Outbound call using PSTN 3102

[leosa]

Hi,

Im having problems with outbound calls throught PSTN on a SPA3102 (as I browsed the forum, the boring SPA...).

Our setup includes two lines, one VoIP and one PSTN. We created the outbound rules this way:
- Numbers starting with "0" are routed on VoIP;
- Starting with "9" throught the PSTN.

Below is a dump from server status to help the experts assist me. As we can see, the call is routed correctly, the rules are respected but still indicates "User not found".

What am I missing?

Best regards,
Leo

---- Dump ----

18:26:07.237 Call::Terminate [CM503008]: Call(98): Call is terminated
18:26:07.177 Call::RouteFailed [CM503015]: Call(98): Attempt to reach "932244000"[sip:[email protected]] failed. Reason: Not Found
18:26:07.177 CallLeg:nFailure [CM503003]: Call(98): Call to sip:[email protected]:5060 has failed; Cause: 404 Not Found; from IP:10.0.1.197:5060
18:26:07.047 CallCtrl:nSelectRouteReq [CM503004]: Call(98): Calling: PSTNline:32244000@(Ln.10001@PSTN)@[Dev:sip:[email protected]:5061]
18:26:07.007 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to "932244000"[sip:[email protected]]
18:26:06.997 Extension:rintEndpointInfo [CM505001]: Ext.10: Device info: Device Identified: [Man: Counterpath;Mod: eyeBeam;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [eyeBeam release 1009r stamp 38169] Transport: [sip:10.0.1.198:5060]
18:26:06.987 CallCtrl:nIncomingCall [CM503001]: Call(98): Incoming call from Ext.10 to "932244000"[sip:[email protected]]

Update: I did what is suggested by renaissance on topic "SPA 3102 Outgoing" and the error changed to:

PSTN Line tab -> SIP settings -> SIP port : 5060
Line1 tab SIP settings -> SIP port : 5061

19:11:28.209 Call::Terminate [CM503008]: Call(8): Call is terminated
19:11:28.189 Call::RouteFailed [CM503015]: Call(8): Attempt to reach "932244000"[sip:[email protected]] failed. Reason: Not Registered
19:11:28.189 Call::RouteFailed [CM503015]: Call(8): Attempt to reach "932244000"[sip:[email protected]] failed. Reason: Not Registered
19:11:28.189 CallCtrl:nSelectRouteReq [CM503016]: Call(8): Target is not registered: PSTNline:32244000 dialed on (AnyLine@PSTN)
19:11:28.189 Line::Line [CM303003]: There are no available outbound lines on gateway PSTN at this time.
19:11:28.129 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to "932244000"[sip:[email protected]]
19:11:28.119 Extension:rintEndpointInfo [CM505001]: Ext.10: Device info: Device Identified: [Man: Counterpath;Mod: eyeBeam;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [eyeBeam release 1009r stamp 38169] Transport: [sip:10.0.1.198:5060]

 

[galal202]

Hello And welcome,
Change pstn port from 5060 to 5061 on server and spa3102
if it does not work reset to factory set and reconfig it again as the following

Hard ware Version must be updated to 5.1.7(GW)
From linksys site

dial **** to enter configuration menu
dial 73738# to reset to factory set
dial 1 to confirm
Hangup (will restart)

unplug pstn line now
dial **** to enter configuration menu
dial 110# to know your web IP

to enable web control
dial 7932# option 1 then 1 to save
plug pstn line now

click admin/advanced
Click ROUTER/WAN SETUP

Internet Connection Settings
Change Connection Type: to static ip


Static IP Settings
: set your network parameters Static IP:,NetMask:,Gateway

set dns provided from your ISP
set ntp to ntp server (option)

in pstn LINE

SIP Settings
leave SIP Port:5061 as it

Proxy and Registration
proxy:your server ip
register expires:300

Subscriber Information
set user id and password

Audio Configuration
Preferred Codec: g729

Dial Plans
Dial Plan 8: (S0<:xxxxx>)
which xxxxx= 3cx virtual number


PSTN-To-VoIP Gateway Setup
PSTN Ring Thru Line 1:No
PSTN CID Number Prefixption
PSTN CID For VoIP CID: yes
PSTN Caller Default DP: 8


FXO Timer Values (sec)
PSTN Answer Delay:From 4 to 6

PSTN Disconnect Detection
Detect PSTN Long Silence
PSTN Long Silence Duration:300

International Control
Line-In-Use Voltage: 10
SPA To PSTN Gain: 15
PSTN To SPA Gain: 15


In LINE1

Proxy and Registration
proxy
register expires:300

Subscriber Information
set user id and password

Audio Configuration
Preferred Codec: g729

Option to connect from remote to 3cx server

In VOICE/SIp
NAT Support Parameters

STUN Enable: yes
STUN Server: stun.fwdnet.net

In pstn line
Nat settings
NAT Mapping Enable: yes

In Line 1
Nat settings
NAT Mapping Enable: yes

 

[leosa]

Hi galal202,

My friend, thank you very much!

I followed your instructions and now we're able to dial with both outbound rules.

3CX is a great piece of software and its community is a huge part of it.

Thanks again,
Leo

 

[galal202]

Hello Leo,
You're welcome