=g0
TRUNK_PROTOCOL_g0=DAHDI
TRUNK_DESCRIPTION_g0=TDM Card
TRUNK_DIRECTION_g0=both
TRUNK_PROVIDER_URL_g0=
TRUNK_TENANT_g0=
TRUNK_STATUS_g0=1
TRUNK_STRIP_g0=0
TRUNK_PREPEND_g0=
TRUNK_CLI_STRIP_g0=0
TRUNK_CLI_PREPEND_g0=
TRUNK_DIALSTRING_g0=
Thank you for the help
invite is often not enough if
invite is often not enough if a different server handles the media stream. try
insecure=port,invite
this is synonymous with 'very'
Starting with Asterisk 1.6 and above, you need to use insecure=invite instead of insecure=very.