I am trying to configure a VegaStream 50 FXO to work with asterisk. The problem that I am having is that the VegaStream does not support incoming registration from asterisk. VegaStream only allows outbound registration. My question is does asterisk allow incoming registration from an FXO? If yes how? Or better yet, has anybody been able to make the VegaStream FXO work with asterisk? According to VegaStream they have many clients using this combo but they haven't been very helpful otherwise.
Hi Isaac,
I am a newbie to Asterisk (hoping to set up a system for my office) and
I have been struggling with the Vega 5010 (10 FXO) as well.
I've had the same problem as you, being able to call out, but not
receive calls. I just found a solution (for my setup atleast).
First off, I have the Vega set up according to some very basic
instructions from this list (for a different Vega) and from the getting
started setup guide from the Vegastream CD.
I have "enable registration" set (under SIP options in the Vega's
web
config), which makes the Vega register with Asterisk. I am currently
testing with only one line coming into port 1, which has an
"Authentication Number" (see PSTN options in Vega's web config) of
01
and a "Interface Number" of 06. Basically, the vega registers with
username 01, but sends the call to asterisk with 06@{vega's ip here} as
the address.
When I'd do a "sip debug" during an incoming call, I'd see
asterisk
responding with a "SIP/2.0 404 Not Found" error, causing the vega to
answer and immediately hang up.
I figured asterisk was looking for SIP user 06, so I added it, but I
still got 404's. Turns out I just needed an EXTENSION, 06. I can now
make calls and receive them, too. Of course, if you have multiple
incoming lines, you'd need extension 06, 07, 08 ... etc, since each port
has its own "Interface Number" (by default), to allow routing of calls
made to different lines.
I hope that helps some. If not, I can send my complete configs,
although I'm sure there's some other problems with them. Now, if only I
could get rid of the echo, I'd be a happy man!
Pete Doyle
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Issac
Simchayof
Sent: Friday, June 09, 2006 7:10 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] FXO registration and VegaStream
I am trying to configure a VegaStream 50 FXO to work with asterisk. The
problem that I am having is that the VegaStream does not support
incoming registration from asterisk. VegaStream only allows outbound
registration.
My question is does asterisk allow incoming registration from an FXO? If
yes how? Or better yet, has anybody been able to make the VegaStream FXO
work with asterisk? According to VegaStream they have many clients using
this combo but they haven't been very helpful otherwise.
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Hi Issac,
Ok, here goes :) Again, my disclaimer-- I'm pretty new to Asterisk, so
I'm sure half of this is not needed or potentially even misconfigured.
You will even see some lines commented out, since I wanted to test if
they were needed--they weren't. I'm hoping to clean everything up and
put it on the wiki -- hopefully next week or two. Also, these are from
Asterisk @ Home, so there might be some changes needed for your setup.
*****************************************************************
Sip.conf - "context" line may differ from A@H Defaults
*****************************************************************
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
*****************************************************************
* Sip_additional.conf -
* I haven't tested DTMF on incoming calls-- you may have to
* change dtmfmode to inband (rfc2833 didn't work for the outgoing
* calls). Also, the context may need to be changed for security?
* I only have an entry for 01 since I am testing with 1 line only
*****************************************************************
... snip ...
[01] ;most lines added by A@H, may not be necessary (i.e. mailbox)
username=01
type=friend
secret=...my vega's password for line 1... (see POTS in Vega's web
config)
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=01@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=device <01>
... snip ...
; commented out, doesn't seem to be needed
;[vega]
;type=user
;dtmfmode=inband
;disallow=all
;context=from-pstn
;allow=ulaw
[vega-gw]
type=peer
host=192.168.1.30 ; my vega's IP address
dtmfmode=inband ;DTMF doesn't work with rfc2833, unfortunately
disallow=all
;context=from-internal ; commenting out, makes context default to
from-sip-external?
allow=ulaw ;only allow ulaw
*****************************************************************
* extensions_additional.conf - dials extension 106 on incoming
* call. I think there's some special A@H magic happening in the
* macro to dial 106. You could just have something like Dial()
* happen here.
*
* After adding the "06" extension, that is when incoming calls
* start going through.
*
* You could also use the s extension somehow, as Mike showed us
* (I need to read up a little!! :) )
*****************************************************************
exten => 06,1,Macro(exten-vm,novm,06)
exten => 06,hint,SIP/106
*****************************************************************
* Configuration Change Report from the Vegastream
* (shows changes from factory settings)
*****************************************************************
Report on configuration changes (verbose)
Configuration changes:
Key: CU: Changed from factory and unsaved.
C-: Changed from factory and saved.
-U: Not changed but unsaved.
[call_control.timers.1]
T301_timeout=90
T301_cause=18
[dsp.g711Alaw64k]
VADU_threshold=0
VP_FIFO_max_delay=160
VP_FIFO_nom_delay=60
echo_tail_size=16
idle_noise_level=-7000
packet_time_max=30
packet_time_min=10
packet_time_step=10
rx_gain=0
tx_gain=0
[dsp.g711Alaw64k.data]
EC_enable=disable
[dsp.g711Alaw64k.voice]
EC_enable=enable
[dsp.g711Ulaw64k] ;I'm only using Ulaw, so this is the only codec set
up
VADU_threshold=0
C- VP_FIFO_max_delay=60
*factory=160
C- VP_FIFO_nom_delay=10 ; I figured reducing this is ok (Asterisk ->
vega is on a LAN), and might reduce delay?
*factory=40
C- echo_tail_size=8 ; EC trains much faster @ 8ms tail for me (we are
close to CO)
*factory=16
idle_noise_level=-7000
C- packet_time_max=20 ; Asterisk requires 20ms packets for ULAW
*factory=30
C- packet_time_min=20 ; Asterisk requires 20ms packets for ULAW
*factory=10
packet_time_step=10
rx_gain=0
tx_gain=0
[dsp.g711Ulaw64k.data]
EC_enable=disable
[dsp.g711Ulaw64k.voice]
EC_enable=enable
[dsp.g729AnnexA]
VADU_threshold=0
VP_FIFO_max_delay=500
VP_FIFO_nom_delay=60
echo_tail_size=16
idle_noise_level=-7000
packet_time_max=80
packet_time_min=10
packet_time_step=10
rx_gain=0
tx_gain=0
[dsp.g729AnnexA.voice]
EC_enable=enable
[dsp.g729]
VADU_threshold=0
VP_FIFO_max_delay=500
VP_FIFO_nom_delay=80
echo_tail_size=16
idle_noise_level=-7000
packet_time_max=80
packet_time_min=10
packet_time_step=10
rx_gain=0
tx_gain=0
[dsp.g729.voice]
EC_enable=enable
[dsp.g7231]
VADU_threshold=0
VP_FIFO_max_delay=500
VP_FIFO_nom_delay=30
echo_tail_size=16
idle_noise_level=-7000
packet_time_max=60
packet_time_min=30
packet_time_step=30
rx_gain=0
tx_gain=0
[dsp.g7231.voice]
EC_enable=enable
[dsp.t38]
FP_FIFO_nom_delay=300
cd_threshold=-33
network_timeout=150
packet_time=40
rate_max=144
rate_min=24
rate_step=24
timeout=15
tx_level=-8
[lan]
ftp=0.0.0.0
C- gateway=192.168.1.1 ; Lan's gateway address (assigned by DHCP, I
think)
*factory=0.0.0.0
C- ip=192.168.1.30 ; IP of Vega box (assigned by DHCP, I think)
*factory=0.0.0.0
C- name=vega50 ; hostname for the vega
*factory=this_hostname
C- ntp=209.204.172.153 ; I think this is an IP address for a public
NTP server
*factory=0.0.0.0
ntp_local_offset=0000
ntp_poll_interval=0
qos_profile=1
subnet=255.255.255.0
C- tftp=192.168.1.10 ; My TFTP server, for downloading firmware
*factory=0.0.0.0
use_dhcp=1
[lan.dhcp]
get_dns=1
get_gateway=1
get_ntp=1
get_tftp=1
[lan.dns_server.1]
C- ip=192.168.1.10
*factory=0.0.0.0
[lan.dns_server.2]
ip=0.0.0.0
[lan.dns_server.3]
ip=0.0.0.0
[lan.host.1]
ip=127.0.0.1
name=loopback
[lan.nat]
enable=0
private_subnet_list_index=1
[lan.nat.port_entry.1]
external_port_min=0
internal_port_range_index=0
name=port_name
[lan.nat.port_list.1]
list=all
name=default_port_list
[lan.nat.profile.1]
external_ip=0.0.0.0
port_list_index=0
[lan.phy]
C- full_duplex=1 ; forced to full duplex - the vega kept going into
half-duplex by default
*factory=0
10baset=1
100basetx=1
[lan.private_subnet.1]
ip=0.0.0.0
name=subnet_name
subnet=255.255.255.0
[lan.private_subnet_list.1]
list=all
name=default_subnet_list
[lan.8021q]
accept_non_tagged=1
enable=0
[logger]
bill_warn_threshold=90
max_billings=100
max_messages=100
prompt=%n%p>
[logger.radius]
max_retry_time=4000
name=this_radius_hostname
retries=4
retry_time=500
window_size=10
[logger.radius.attributes]
overload_session_id=cisco_compatible_format
[logger.radius.attributes.accounting]
acct_delay_time=1
acct_input_octets=1
acct_output_octets=1
acct_session_id=1
acct_session_time=1
acct_status_type=1
acct_terminate_cause=1
[logger.radius.attributes.cisco_vsa]
call_origin=1
call_type=1
connect_time=1
connection_id=1
disconnect_cause=1
disconnect_time=1
gateway_id=1
remote_gateway_id=1
setup_time=1
voice_quality=1
[logger.radius.attributes.standard]
called_station_id=1
calling_station_id=1
nas_identifier=1
nas_ip_address=1
nas_port=1
nas_port_type=1
user_name=1
[logger.radius.server.1]
enable=0
ipname=0.0.0.0
port=1813
secret=testing123
[logger.radius.server.2]
enable=0
ipname=0.0.0.0
port=1813
secret=testing123
[media.cap.1]
C- codec=g711Ulaw64k ; set default codec to Ulaw
*factory=g7231
[media.cap.2]
C- codec=g729 ; (set secondary codec preference to g729)
*factory=g711Alaw64k
[media.cap.3]
C- codec=g7231 ; (set 3rd codec preference to g723)
*factory=g711Ulaw64k
[media.cap.4]
codec=t38tcp
[media.cap.5]
codec=t38udp
[media.control.1]
dynamic_update=0
dynamic_update_freq=0
[media.packet.g711Alaw64k.1]
C- VADU_enable_flag=0
*factory=1
C- out_of_band_DTMF=1
*factory=0
C- packet_time=20
*factory=30
[media.packet.g711Alaw64k.2]
VADU_enable_flag=0
out_of_band_DTMF=0
packet_time=20
[media.packet.g711Ulaw64k.1]
C- VADU_enable_flag=0
*factory=1
out_of_band_DTMF=0 ;I tried setting this to 1, but asterisk didn't
pick up on the DTMF tones
packet_time=20
[media.packet.g711Ulaw64k.2]
VADU_enable_flag=0
out_of_band_DTMF=0
packet_time=20
[media.packet.g729AnnexA.1]
C- VADU_enable_flag=0
*factory=1
C- out_of_band_DTMF=1
*factory=0
packet_time=20
[media.packet.g729.1]
VADU_enable_flag=0
C- out_of_band_DTMF=1 ; not sure if this is correct anymore - I'm
testing with only ULAW at this point
*factory=0
packet_time=20
[media.packet.g7231.1]
C- VADU_enable_flag=0
*factory=1
out_of_band_DTMF=1
packet_time=30
[media.packet.t38tcp.1]
max_rate=144
tcf=local
[media.packet.t38udp.1]
max_rate=144
tcf=transferred
[mib2.communities.1]
get=1
name=public
set=1
traps=1
[mib2.managers.1]
community=public
ip=0.0.0.0
subnet=255.255.255.0
[mib2.system]
sysContact=www.abcdefghijwhatever.com
sysLocation=PlanetEarth
[planner.group.1]
active_times=0000-2359
cause=0
gatekeeper=off
lan=active
name=LAN_Up
priority=0
[planner.group.2]
active_times=0000-2359
cause=0
gatekeeper=off
lan=inactive
name=LAN_Down
priority=0
[planner.group.3]
C- active_times=0000-2359
*factory=New entry
C- cause=34
*factory=New entry
C- gatekeeper=off
*factory=New entry
C- lan=active
*factory=New entry
C- name=POTS
*factory=New entry
C- priority=0
*factory=New entry
[planner.post_profile]
enable=0
[planner.post_profile.plan.1]
dest=TYPE:international
enable=0
name=International
srce=TEL:00<.*>
[planner.profile.1]
C- enable=0
*factory=1
name=default
[planner.profile.1.plan.1]
cost=0
dest=IF:99,TEL:<1>
group=1
name=Normal
srce=IF:0[6-9],TEL:<.*>
[planner.profile.1.plan.2]
cost=0
dest=IF:99,TEL:<1>
group=1
name=Normal
srce=IF:1[0-3],TEL:<.*>
[planner.profile.1.plan.3]
cost=0
dest=IF:<1>,TEL:<1>
group=1
name=Normal
srce=IF:99,TEL:<..>
[planner.profile.1.plan.4]
cost=0
dest=IF:56
group=2
name=Fallback1
srce=IF:0[6-9]
[planner.profile.1.plan.5]
cost=0
dest=IF:57
group=2
name=Fallback2
srce=IF:1[0-3]
[planner.profile.2]
C- enable=1
*factory=New entry
C- name=FXOInAndOutAnyPort
*factory=New entry
[planner.profile.2.plan.1]
C- cost=0
*factory=New entry
C- dest=IF:99,TEL:<1>
*factory=New entry
C- group=3
*factory=New entry
C- name=IncomingAnyPort
*factory=New entry
C- srce=IF:<[^9].>
*factory=New entry
[planner.profile.2.plan.2]
C- cost=0
*factory=New entry
C- dest=IF:06,TEL:<1>
*factory=New entry
C- group=3
*factory=New entry
C- name=To_FXO1
*factory=New entry
C- srce=IF:99,TEL:8<.*> ; I have asterisk dial an "8" before
the
number to tell the vega to choose the first available port for dialing
out. This part detects that. Actually, I may need to do some tweaking
here, this might work for line 1 only.
*factory=New entry
[planner.whitelist]
enable=0
[planner.whitelist.1]
name=default
number=IF:.*
[pots.port.1]
C- callerid=off ; Prevent 6 second wait for incoming lines without
caller id (like mine)
*factory=on
enable=1
fx_profile=1
C- lyr1=g711Ulaw64k ; default to ulaw codec
*factory=g711Alaw64k
C- nt=0
*factory=1
C- tx_gain=1 ; I'm not 100% sure - I think you have to set this to 1
to get your TX/RX gains to take effect
*factory=0
[pots.port.1.if.1]
auth_username=port1
auth_usernumber=01
cost=1
dn=06
interface=06
C- password=...my password for port 1... (removed for obvious reasons
:) )
*factory=user1
profile=1
reg_enable=1
ring_index=2
username=port1
usernumber=01
[pots.port.2]
C- callerid=off
*factory=on
enable=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
C- nt=0
*factory=1
C- tx_gain=1
*factory=0
[pots.port.2.if.1]
auth_username=port2
auth_usernumber=02
cost=1
dn=07
interface=07
C- password=...my password for port 2...
*factory=user2
profile=1
reg_enable=1
ring_index=2
username=port2
usernumber=02
[pots.port.3]
C- callerid=off
*factory=on
enable=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
C- nt=0
*factory=1
tx_gain=0
[pots.port.3.if.1]
auth_username=port3
auth_usernumber=03
cost=1
dn=08
interface=08
password=user3
profile=1
reg_enable=1
ring_index=2
username=port3
usernumber=03
[pots.port.4]
callerid=on
C- enable=0 ;set to 0 to prevent from registering with asterisk (since
I'm testing with 1 line only)
*factory=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
C- nt=0
*factory=1
tx_gain=0
[pots.port.4.if.1]
auth_username=port4
auth_usernumber=04
cost=1
dn=09
interface=09
password=user4
profile=1
reg_enable=1
ring_index=2
username=port4
usernumber=04
[pots.port.5]
callerid=on
C- enable=0
*factory=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
C- nt=0
*factory=1
tx_gain=0
[pots.port.5.if.1]
auth_username=port5
auth_usernumber=05
cost=1
dn=10
interface=10
password=user5
profile=1
reg_enable=1
ring_index=2
username=port5
usernumber=05
[pots.port.6]
callerid=on
C- enable=0
*factory=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
C- nt=0
*factory=1
tx_gain=0
[pots.port.6.if.1]
auth_username=port6
auth_usernumber=06
cost=1
dn=11
interface=11
password=user6
profile=1
reg_enable=1
ring_index=2
username=port6
usernumber=06
[pots.port.7]
callerid=on
C- enable=0
*factory=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
C- nt=0
*factory=1
tx_gain=0
[pots.port.7.if.1]
auth_username=port7
auth_usernumber=07
cost=1
dn=12
interface=12
password=user7
profile=1
reg_enable=1
ring_index=2
username=port7
usernumber=07
[pots.port.8]
callerid=on
C- enable=0
*factory=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
C- nt=0
*factory=1
tx_gain=0
[pots.port.8.if.1]
auth_username=port8
auth_usernumber=08
cost=1
dn=13
interface=13
password=user8
profile=1
reg_enable=1
ring_index=2
username=port8
usernumber=08
[pots.port.9]
callerid=on
C- enable=0
*factory=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
nt=0
tx_gain=0
[pots.port.9.if.1]
auth_username=port9
auth_usernumber=09
cost=1
dn=56
interface=56
password=user9
profile=1
reg_enable=1
ring_index=2
username=port9
usernumber=09
[pots.port.10]
callerid=on
C- enable=0
*factory=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
nt=0
tx_gain=0
[pots.port.10.if.1]
auth_username=port10
auth_usernumber=10
cost=1
dn=57
interface=57
password=user10
profile=1
reg_enable=1
ring_index=2
username=port10
usernumber=10
[pots.profile.1]
C- auth_username_prefix=vega50_
*factory=NULL
C- auth_username_suffix=NULL
*factory=unit1
auth_usernumber_prefix=NULL
C- auth_usernumber_suffix=NULL
*factory=01
callerid_type=off
callerid_wait=6000
dtmf_dial_digit=*
dtmf_dial_timeout=10
line_busy_cause=17
C- username_prefix=vega50_
*factory=NULL
C- username_suffix=NULL
*factory=unit1
usernumber_prefix=NULL
C- usernumber_suffix=NULL
*factory=01
voice_detect=0
[qos_profile.stats]
cdr_detail=low
enable=0
max_no_cdrs=100
monitoring_interval=300
monitoring_threshold=50
qos_warn_threshold=80
[qos_profile.stats.events.call.average_jitter]
enable=0
threshold=50
[qos_profile.stats.events.call.jitter_buf_overflow]
enable=0
[qos_profile.stats.events.call.jitter_buf_underflow]
enable=0
[qos_profile.stats.events.call.packet_error_rate]
enable=0
threshold_rate=5
[qos_profile.stats.events.call.packet_loss]
enable=0
threshold_rate=5
[qos_profile.stats.events.call.pkt_playout_delay]
enable=0
threshold=250
[qos_profile.stats.events.gateway.average_jitter]
enable=0
threshold=50
[qos_profile.stats.events.gateway.lan_link]
enable=0
[qos_profile.stats.events.gateway.packet_loss]
enable=0
threshold_rate=5
[qos_profile.stats.events.gateway.pkt_playout_delay]
enable=0
threshold=250
[qos_profile.stats.report]
frequency=50
method=off
type=gateway
[qos_profile.1]
name=Default
[qos_profile.1.tos]
default_priority=0x00
media_priority=0x00
signalling_priority=0x00
[qos_profile.1.8021q]
default_priority=0
media_priority=0
signalling_priority=0
vlan_id=0
vlan_name=Default
[qos_profile.2]
name=Voice
[qos_profile.2.tos]
default_priority=0x00
media_priority=0x00
signalling_priority=0x00
[qos_profile.2.8021q]
default_priority=0
media_priority=0
signalling_priority=0
[sip]
PRACK=off
C- RTP_AVP=0,8,18,4
*factory=0
T1=500
T2=4000
C- accept_non_proxy_invites=1
*factory=0
cost=1
C- default_proxy=192.168.1.251 ; my asterisk server
*factory=0.0.0.0
dtmf_info=mode1
dtmf_transport=rfc2833
enable_fax=1
enable_modem=1
enable_t38=1
fax_detect=terminating
interface=99
local_rx_port=5060
max_calls=60
media_control_profile=0
modem_detect=terminating
qos_profile=0
C- reg_domain=192.168.1.30
*factory=abcdefghijwhatever.com
reg_enable=1
reg_expiry=600
C- reg_on_startup=1
*factory=0
C- reg_proxy=192.168.1.251
*factory=0.0.0.0
reg_remote_rx_port=5060
reg_req_uri_port=5060
remote_rx_port=5060
req_uri_port=5060
rfc2833_payload=96
sig_transport=udp
signalling_app_id=none
[sip.backup_proxy]
min_valid_response=180
mode=normal
timeout_ms=5000
[sip.backup_proxy.1]
C- enable=0
*factory=1
ipname=0.0.0.0
port=5060
[sip.backup_proxy.2]
C- enable=0
*factory=1
[suppserv]
enable=0
[suppserv.profile.1]
code_blind_xfer=*98*
code_call_clear=*52
code_call_cycle=!
code_consult_xfer=*99
dial_timeout=10
recall=!
termination=#
xfer_on_hangup=1
[tones]
busytone_seq=3
callwait1_seq=6
callwait2_seq=7
dialtone_seq=1
fastbusy_seq=4
ringback_seq=5
stutterd_seq=2
[tones.def.1]
amp1=6000
amp2=6000
amp3=0
amp4=0
freq1=350
freq2=440
freq3=0
freq4=0
name=dialtone
off_time=0
on_time=0
repeat=1
[tones.def.2]
amp1=6000
amp2=6000
amp3=0
amp4=0
freq1=350
freq2=440
freq3=0
freq4=0
name=stutter_dialtone
off_time=100
on_time=100
repeat=1
[tones.def.3]
amp1=5000
amp2=5000
amp3=0
amp4=0
freq1=480
freq2=620
freq3=0
freq4=0
name=busy
off_time=500
on_time=500
repeat=1
[tones.def.4]
amp1=5000
amp2=5000
amp3=0
amp4=0
freq1=480
freq2=620
freq3=0
freq4=0
name=fastbusy
off_time=300
on_time=300
repeat=1
[tones.def.5]
amp1=5000
amp2=5000
amp3=0
amp4=0
freq1=480
freq2=440
freq3=0
freq4=0
name=ringing
off_time=4000
on_time=2000
repeat=1
[tones.def.6]
amp1=32000
amp2=32000
amp3=32000
amp4=32000
freq1=1400
freq2=2060
freq3=2450
freq4=2600
name=offhook_warning
off_time=100
on_time=100
repeat=1
[tones.def.7]
amp1=5000
amp2=0
amp3=0
amp4=0
freq1=440
freq2=0
freq3=0
freq4=0
name=callwait
off_time=50
on_time=300
repeat=0
[tones.net]
disc=0
fail=0
ring=1
[tones.seq.1]
name=dial_seq
repeat=0
[tones.seq.1.tone.1]
duration=600000
play_tone=1
[tones.seq.1.tone.2]
duration=0
play_tone=6
[tones.seq.2]
name=stutter_dial_seq
repeat=0
[tones.seq.2.tone.1]
duration=2000
play_tone=2
[tones.seq.2.tone.2]
duration=598000
play_tone=1
[tones.seq.2.tone.3]
duration=0
play_tone=6
[tones.seq.3]
name=busy_seq
repeat=0
[tones.seq.3.tone.1]
duration=0
play_tone=3
[tones.seq.4]
name=fastbusy_seq
repeat=0
[tones.seq.4.tone.1]
duration=0
play_tone=4
[tones.seq.5]
name=ringing_seq
repeat=0
[tones.seq.5.tone.1]
duration=0
play_tone=5
[tones.seq.6]
name=callwait1_seq
repeat=0
[tones.seq.6.tone.1]
duration=350
play_tone=7
[tones.seq.7]
name=callwait2_seq
[tones.seq.7.tone.1]
duration=150
play_tone=7
[tones.seq.7.tone.2]
duration=150
play_tone=132
[tones.seq.7.tone.3]
duration=150
play_tone=7
[tones.seq.7.tone.4]
duration=150
play_tone=132
[tones.seq.7.tone.5]
duration=300
play_tone=7
[users.admin]
billing=0
logging=3
prompt=%u%p>
remote_access=1
C- timeout=1200 ; increase web admin timeout to 20 minutes instead of
the way-too-short 4 minutes!
*factory=240
[users.billing]
billing=1
logging=0
prompt=%u%p>
remote_access=1
timeout=0
[users.user]
billing=0
logging=3
prompt=%u%p>
remote_access=1
timeout=0
Total changed: 89 Unsaved: 0
*****************************************************************
* Additional notes
*****************************************************************
Things I need to work through still:
- Reduce echo much more
- Get all lines working correctly / config in asterisk
- currently only have line 1 working (due to early testing)
- Route certain lines to different extensions (use extensions
06,07,08,09 ... etc)
- Integrate incoming calls from vega to work however A@H deals with
incoming calls (use correct macro on incoming call), so that I can
configure behavior from AMP (Automated Attendants, etc).
Hopefully something here will help you. I hope to re-do / clean up much
of my config in the next couple weeks. Hopefully I will be able post
the results to the wiki (see Vegastream). HTH!
Good Luck!
Pete Doyle
--
Children of the Nations
http://www.cotni.org
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Issac
Simchayof
Sent: Saturday, June 10, 2006 6:23 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] FXO registration and VegaStream
Pete,
Thanks for the reply! If you don't mind I would love to take a look at
the script I am sure it will be great help.
Thanks,
Issac
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Peter
Doyle
Sent: Saturday, June 10, 2006 5:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] FXO registration and VegaStream
Hi Isaac,
I am a newbie to Asterisk (hoping to set up a system for my office) and
I have been struggling with the Vega 5010 (10 FXO) as well.
I've had the same problem as you, being able to call out, but not
receive calls. I just found a solution (for my setup atleast).
First off, I have the Vega set up according to some very basic
instructions from this list (for a different Vega) and from the getting
started setup guide from the Vegastream CD.
I have "enable registration" set (under SIP options in the Vega's
web
config), which makes the Vega register with Asterisk. I am currently
testing with only one line coming into port 1, which has an
"Authentication Number" (see PSTN options in Vega's web config) of
01
and a "Interface Number" of 06. Basically, the vega registers with
username 01, but sends the call to asterisk with 06@{vega's ip here} as
the address.
When I'd do a "sip debug" during an incoming call, I'd see
asterisk
responding with a "SIP/2.0 404 Not Found" error, causing the vega to
answer and immediately hang up.
I figured asterisk was looking for SIP user 06, so I added it, but I
still got 404's. Turns out I just needed an EXTENSION, 06. I can now
make calls and receive them, too. Of course, if you have multiple
incoming lines, you'd need extension 06, 07, 08 ... etc, since each port
has its own "Interface Number" (by default), to allow routing of calls
made to different lines.
I hope that helps some. If not, I can send my complete configs,
although I'm sure there's some other problems with them. Now, if only I
could get rid of the echo, I'd be a happy man!
Pete Doyle
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Issac
Simchayof
Sent: Friday, June 09, 2006 7:10 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] FXO registration and VegaStream
I am trying to configure a VegaStream 50 FXO to work with asterisk. The
problem that I am having is that the VegaStream does not support
incoming registration from asterisk. VegaStream only allows outbound
registration.
My question is does asterisk allow incoming registration from an FXO? If
yes how? Or better yet, has anybody been able to make the VegaStream FXO
work with asterisk? According to VegaStream they have many clients using
this combo but they haven't been very helpful otherwise.
_______________________________________________
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Issac,
I think the "destroying calls" part is coming from having the
registration fail. Instead of using [13] in sip.conf, try [08] (based
on what you just sent). THEN, in extensions.conf, make sure you can
handle extension 13. Extension 13 probably has to be in the same
context as your sip.conf entry for [08] (i.e. from-trunk in this
case)--I'm not 100% sure about this, but I think that's the way it
works. Then reload and see if you can call in (on port 8's phone line).
Basically, it seems like the vega registers using account n (i.e. in
sip.conf), but sends calls (by default) to extension n+5 (i.e in
extensions.conf)
You can see in my example, sip.conf has an entry for [01] (I'm testing
with port 1 only right now). However, calls to port 1 are sent (by the
vega) to extension 06 in extensions.conf.
So, when my line 1 rings, extension 06 runs.
[In your example, you can see this in the From line you sent (from
08@...) and the To line (to: 13@...).]
Pete
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Issac
Simchayof
Sent: Monday, June 12, 2006 8:41 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] FXO registration and VegaStream
I do have extension 13 in sip.conf but I still get Destroying call on
all
incoming calls coming from VegaStream.
[13]
type=user
dtmfmode=inband
disallow=all
context=from-trunk
allow=alaw
<-- SIP read from 192.168.0.5:5060:
ACK sip:13@209.219.90.167:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.5:5060;branch=z9hG4bK-vega1-000A-0001-002A-C9DAAC64
From: "FJLine2" <sip:08@192.168.0.5>;tag=0000-002B-F7607240
To: <sip:13@209.219.90.167>;tag=as572a6b57
Max-Forwards: 70
Call-ID: 0014-002A-D9212F10-0@192.168.0.5
CSeq: 6433460 ACK
Contact: <sip:13@192.168.0.5:5060;maddr=192.168.0.5>
User-Agent: VEGAPOTS/09.02.07xS008
Content-Length: 0
--- (10 headers 0 lines)---
Destroying call '0014-002A-D9212F10-0@192.168.0.5'
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Peter
Doyle
Sent: Monday, June 12, 2006 1:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] FXO registration and VegaStream
Hi Issac,
Ok, here goes :) Again, my disclaimer-- I'm pretty new to Asterisk, so
I'm sure half of this is not needed or potentially even misconfigured.
You will even see some lines commented out, since I wanted to test if
they were needed--they weren't. I'm hoping to clean everything up and
put it on the wiki -- hopefully next week or two. Also, these are from
Asterisk @ Home, so there might be some changes needed for your setup.
*****************************************************************
Sip.conf - "context" line may differ from A@H Defaults
*****************************************************************
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
*****************************************************************
* Sip_additional.conf -
* I haven't tested DTMF on incoming calls-- you may have to
* change dtmfmode to inband (rfc2833 didn't work for the outgoing
* calls). Also, the context may need to be changed for security?
* I only have an entry for 01 since I am testing with 1 line only
*****************************************************************
... snip ...
[01] ;most lines added by A@H, may not be necessary (i.e. mailbox)
username=01
type=friend
secret=...my vega's password for line 1... (see POTS in Vega's web
config)
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=01@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=device <01>
... snip ...
; commented out, doesn't seem to be needed
;[vega]
;type=user
;dtmfmode=inband
;disallow=all
;context=from-pstn
;allow=ulaw
[vega-gw]
type=peer
host=192.168.1.30 ; my vega's IP address
dtmfmode=inband ;DTMF doesn't work with rfc2833, unfortunately
disallow=all
;context=from-internal ; commenting out, makes context default to
from-sip-external?
allow=ulaw ;only allow ulaw
*****************************************************************
* extensions_additional.conf - dials extension 106 on incoming
* call. I think there's some special A@H magic happening in the
* macro to dial 106. You could just have something like Dial()
* happen here.
*
* After adding the "06" extension, that is when incoming calls
* start going through.
*
* You could also use the s extension somehow, as Mike showed us
* (I need to read up a little!! :) )
*****************************************************************
exten => 06,1,Macro(exten-vm,novm,06)
exten => 06,hint,SIP/106
*****************************************************************
* Configuration Change Report from the Vegastream
* (shows changes from factory settings)
*****************************************************************
Report on configuration changes (verbose)
Configuration changes:
Key: CU: Changed from factory and unsaved.
C-: Changed from factory and saved.
-U: Not changed but unsaved.
[call_control.timers.1]
T301_timeout=90
T301_cause=18
[dsp.g711Alaw64k]
VADU_threshold=0
VP_FIFO_max_delay=160
VP_FIFO_nom_delay=60
echo_tail_size=16
idle_noise_level=-7000
packet_time_max=30
packet_time_min=10
packet_time_step=10
rx_gain=0
tx_gain=0
[dsp.g711Alaw64k.data]
EC_enable=disable
[dsp.g711Alaw64k.voice]
EC_enable=enable
[dsp.g711Ulaw64k] ;I'm only using Ulaw, so this is the only codec set
up
VADU_threshold=0
C- VP_FIFO_max_delay=60
*factory=160
C- VP_FIFO_nom_delay=10 ; I figured reducing this is ok (Asterisk ->
vega is on a LAN), and might reduce delay?
*factory=40
C- echo_tail_size=8 ; EC trains much faster @ 8ms tail for me (we are
close to CO)
*factory=16
idle_noise_level=-7000
C- packet_time_max=20 ; Asterisk requires 20ms packets for ULAW
*factory=30
C- packet_time_min=20 ; Asterisk requires 20ms packets for ULAW
*factory=10
packet_time_step=10
rx_gain=0
tx_gain=0
[dsp.g711Ulaw64k.data]
EC_enable=disable
[dsp.g711Ulaw64k.voice]
EC_enable=enable
[dsp.g729AnnexA]
VADU_threshold=0
VP_FIFO_max_delay=500
VP_FIFO_nom_delay=60
echo_tail_size=16
idle_noise_level=-7000
packet_time_max=80
packet_time_min=10
packet_time_step=10
rx_gain=0
tx_gain=0
[dsp.g729AnnexA.voice]
EC_enable=enable
[dsp.g729]
VADU_threshold=0
VP_FIFO_max_delay=500
VP_FIFO_nom_delay=80
echo_tail_size=16
idle_noise_level=-7000
packet_time_max=80
packet_time_min=10
packet_time_step=10
rx_gain=0
tx_gain=0
[dsp.g729.voice]
EC_enable=enable
[dsp.g7231]
VADU_threshold=0
VP_FIFO_max_delay=500
VP_FIFO_nom_delay=30
echo_tail_size=16
idle_noise_level=-7000
packet_time_max=60
packet_time_min=30
packet_time_step=30
rx_gain=0
tx_gain=0
[dsp.g7231.voice]
EC_enable=enable
[dsp.t38]
FP_FIFO_nom_delay=300
cd_threshold=-33
network_timeout=150
packet_time=40
rate_max=144
rate_min=24
rate_step=24
timeout=15
tx_level=-8
[lan]
ftp=0.0.0.0
C- gateway=192.168.1.1 ; Lan's gateway address (assigned by DHCP, I
think)
*factory=0.0.0.0
C- ip=192.168.1.30 ; IP of Vega box (assigned by DHCP, I think)
*factory=0.0.0.0
C- name=vega50 ; hostname for the vega
*factory=this_hostname
C- ntp=209.204.172.153 ; I think this is an IP address for a public
NTP server
*factory=0.0.0.0
ntp_local_offset=0000
ntp_poll_interval=0
qos_profile=1
subnet=255.255.255.0
C- tftp=192.168.1.10 ; My TFTP server, for downloading firmware
*factory=0.0.0.0
use_dhcp=1
[lan.dhcp]
get_dns=1
get_gateway=1
get_ntp=1
get_tftp=1
[lan.dns_server.1]
C- ip=192.168.1.10
*factory=0.0.0.0
[lan.dns_server.2]
ip=0.0.0.0
[lan.dns_server.3]
ip=0.0.0.0
[lan.host.1]
ip=127.0.0.1
name=loopback
[lan.nat]
enable=0
private_subnet_list_index=1
[lan.nat.port_entry.1]
external_port_min=0
internal_port_range_index=0
name=port_name
[lan.nat.port_list.1]
list=all
name=default_port_list
[lan.nat.profile.1]
external_ip=0.0.0.0
port_list_index=0
[lan.phy]
C- full_duplex=1 ; forced to full duplex - the vega kept going into
half-duplex by default
*factory=0
10baset=1
100basetx=1
[lan.private_subnet.1]
ip=0.0.0.0
name=subnet_name
subnet=255.255.255.0
[lan.private_subnet_list.1]
list=all
name=default_subnet_list
[lan.8021q]
accept_non_tagged=1
enable=0
[logger]
bill_warn_threshold=90
max_billings=100
max_messages=100
prompt=%n%p>
[logger.radius]
max_retry_time=4000
name=this_radius_hostname
retries=4
retry_time=500
window_size=10
[logger.radius.attributes]
overload_session_id=cisco_compatible_format
[logger.radius.attributes.accounting]
acct_delay_time=1
acct_input_octets=1
acct_output_octets=1
acct_session_id=1
acct_session_time=1
acct_status_type=1
acct_terminate_cause=1
[logger.radius.attributes.cisco_vsa]
call_origin=1
call_type=1
connect_time=1
connection_id=1
disconnect_cause=1
disconnect_time=1
gateway_id=1
remote_gateway_id=1
setup_time=1
voice_quality=1
[logger.radius.attributes.standard]
called_station_id=1
calling_station_id=1
nas_identifier=1
nas_ip_address=1
nas_port=1
nas_port_type=1
user_name=1
[logger.radius.server.1]
enable=0
ipname=0.0.0.0
port=1813
secret=testing123
[logger.radius.server.2]
enable=0
ipname=0.0.0.0
port=1813
secret=testing123
[media.cap.1]
C- codec=g711Ulaw64k ; set default codec to Ulaw
*factory=g7231
[media.cap.2]
C- codec=g729 ; (set secondary codec preference to g729)
*factory=g711Alaw64k
[media.cap.3]
C- codec=g7231 ; (set 3rd codec preference to g723)
*factory=g711Ulaw64k
[media.cap.4]
codec=t38tcp
[media.cap.5]
codec=t38udp
[media.control.1]
dynamic_update=0
dynamic_update_freq=0
[media.packet.g711Alaw64k.1]
C- VADU_enable_flag=0
*factory=1
C- out_of_band_DTMF=1
*factory=0
C- packet_time=20
*factory=30
[media.packet.g711Alaw64k.2]
VADU_enable_flag=0
out_of_band_DTMF=0
packet_time=20
[media.packet.g711Ulaw64k.1]
C- VADU_enable_flag=0
*factory=1
out_of_band_DTMF=0 ;I tried setting this to 1, but asterisk didn't
pick up on the DTMF tones
packet_time=20
[media.packet.g711Ulaw64k.2]
VADU_enable_flag=0
out_of_band_DTMF=0
packet_time=20
[media.packet.g729AnnexA.1]
C- VADU_enable_flag=0
*factory=1
C- out_of_band_DTMF=1
*factory=0
packet_time=20
[media.packet.g729.1]
VADU_enable_flag=0
C- out_of_band_DTMF=1 ; not sure if this is correct anymore - I'm
testing with only ULAW at this point
*factory=0
packet_time=20
[media.packet.g7231.1]
C- VADU_enable_flag=0
*factory=1
out_of_band_DTMF=1
packet_time=30
[media.packet.t38tcp.1]
max_rate=144
tcf=local
[media.packet.t38udp.1]
max_rate=144
tcf=transferred
[mib2.communities.1]
get=1
name=public
set=1
traps=1
[mib2.managers.1]
community=public
ip=0.0.0.0
subnet=255.255.255.0
[mib2.system]
sysContact=www.abcdefghijwhatever.com
sysLocation=PlanetEarth
[planner.group.1]
active_times=0000-2359
cause=0
gatekeeper=off
lan=active
name=LAN_Up
priority=0
[planner.group.2]
active_times=0000-2359
cause=0
gatekeeper=off
lan=inactive
name=LAN_Down
priority=0
[planner.group.3]
C- active_times=0000-2359
*factory=New entry
C- cause=34
*factory=New entry
C- gatekeeper=off
*factory=New entry
C- lan=active
*factory=New entry
C- name=POTS
*factory=New entry
C- priority=0
*factory=New entry
[planner.post_profile]
enable=0
[planner.post_profile.plan.1]
dest=TYPE:international
enable=0
name=International
srce=TEL:00<.*>
[planner.profile.1]
C- enable=0
*factory=1
name=default
[planner.profile.1.plan.1]
cost=0
dest=IF:99,TEL:<1>
group=1
name=Normal
srce=IF:0[6-9],TEL:<.*>
[planner.profile.1.plan.2]
cost=0
dest=IF:99,TEL:<1>
group=1
name=Normal
srce=IF:1[0-3],TEL:<.*>
[planner.profile.1.plan.3]
cost=0
dest=IF:<1>,TEL:<1>
group=1
name=Normal
srce=IF:99,TEL:<..>
[planner.profile.1.plan.4]
cost=0
dest=IF:56
group=2
name=Fallback1
srce=IF:0[6-9]
[planner.profile.1.plan.5]
cost=0
dest=IF:57
group=2
name=Fallback2
srce=IF:1[0-3]
[planner.profile.2]
C- enable=1
*factory=New entry
C- name=FXOInAndOutAnyPort
*factory=New entry
[planner.profile.2.plan.1]
C- cost=0
*factory=New entry
C- dest=IF:99,TEL:<1>
*factory=New entry
C- group=3
*factory=New entry
C- name=IncomingAnyPort
*factory=New entry
C- srce=IF:<[^9].>
*factory=New entry
[planner.profile.2.plan.2]
C- cost=0
*factory=New entry
C- dest=IF:06,TEL:<1>
*factory=New entry
C- group=3
*factory=New entry
C- name=To_FXO1
*factory=New entry
C- srce=IF:99,TEL:8<.*> ; I have asterisk dial an "8" before
the
number to tell the vega to choose the first available port for dialing
out. This part detects that. Actually, I may need to do some tweaking
here, this might work for line 1 only.
*factory=New entry
[planner.whitelist]
enable=0
[planner.whitelist.1]
name=default
number=IF:.*
[pots.port.1]
C- callerid=off ; Prevent 6 second wait for incoming lines without
caller id (like mine)
*factory=on
enable=1
fx_profile=1
C- lyr1=g711Ulaw64k ; default to ulaw codec
*factory=g711Alaw64k
C- nt=0
*factory=1
C- tx_gain=1 ; I'm not 100% sure - I think you have to set this to 1
to get your TX/RX gains to take effect
*factory=0
[pots.port.1.if.1]
auth_username=port1
auth_usernumber=01
cost=1
dn=06
interface=06
C- password=...my password for port 1... (removed for obvious reasons
:) )
*factory=user1
profile=1
reg_enable=1
ring_index=2
username=port1
usernumber=01
[pots.port.2]
C- callerid=off
*factory=on
enable=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
C- nt=0
*factory=1
C- tx_gain=1
*factory=0
[pots.port.2.if.1]
auth_username=port2
auth_usernumber=02
cost=1
dn=07
interface=07
C- password=...my password for port 2...
*factory=user2
profile=1
reg_enable=1
ring_index=2
username=port2
usernumber=02
[pots.port.3]
C- callerid=off
*factory=on
enable=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
C- nt=0
*factory=1
tx_gain=0
[pots.port.3.if.1]
auth_username=port3
auth_usernumber=03
cost=1
dn=08
interface=08
password=user3
profile=1
reg_enable=1
ring_index=2
username=port3
usernumber=03
[pots.port.4]
callerid=on
C- enable=0 ;set to 0 to prevent from registering with asterisk (since
I'm testing with 1 line only)
*factory=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
C- nt=0
*factory=1
tx_gain=0
[pots.port.4.if.1]
auth_username=port4
auth_usernumber=04
cost=1
dn=09
interface=09
password=user4
profile=1
reg_enable=1
ring_index=2
username=port4
usernumber=04
[pots.port.5]
callerid=on
C- enable=0
*factory=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
C- nt=0
*factory=1
tx_gain=0
[pots.port.5.if.1]
auth_username=port5
auth_usernumber=05
cost=1
dn=10
interface=10
password=user5
profile=1
reg_enable=1
ring_index=2
username=port5
usernumber=05
[pots.port.6]
callerid=on
C- enable=0
*factory=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
C- nt=0
*factory=1
tx_gain=0
[pots.port.6.if.1]
auth_username=port6
auth_usernumber=06
cost=1
dn=11
interface=11
password=user6
profile=1
reg_enable=1
ring_index=2
username=port6
usernumber=06
[pots.port.7]
callerid=on
C- enable=0
*factory=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
C- nt=0
*factory=1
tx_gain=0
[pots.port.7.if.1]
auth_username=port7
auth_usernumber=07
cost=1
dn=12
interface=12
password=user7
profile=1
reg_enable=1
ring_index=2
username=port7
usernumber=07
[pots.port.8]
callerid=on
C- enable=0
*factory=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
C- nt=0
*factory=1
tx_gain=0
[pots.port.8.if.1]
auth_username=port8
auth_usernumber=08
cost=1
dn=13
interface=13
password=user8
profile=1
reg_enable=1
ring_index=2
username=port8
usernumber=08
[pots.port.9]
callerid=on
C- enable=0
*factory=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
nt=0
tx_gain=0
[pots.port.9.if.1]
auth_username=port9
auth_usernumber=09
cost=1
dn=56
interface=56
password=user9
profile=1
reg_enable=1
ring_index=2
username=port9
usernumber=09
[pots.port.10]
callerid=on
C- enable=0
*factory=1
fx_profile=1
C- lyr1=g711Ulaw64k
*factory=g711Alaw64k
nt=0
tx_gain=0
[pots.port.10.if.1]
auth_username=port10
auth_usernumber=10
cost=1
dn=57
interface=57
password=user10
profile=1
reg_enable=1
ring_index=2
username=port10
usernumber=10
[pots.profile.1]
C- auth_username_prefix=vega50_
*factory=NULL
C- auth_username_suffix=NULL
*factory=unit1
auth_usernumber_prefix=NULL
C- auth_usernumber_suffix=NULL
*factory=01
callerid_type=off
callerid_wait=6000
dtmf_dial_digit=*
dtmf_dial_timeout=10
line_busy_cause=17
C- username_prefix=vega50_
*factory=NULL
C- username_suffix=NULL
*factory=unit1
usernumber_prefix=NULL
C- usernumber_suffix=NULL
*factory=01
voice_detect=0
[qos_profile.stats]
cdr_detail=low
enable=0
max_no_cdrs=100
monitoring_interval=300
monitoring_threshold=50
qos_warn_threshold=80
[qos_profile.stats.events.call.average_jitter]
enable=0
threshold=50
[qos_profile.stats.events.call.jitter_buf_overflow]
enable=0
[qos_profile.stats.events.call.jitter_buf_underflow]
enable=0
[qos_profile.stats.events.call.packet_error_rate]
enable=0
threshold_rate=5
[qos_profile.stats.events.call.packet_loss]
enable=0
threshold_rate=5
[qos_profile.stats.events.call.pkt_playout_delay]
enable=0
threshold=250
[qos_profile.stats.events.gateway.average_jitter]
enable=0
threshold=50
[qos_profile.stats.events.gateway.lan_link]
enable=0
[qos_profile.stats.events.gateway.packet_loss]
enable=0
threshold_rate=5
[qos_profile.stats.events.gateway.pkt_playout_delay]
enable=0
threshold=250
[qos_profile.stats.report]
frequency=50
method=off
type=gateway
[qos_profile.1]
name=Default
[qos_profile.1.tos]
default_priority=0x00
media_priority=0x00
signalling_priority=0x00
[qos_profile.1.8021q]
default_priority=0
media_priority=0
signalling_priority=0
vlan_id=0
vlan_name=Default
[qos_profile.2]
name=Voice
[qos_profile.2.tos]
default_priority=0x00
media_priority=0x00
signalling_priority=0x00
[qos_profile.2.8021q]
default_priority=0
media_priority=0
signalling_priority=0
[sip]
PRACK=off
C- RTP_AVP=0,8,18,4
*factory=0
T1=500
T2=4000
C- accept_non_proxy_invites=1
*factory=0
cost=1
C- default_proxy=192.168.1.251 ; my asterisk server
*factory=0.0.0.0
dtmf_info=mode1
dtmf_transport=rfc2833
enable_fax=1
enable_modem=1
enable_t38=1
fax_detect=terminating
interface=99
local_rx_port=5060
max_calls=60
media_control_profile=0
modem_detect=terminating
qos_profile=0
C- reg_domain=192.168.1.30
*factory=abcdefghijwhatever.com
reg_enable=1
reg_expiry=600
C- reg_on_startup=1
*factory=0
C- reg_proxy=192.168.1.251
*factory=0.0.0.0
reg_remote_rx_port=5060
reg_req_uri_port=5060
remote_rx_port=5060
req_uri_port=5060
rfc2833_payload=96
sig_transport=udp
signalling_app_id=none
[sip.backup_proxy]
min_valid_response=180
mode=normal
timeout_ms=5000
[sip.backup_proxy.1]
C- enable=0
*factory=1
ipname=0.0.0.0
port=5060
[sip.backup_proxy.2]
C- enable=0
*factory=1
[suppserv]
enable=0
[suppserv.profile.1]
code_blind_xfer=*98*
code_call_clear=*52
code_call_cycle=!
code_consult_xfer=*99
dial_timeout=10
recall=!
termination=#
xfer_on_hangup=1
[tones]
busytone_seq=3
callwait1_seq=6
callwait2_seq=7
dialtone_seq=1
fastbusy_seq=4
ringback_seq=5
stutterd_seq=2
[tones.def.1]
amp1=6000
amp2=6000
amp3=0
amp4=0
freq1=350
freq2=440
freq3=0
freq4=0
name=dialtone
off_time=0
on_time=0
repeat=1
[tones.def.2]
amp1=6000
amp2=6000
amp3=0
amp4=0
freq1=350
freq2=440
freq3=0
freq4=0
name=stutter_dialtone
off_time=100
on_time=100
repeat=1
[tones.def.3]
amp1=5000
amp2=5000
amp3=0
amp4=0
freq1=480
freq2=620
freq3=0
freq4=0
name=busy
off_time=500
on_time=500
repeat=1
[tones.def.4]
amp1=5000
amp2=5000
amp3=0
amp4=0
freq1=480
freq2=620
freq3=0
freq4=0
name=fastbusy
off_time=300
on_time=300
repeat=1
[tones.def.5]
amp1=5000
amp2=5000
amp3=0
amp4=0
freq1=480
freq2=440
freq3=0
freq4=0
name=ringing
off_time=4000
on_time=2000
repeat=1
[tones.def.6]
amp1=32000
amp2=32000
amp3=32000
amp4=32000
freq1=1400
freq2=2060
freq3=2450
freq4=2600
name=offhook_warning
off_time=100
on_time=100
repeat=1
[tones.def.7]
amp1=5000
amp2=0
amp3=0
amp4=0
freq1=440
freq2=0
freq3=0
freq4=0
name=callwait
off_time=50
on_time=300
repeat=0
[tones.net]
disc=0
fail=0
ring=1
[tones.seq.1]
name=dial_seq
repeat=0
[tones.seq.1.tone.1]
duration=600000
play_tone=1
[tones.seq.1.tone.2]
duration=0
play_tone=6
[tones.seq.2]
name=stutter_dial_seq
repeat=0
[tones.seq.2.tone.1]
duration=2000
play_tone=2
[tones.seq.2.tone.2]
duration=598000
play_tone=1
[tones.seq.2.tone.3]
duration=0
play_tone=6
[tones.seq.3]
name=busy_seq
repeat=0
[tones.seq.3.tone.1]
duration=0
play_tone=3
[tones.seq.4]
name=fastbusy_seq
repeat=0
[tones.seq.4.tone.1]
duration=0
play_tone=4
[tones.seq.5]
name=ringing_seq
repeat=0
[tones.seq.5.tone.1]
duration=0
play_tone=5
[tones.seq.6]
name=callwait1_seq
repeat=0
[tones.seq.6.tone.1]
duration=350
play_tone=7
[tones.seq.7]
name=callwait2_seq
[tones.seq.7.tone.1]
duration=150
play_tone=7
[tones.seq.7.tone.2]
duration=150
play_tone=132
[tones.seq.7.tone.3]
duration=150
play_tone=7
[tones.seq.7.tone.4]
duration=150
play_tone=132
[tones.seq.7.tone.5]
duration=300
play_tone=7
[users.admin]
billing=0
logging=3
prompt=%u%p>
remote_access=1
C- timeout=1200 ; increase web admin timeout to 20 minutes instead of
the way-too-short 4 minutes!
*factory=240
[users.billing]
billing=1
logging=0
prompt=%u%p>
remote_access=1
timeout=0
[users.user]
billing=0
logging=3
prompt=%u%p>
remote_access=1
timeout=0
Total changed: 89 Unsaved: 0
*****************************************************************
* Additional notes
*****************************************************************
Things I need to work through still:
- Reduce echo much more
- Get all lines working correctly / config in asterisk
- currently only have line 1 working (due to early testing)
- Route certain lines to different extensions (use extensions
06,07,08,09 ... etc)
- Integrate incoming calls from vega to work however A@H deals with
incoming calls (use correct macro on incoming call), so that I can
configure behavior from AMP (Automated Attendants, etc).
Hopefully something here will help you. I hope to re-do / clean up much
of my config in the next couple weeks. Hopefully I will be able post
the results to the wiki (see Vegastream). HTH!
Good Luck!
Pete Doyle
--
Children of the Nations
http://www.cotni.org
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Issac
Simchayof
Sent: Saturday, June 10, 2006 6:23 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] FXO registration and VegaStream
Pete,
Thanks for the reply! If you don't mind I would love to take a look at
the script I am sure it will be great help.
Thanks,
Issac
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Peter
Doyle
Sent: Saturday, June 10, 2006 5:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] FXO registration and VegaStream
Hi Isaac,
I am a newbie to Asterisk (hoping to set up a system for my office) and
I have been struggling with the Vega 5010 (10 FXO) as well.
I've had the same problem as you, being able to call out, but not
receive calls. I just found a solution (for my setup atleast).
First off, I have the Vega set up according to some very basic
instructions from this list (for a different Vega) and from the getting
started setup guide from the Vegastream CD.
I have "enable registration" set (under SIP options in the Vega's
web
config), which makes the Vega register with Asterisk. I am currently
testing with only one line coming into port 1, which has an
"Authentication Number" (see PSTN options in Vega's web config) of
01
and a "Interface Number" of 06. Basically, the vega registers with
username 01, but sends the call to asterisk with 06@{vega's ip here} as
the address.
When I'd do a "sip debug" during an incoming call, I'd see
asterisk
responding with a "SIP/2.0 404 Not Found" error, causing the vega to
answer and immediately hang up.
I figured asterisk was looking for SIP user 06, so I added it, but I
still got 404's. Turns out I just needed an EXTENSION, 06. I can now
make calls and receive them, too. Of course, if you have multiple
incoming lines, you'd need extension 06, 07, 08 ... etc, since each port
has its own "Interface Number" (by default), to allow routing of calls
made to different lines.
I hope that helps some. If not, I can send my complete configs,
although I'm sure there's some other problems with them. Now, if only I
could get rid of the echo, I'd be a happy man!
Pete Doyle
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Issac
Simchayof
Sent: Friday, June 09, 2006 7:10 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] FXO registration and VegaStream
I am trying to configure a VegaStream 50 FXO to work with asterisk. The
problem that I am having is that the VegaStream does not support
incoming registration from asterisk. VegaStream only allows outbound
registration.
My question is does asterisk allow incoming registration from an FXO? If
yes how? Or better yet, has anybody been able to make the VegaStream FXO
work with asterisk? According to VegaStream they have many clients using
this combo but they haven't been very helpful otherwise.
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