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Patton 4114 Configuration


halalabu
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Hi Folks,

 

I'm trying to set up a Patton 4114 as an FXO gateway connected to POTS lines. I have my Snom set up and working internally, but when I try to dial through the gateway, i get the error "not in service, please check the number and dial again". The off-hook light does light up on the 4114 when I place a call, so it is communicating at some level. I feel like I'm missing something simple but I don't know what it is. Any help would be greatly appreciated. Just an FYI you'll see the 4114 is configured for DHCP, I have a reservation set on the router for its MAC address, so it is a static address. Thanks so much!

 

Hal

 

SNOM PBX, SNTP Server, DNS Server, are all located at 10.0.0.202 (also services.alabu.com)

Router/Gateway are located at 10.0.0.1

Patton 4114 is located at 10.0.0.203

 

Network Equipment:

Cisco SRP541W Router

Cisco SG300 Managed Switch

Mac Mini Server running OS X 10.7 Lion

 

Here is my Patton configuration:

 

####################################################################################
# Configuration file for SmartNode with: 1 FXO and 0 FXS
# With one ethernet port
# Configuration file name is alabu-fxo
#      Snome One IPPBX
####################################################################################
# LAN Interface Data 
# IP Address dhcp
# IP Mask 
####################################################################################
# WAN Interface Data 
# IP Address dhcp
# IP Mask 
####################################################################################
# Default Gateway 10.0.0.1
####################################################################################
# IP-PBX/SIP PROXY DATA 
# IP-PBX Ip Address /  Name / Domain services.alabu.com
# IP-PBX SIP Port 5060
####################################################################################
# LOCALIZATION DATA 
# Country United States
####################################################################################
# Supported firmware versions R5.x
# Tool version V2_1 06222011
# Adapted: Ernesto Casas 
# for comments, feedback or suggestions please send an email to: ecasas@patton.com
####################################################################################
####################################################################################
webserver port 80 language en
sntp-client server 10.0.0.202
dns-client server 10.0.0.202

system
ic voice 0
low-bitrate-codec g729
profile ppp default

# Call Progress tones based on country selection. Your selected country is: United States

profile call-progress-tone defaultDialtone
flush-play-list 
play 1 1000 350 -13 440 -13 
profile call-progress-tone defaultAlertingtone 
flush-play-list 
play 1 1000 440 -19 480 -19 
pause 2 3000 
profile call-progress-tone defaultBusytone 
flush-play-list 
play 1 500 480 -24 620 -24 
pause 2 500 
profile call-progress-tone defaultReleasetone 
flush-play-list 
play 1 250 480 -24 620 -24 
pause 2 250 
profile call-progress-tone defaultCongestiontone 
flush-play-list 
play 1 250 480 -24 620 -24 
pause 2 250


profile tone-set default
profile voip default
 codec 1 g711alaw64k rx-length 20 tx-length 20
 codec 2 g711ulaw64k rx-length 20 tx-length 20
 fax transmission 1 relay t38-udp
 fax transmission 2 bypass g711alaw64k
profile pstn default
profile sip default
profile aaa default
 method 1 local
 method 2 none

context ip router
interface IF_IP_LAN
    ipaddress dhcp 
   tcp adjust-mss rx mtu
   tcp adjust-mss tx mtu

interface IF_IP_WAN
    ipaddress dhcp 
   tcp adjust-mss rx mtu
   tcp adjust-mss tx mtu

context ip router
    route 0.0.0.0 0.0.0.0  10.0.0.1
context cs switch
    digit-collection timeout 2





 interface sip IF_SIP_1
       bind context sip-gateway GW_SIP_ALL_LINES
       no early-connect 
       early-disconnect
       route call dest-service HUNT_FXO
       remote services.alabu.com 5060
       address-translation outgoing-call request-uri user-part fix 10002 host-part to-header target-param none
       address-translation incoming-call called-e164 request-uri



interface fxo IF_FXO_1
         route call dest-interface IF_SIP_1
        loop-break-duration min 200 max 1000
         ring-number on-caller-id 
        mute-dialing
          disconnect-signal loop-break
        disconnect-signal busy-tone
        dial-after timeout 1



service hunt-group HUNT_FXO
cyclic
drop-cause normal-unspecified
   drop-cause no-circuit-channel-available
   drop-cause network-out-of-order
   drop-cause temporary-failure
   drop-cause switching-equipment-congestion
   drop-cause access-info-discarded
   drop-cause circuit-channel-not-available
   drop-cause resources-unavailable
   route call 1 dest-interface IF_FXO_1


context cs switch
 no shutdown



authentication-service AS_ALL_LINES


     username 10002 password 10002


location-services LS_ALL_LINES


    identity 10002


 port fxo 0 0
     use profile fxo us
    encapsulation cc-fxo
     bind interface IF_FXO_1 switch
     no shutdown

context sip-gateway GW_SIP_ALL_LINES
       interface LAN
       bind interface IF_IP_LAN context router port 5060
context sip-gateway GW_SIP_ALL_LINES
   no shutdown



port ethernet 0 0
 medium auto
 encapsulation ip
 bind interface IF_IP_LAN router
 no shutdown


################################################# END ##################################################

 

 

Here is my Snom Configuration.

Screen%20Shot%202012-05-02%20at%2012.43.29%20PM.png

 

Screen%20Shot%202012-05-02%20at%2012.45.43%20PM.png

 

Here is my Snom logs after trying to place one call. I replaced the phone number dialed with "X"s.

 

[5] 2012/05/02 12:46:18:	SIP Rx tls:10.0.0.101:4864:
INVITE sip:XXXXXXX@services.alabu.com;user=phone SIP/2.0
Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-67m9mx0j60lv;rport
From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv
To: <sip:XXXXXXX@services.alabu.com;user=phone>
Call-ID: 189b273cef94-lwpfxubupkbu
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:75902@10.0.0.101:4864;transport=tls;line=0ii3wcy6>;reg-id=1
X-Serialnumber: 0004134535DF
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom821/8.4.18
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Require: buttons
Content-Type: application/sdp
Content-Length: 518

v=0
o=root 893378661 893378661 IN IP4 10.0.0.101
s=call
c=IN IP4 10.0.0.101
t=0 0
m=audio 60964 RTP/AVP 9 0 8 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:7K0KKEDkZy8CGgPPJpamWwWhwLYVlW1Ovwh8/LyP
a=rtpmap:9 g722/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt
a=sendrecv
[1] 2012/05/02 12:46:18:	UDP: TOS could not be set
[1] 2012/05/02 12:46:18:	Last message repeated 2 times
[5] 2012/05/02 12:46:18:	SIP Rx tls:10.0.0.101:4864:
INVITE sip:XXXXXXX@services.alabu.com;user=phone SIP/2.0
Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-67m9mx0j60lv;rport
From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv
To: <sip:XXXXXXX@services.alabu.com;user=phone>
Call-ID: 189b273cef94-lwpfxubupkbu
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:75902@10.0.0.101:4864;transport=tls;line=0ii3wcy6>;reg-id=1
X-Serialnumber: 0004134535DF
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom821/8.4.18
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Require: buttons
Content-Type: application/sdp
Content-Length: 518

v=0
o=root 893378661 893378661 IN IP4 10.0.0.101
s=call
c=IN IP4 10.0.0.101
t=0 0
m=audio 60964 RTP/AVP 9 0 8 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:7K0KKEDkZy8CGgPPJpamWwWhwLYVlW1Ovwh8/LyP
a=rtpmap:9 g722/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt
a=sendrecv
[5] 2012/05/02 12:46:18:	SIP Tx tls:10.0.0.101:4864:
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-67m9mx0j60lv;rport=4864
From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv
To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625
Call-ID: 189b273cef94-lwpfxubupkbu
CSeq: 1 INVITE
Content-Length: 0

[5] 2012/05/02 12:46:18:	Dialplan "Alabu Skin Care": Match 518XXXXXXX@services.alabu.com to <sip:518XXXXXXX@10.0.0.203:5060;user=phone> on trunk FXO 0
[1] 2012/05/02 12:46:18:	UDP: TOS could not be set
[1] 2012/05/02 12:46:18:	Last message repeated 2 times
[5] 2012/05/02 12:46:18:	SIP Tx udp:10.0.0.203:5060:
INVITE sip:518XXXXXXX@10.0.0.203:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.202:5060;branch=z9hG4bK-4c308048256b66b1a6c4606b9073cb7a;rport
From: "John Doe" <sip:75902@services.alabu.com>;tag=843224298
To: <sip:518XXXXXXX@10.0.0.203:5060;user=phone>
Call-ID: 0a73e210@pbx
CSeq: 10349 INVITE
Max-Forwards: 70
Contact: <sip:75902@10.0.0.202:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.0.3981
P-Asserted-Identity: "FXO 0" <sip:75902@services.alabu.com>
Content-Type: application/sdp
Content-Length: 333

v=0
o=- 1035386651 1035386651 IN IP4 10.0.0.202
s=-
c=IN IP4 10.0.0.202
t=0 0
m=audio 58570 RTP/AVP 0 8 9 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[5] 2012/05/02 12:46:18:	SIP Tx tls:10.0.0.101:4864:
SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-67m9mx0j60lv;rport=4864
From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv
To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625
Call-ID: 189b273cef94-lwpfxubupkbu
CSeq: 1 INVITE
Contact: <sip:75902@10.0.0.202:5081;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.0.3981
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 429

v=0
o=- 1654920880 1654920880 IN IP4 10.0.0.202
s=-
c=IN IP4 10.0.0.202
t=0 0
m=audio 58242 RTP/AVP 0 8 9 2 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:aEgAj6nBDSI9lfuXjfuoFknhRrbgteBut8zJ8MW7
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[5] 2012/05/02 12:46:18:	SIP Rx udp:10.0.0.203:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.202:5060;branch=z9hG4bK-4c308048256b66b1a6c4606b9073cb7a;rport=5060;received=10.0.0.202
From: "John Doe" <sip:75902@services.alabu.com>;tag=843224298
To: <sip:518XXXXXXX@10.0.0.203:5060;user=phone>
Call-ID: 0a73e210@pbx
CSeq: 10349 INVITE
Server: Patton SN4114 JO EUI 00A0BA071F53 R6.1 2012-03-07 H323 SIP FXS FXO M5T SIP Stack/4.0.30.30
Content-Length: 0

[5] 2012/05/02 12:46:19:	SIP Rx tls:10.0.0.101:4864:
PRACK sip:75902@10.0.0.202:5081;transport=tls SIP/2.0
Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-92icuy4qau22;rport
From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv
To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625
Call-ID: 189b273cef94-lwpfxubupkbu
CSeq: 2 PRACK
Max-Forwards: 70
Contact: <sip:75902@10.0.0.101:4864;transport=tls;line=0ii3wcy6>;reg-id=1
RAck: 1 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Proxy-Require: buttons
Content-Length: 0

[5] 2012/05/02 12:46:19:	SIP Tx tls:10.0.0.101:4864:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-92icuy4qau22;rport=4864
From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv
To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625
Call-ID: 189b273cef94-lwpfxubupkbu
CSeq: 2 PRACK
Contact: <sip:75902@10.0.0.202:5081;transport=tls>
User-Agent: snom-PBX/2011-4.2.0.3981
Content-Length: 0

[5] 2012/05/02 12:46:22:	SIP Rx udp:10.0.0.203:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.202:5060;branch=z9hG4bK-4c308048256b66b1a6c4606b9073cb7a;rport=5060;received=10.0.0.202
From: "John Doe" <sip:75902@services.alabu.com>;tag=843224298
To: <sip:518XXXXXXX@10.0.0.203:5060;user=phone>;tag=985585227
Call-ID: 0a73e210@pbx
CSeq: 10349 INVITE
Contact: <sip:518XXXXXXX@10.0.0.203:5060>
Server: Patton SN4114 JO EUI 00A0BA071F53 R6.1 2012-03-07 H323 SIP FXS FXO M5T SIP Stack/4.0.30.30
Supported: replaces
Content-Type: application/sdp
Content-Length: 215

v=0
o=MxSIP 0 10 IN IP4 10.0.0.203
s=SIP Call
c=IN IP4 10.0.0.203
t=0 0
m=audio 4872 RTP/AVP 0 8 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
[5] 2012/05/02 12:46:22:	SIP Tx udp:10.0.0.203:5060:
ACK sip:518XXXXXXX@10.0.0.203:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.202:5060;branch=z9hG4bK-a9c127f03d3880e3dab8fef72925ef42;rport
From: "John Doe" <sip:75902@services.alabu.com>;tag=843224298
To: <sip:518XXXXXXX@10.0.0.203:5060;user=phone>;tag=985585227
Call-ID: 0a73e210@pbx
CSeq: 10349 ACK
Max-Forwards: 70
Contact: <sip:75902@10.0.0.202:5060;transport=udp>
P-Asserted-Identity: "FXO 0" <sip:75902@services.alabu.com>
Content-Length: 0

[5] 2012/05/02 12:46:22:	SIP Tx tls:10.0.0.101:4864:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-67m9mx0j60lv;rport=4864
From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv
To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625
Call-ID: 189b273cef94-lwpfxubupkbu
CSeq: 1 INVITE
Contact: <sip:75902@10.0.0.202:5081;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snom-PBX/2011-4.2.0.3981
Content-Type: application/sdp
Content-Length: 429

v=0
o=- 1654920880 1654920880 IN IP4 10.0.0.202
s=-
c=IN IP4 10.0.0.202
t=0 0
m=audio 58242 RTP/AVP 0 8 9 2 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:aEgAj6nBDSI9lfuXjfuoFknhRrbgteBut8zJ8MW7
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv
[5] 2012/05/02 12:46:22:	SIP Rx tls:10.0.0.101:4864:
ACK sip:75902@10.0.0.202:5081;transport=tls SIP/2.0
Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-rfv1on8evf71;rport
From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv
To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625
Call-ID: 189b273cef94-lwpfxubupkbu
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:75902@10.0.0.101:4864;transport=tls;line=0ii3wcy6>;reg-id=1
Proxy-Require: buttons
Content-Length: 0

[5] 2012/05/02 12:46:24:	SIP Rx tls:10.0.0.101:4864:
BYE sip:75902@10.0.0.202:5081;transport=tls SIP/2.0
Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-c742rpx0rfw5;rport
From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv
To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625
Call-ID: 189b273cef94-lwpfxubupkbu
CSeq: 3 BYE
Max-Forwards: 70
Contact: <sip:75902@10.0.0.101:4864;transport=tls;line=0ii3wcy6>;reg-id=1
User-Agent: snom821/8.4.18
RTP-RxStat: Total_Rx_Pkts=272,Rx_Pkts=265,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=267,Tx_Pkts=267,Remote_Tx_Pkts=250
Proxy-Require: buttons
Content-Length: 0

[5] 2012/05/02 12:46:24:	SIP Tx tls:10.0.0.101:4864:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 10.0.0.101:4864;branch=z9hG4bK-c742rpx0rfw5;rport=4864
From: "John Doe" <sip:75902@services.alabu.com>;tag=e4ohh51ccv
To: <sip:XXXXXXX@services.alabu.com;user=phone>;tag=1a34357625
Call-ID: 189b273cef94-lwpfxubupkbu
CSeq: 3 BYE
Contact: <sip:75902@10.0.0.202:5081;transport=tls>
User-Agent: snom-PBX/2011-4.2.0.3981
Content-Length: 0

[5] 2012/05/02 12:46:24:	SIP Tx udp:10.0.0.203:5060:
BYE sip:518XXXXXXX@10.0.0.203:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.202:5060;branch=z9hG4bK-50b51dc4e9ad2ad0b2059ae4e803fb20;rport
From: "John Doe" <sip:75902@services.alabu.com>;tag=843224298
To: <sip:518XXXXXXX@10.0.0.203:5060;user=phone>;tag=985585227
Call-ID: 0a73e210@pbx
CSeq: 10350 BYE
Max-Forwards: 70
Contact: <sip:75902@10.0.0.202:5060;transport=udp>
P-Asserted-Identity: "FXO 0" <sip:75902@services.alabu.com>
Content-Length: 0

[5] 2012/05/02 12:46:24:	SIP Rx udp:10.0.0.203:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.202:5060;branch=z9hG4bK-50b51dc4e9ad2ad0b2059ae4e803fb20;rport=5060;received=10.0.0.202
From: "John Doe" <sip:75902@services.alabu.com>;tag=843224298
To: <sip:518XXXXXXX@10.0.0.203:5060;user=phone>;tag=985585227
Call-ID: 0a73e210@pbx
CSeq: 10350 BYE
Server: Patton SN4114 JO EUI 00A0BA071F53 R6.1 2012-03-07 H323 SIP FXS FXO M5T SIP Stack/4.0.30.30
Content-Length: 0

[5] 2012/05/02 12:46:24:	BYE Response: Terminate 0a73e210@pbx

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The SIP log looks good/beautiful. To me it seems that the carrier is generating the prompt. Maybe there is a "1" missing at the beginning of the number? Try to connect a analog phone to the line, dial the number through the analog line and see if you get the same answer.

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I would get this error if the port didn't have dial tone. Check for dial tone on port 1 first...and I have some notes below:

 

You have the gateway set to dhcp and the server set to services.alabu.com 5060.

 

I would first give the adapter a static IP (if it doesn't have a dedicated DHCP address, something like 10.0.0.41 255.255.255.0), and I see you have the PBX pointing to an internal IP for the gateway so I assume the internal IP of the PBX is a 10.0.0.x IP? Point the gateway there.

 

You seem to have set up an identity:

 

interface sip IF_SIP_1

bind context sip-gateway GW_SIP_ALL_LINES

no early-connect

early-disconnect

route call dest-service HUNT_FXO

remote services.alabu.com 5060

address-translation outgoing-call request-uri user-part fix 10002 host-part to-header target-param none

address-translation incoming-call called-e164 request-uri

 

location-services LS_ALL_LINES

 

 

identity 10002

 

I don't know what this does but my setup didn't have it. If the above steps didn't work, try deleting the 10002 from both areas.

 

 

Everything else seems to be the same way I set my config file up.

 

Also, set the Explicitly list address for inbound traffic in the PBX trunk after you get it up and running, if you don't it leaves the system open for someone else to use try and use your PBX.

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