Patton 4112 - Silence on line.

[eihli]

We have a Patton 4112 set up using the generated config. When a call is placed with a 9 prefix, it is routed through the analog line connected to the Patton. It looks like things are connecting fine in the server activity log, but we don't hear any sound after the call is placed and the number we are dialing never rings.

I'll post what I have. Let me know what other info is needed to troubleshoot this. Thanks ahead of time for any help.

Here is the server log:
Silence....
17:03:49.102 Currently active calls - 1: [1264]
17:03:17.100 Currently active calls - 1: [1264]
17:02:47.080 Currently active calls - 1: [1264]
17:02:15.079 Currently active calls - 1: [1264]
17:01:45.077 Currently active calls - 1: [1264]
17:01:22.312 [CM503007]: Call(1264): Device joined: sip:[email protected]:5060
17:01:22.310 [CM503007]: Call(1264): Device joined: sip:[email protected]:5060
17:01:22.310 [CM503022]: Call(1264): Call recording is started, audio file: E:\3CX Recordings\305\[Security Office]_305-93041571_20110302170122(1264).wav
17:01:22.299 [CM505002]: Gateway:[Patton Backup] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Patton SN4112 JO EUI 00A0BA059D79 R5.6 2010-07-15 H323 SIP FXS FXO M5T SIP Stack/4.0.29.29] PBX contact: [sip:[email protected]:5060]
17:01:22.299 [CM503002]: Call(1264): Alerting sip:[email protected]:5060
17:01:18.300 [CM503025]: Call(1264): Calling PSTNline:13373041571@(Ln.10003@Patton Backup)@[Dev:sip:[email protected]:5060]
17:01:18.271 [CM503004]: Call(1264): Route 1: PSTNline:13373041571@(Ln.10003@Patton Backup)@[Dev:sip:[email protected]:5060,Dev:sip:[email protected]:5062]
17:01:18.261 [CM503010]: Making route(s) to <sip:[email protected]>
17:01:18.259 [CM505001]: Ext.305: Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA942-6.1.5(a)] PBX contact: [sip:[email protected]:5060]
17:01:18.250 [CM503001]: Call(1264): Incoming call from Ext.305 to <sip:[email protected]>

Here is the config:
#----------------------------------------------------------------#
# #
# SN4112/JO/EUI #
# R5.6 2010-07-15 H323 SIP FXS FXO #
# 2011-03-02T23:28:36 #
# SN/00A0BA059D79 #
# Generated configuration file #
# #
#----------------------------------------------------------------#

cli version 3.20
clock local default-offset +00:00
webserver port 80 language en

system

ic voice 0
low-bitrate-codec g729

profile ppp default

profile call-progress-tone defaultDialtone
play 1 1000 350 -13 440 -13

profile call-progress-tone defaultAlertingtone
play 1 1000 440 -19 480 -19
pause 2 3000

profile call-progress-tone defaultBusytone
play 1 500 480 -24 620 -24
pause 2 500

profile call-progress-tone defaultReleasetone
play 1 250 480 -24 620 -24
pause 2 250

profile call-progress-tone defaultCongestiontone
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

profile pstn default

profile sip default
no autonomous-transitioning

profile aaa default
method 1 local
method 2 none

context ip router

interface IF_IP_LAN
ipaddress 192.168.1.3 255.255.255.0
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

context cs switch
digit-collection timeout 2

interface sip IF_SIP_0
bind context sip-gateway GW_SIP_0
route call dest-interface IF_FXO_0
remote 192.168.1.106 5060
early-connect
early-disconnect
address-translation outgoing-call request-uri user-part fix 10003 host-part to-header target-param none

interface sip IF_SIP_1
bind context sip-gateway GW_SIP_1
route call dest-interface IF_FXO_1
remote 192.168.1.106 5060
early-connect
early-disconnect
address-translation outgoing-call request-uri user-part fix 10004 host-part to-header target-param none

interface fxo IF_FXO_0
route call dest-interface IF_SIP_0
loop-break-duration min 60 max 1000
disconnect-signal loop-break
disconnect-signal busy-tone
ring-number on-caller-id
dial-after timeout 1
mute-dialing

interface fxo IF_FXO_1
route call dest-interface IF_SIP_1
loop-break-duration min 60 max 1000
disconnect-signal loop-break
disconnect-signal busy-tone
ring-number on-caller-id
dial-after timeout 1
mute-dialing

context cs switch
no shutdown

authentication-service AS_ALL_LINES
realm 1 3CXPhoneSystem
username 10003 password xG9fBlsMRfgO35Hi6/G3jA== encrypted
username 10004 password fzoqYA6NNosfsq6Eef+QDw== encrypted

location-service LS_10003
domain 1 192.168.1.106

identity-group default

authentication outbound
authenticate 1 authentication-service AS_ALL_LINES username 10003

identity 10003

authentication outbound
authenticate 1 authentication-service AS_ALL_LINES

registration outbound
registrar 192.168.1.106 5060
lifetime 300
register auto

location-service LS_10004
domain 1 192.168.1.106

identity-group default

authentication outbound
authenticate 1 authentication-service AS_ALL_LINES username 10004

identity 10004

authentication outbound
authenticate 1 authentication-service AS_ALL_LINES

registration outbound
registrar 192.168.1.106 5060
lifetime 300
register auto

context sip-gateway GW_SIP_0

interface LAN
bind interface IF_IP_LAN context router port 5060

context sip-gateway GW_SIP_0
bind location-service LS_10003
no shutdown

context sip-gateway GW_SIP_1

interface LAN
bind interface IF_IP_LAN context router port 5062

context sip-gateway GW_SIP_1
bind location-service LS_10004
no shutdown

port ethernet 0 0
medium 10 half
encapsulation ip
bind interface IF_IP_LAN router
no shutdown

port fxo 0 0
use profile fxo us
encapsulation cc-fxo
bind interface IF_FXO_0 switch
no shutdown

port fxo 0 1
use profile fxo us
encapsulation cc-fxo
bind interface IF_FXO_1 switch
no shutdown

 

[ccomley]

Did you ever resolve this?

I'm setting up a 4112 today.

1) I got no dialling in or out - then I found the Patton was showing the line as "down" with the supplied calble. Swapped to a workign cable nicked form the back of a phone and it was fine.

But

2) Now I can dial in or out - but when the call connects, no audio passes. If this was a SIP call I'd suspect STUN settings or similar, but the Patton is on the same LAN segment as the 3CX...

 

[ccomley]

Just realised I never came back to this,

We *did* solve it - we realised there was no Default Gateway setting in the Patton and it wasn't getting one from the downloaded config. I realise it doesn't matter for 90% of situations but in our case the phones themselves are NOT on the same LAN range as the PBX and the Patton. Add a gateway to give the Patton a route to the phones and Bingo - good as you could ask for.