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We are connecting to a Asterisk/Indosoft Conference bridge using SIP Trunks. I have the ShoreTel side set up as they instructed. Below is the Asterisk config:
-----------------------------SNIP-----------------------------------
[general]
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
nat=yes
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
;externhost = sheeju.dyndns.org
externip = 172.31.5.46
;localnet = 192.168.1.1/255.255.255.0
externrefresh=10
disallow=all
allow=ulaw
allow=alaw
allow=gsm
;dtmfmode=rfc2833
dtmfmode=inband
language=en
; See doc/ip-tos.txt for a description of these parameters.
tos_sip=cs3 ; Sets TOS for SIP packets.
tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;Define our shoretel user
[shoresip]
;Put the IP address of the Shoretel box below
host = 172.30.1.12
port = 5060
type = peer
allow = ulaw
dtmfmode = rfc2833
reinvite = no
canreinvite = no
;When this user(shoretel) dials out, send to the context below
context = default
nat = no
;Maybe add trunk=yes to conserve bandwidth
----------------------------SNIP-----------------------------------
I have the conf bridge extension set up as an Off System Extension on the ShoreTel. Calls to the extension result in a reorder tone.
If we take the config out, calls will go through, but something gets messed up with the disconnect and the call will not drop, resulting in major problems with the end user not being able to dial out.....help?
-----------------------------SNIP-----------------------------------
[general]
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
nat=yes
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
;externhost = sheeju.dyndns.org
externip = 172.31.5.46
;localnet = 192.168.1.1/255.255.255.0
externrefresh=10
disallow=all
allow=ulaw
allow=alaw
allow=gsm
;dtmfmode=rfc2833
dtmfmode=inband
language=en
; See doc/ip-tos.txt for a description of these parameters.
tos_sip=cs3 ; Sets TOS for SIP packets.
tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;Define our shoretel user
[shoresip]
;Put the IP address of the Shoretel box below
host = 172.30.1.12
port = 5060
type = peer
allow = ulaw
dtmfmode = rfc2833
reinvite = no
canreinvite = no
;When this user(shoretel) dials out, send to the context below
context = default
nat = no
;Maybe add trunk=yes to conserve bandwidth
----------------------------SNIP-----------------------------------
I have the conf bridge extension set up as an Off System Extension on the ShoreTel. Calls to the extension result in a reorder tone.
If we take the config out, calls will go through, but something gets messed up with the disconnect and the call will not drop, resulting in major problems with the end user not being able to dial out.....help?