GrandStream 4108 Outbound Call not work

[itmnetworks]

Hello,

I have GrandStream 4108 and 3cx and one PSTN Line.

I setup my 4108 using DTMF to use on Brazil ( My Country ).

I receive all calls, but not create outbound calls, pabx delvery for me busy signal

If i use Stage Method = 2 and use phone number example: 991 40629422 the ippabx show for me dial tone and i need type again 40629422 to complete my call

if i use Stage Method = 1 and use phone number example: 991 40629422 the ippabx show buse tone for me.

please see my logs using Stage Method = 1 :

14:56:25.033 [CM503020]: Normal call termination. Reason: Forbidden
14:56:25.033 [CM503016]: Call(107): Attempt to reach <sip:[email protected]> failed. Reason: Forbidden
14:56:25.033 [CM503003]: Call(107): Call to sip:[email protected]:5060 has failed; Cause: 403 ; from IP:192.168.4.51:5060
14:56:25.001 [CM503025]: Call(107): Calling PSTNline:40629422@(Ln.10012@GrandStream4108)@[Dev:sip:[email protected]:5060;transport=udp]
14:56:25.001 [MS210002] C:107.2:Offer provided. Connection(transcoding mode): 192.168.4.91:7400(7401)
14:56:24.955 [CM503004]: Call(107): Route 1: PSTNline:40629422@(Ln.10012@GrandStream4108)@[Dev:sip:[email protected]:5060;transport=udp]
14:56:24.955 [MS210000] C:107.1:Offer received. RTP connection: 192.168.4.62:30146(30147)
14:56:24.955 [CM503010]: Making route(s) to <sip:[email protected]>
14:56:24.955 Remote SDP is set for legC:107.1
14:56:24.955 [CM505001]: Ext.1002: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Cisco-CP7960G/8.0] PBX contact: [sip:[email protected]:5060]
14:56:24.955 [CM503001]: Call(107): Incoming call from Ext.1002 to <sip:[email protected]>
14:56:24.955 [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.4.62:5060;branch=z9hG4bK7d68c194
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=udp>
To: <sip:[email protected]>
From: "1002"<sip:[email protected]>;tag=001aa1c6d183003722e0b2d9-0bb7450d
Call-ID: [email protected]
CSeq: 102 INVITE
Expires: 180
Accept: application/sdp
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, UPDATE
Proxy-Authorization: Digest username="1002",realm="3CXPhoneSystem",uri="sip:[email protected]",response="d9ef7880b19642971cc00b7a1953867e",nonce="414d535c0215dac875:23c609e189728d13ec02e07e512b5fe3",algorithm=MD5
Supported: replaces, join, norefersub
User-Agent: Cisco-CP7960G/8.0
Content-Length: 0
Remote-Party-ID: "1002" <sip:[email protected]>;party=calling;id-type=subscriber;privacy=off;screen=yes


My 3cx is: 3CXPSSB

On my GrandStream 4108 i have:

Product Model: GXW4108
Software Version: Program-- 1.3.1.6 Loader-- 1.1.3.4 Boot-- 1.1.3.2

FXO Lines Menu:

Wait for Dial-Tone(Y/N): ch1-8:N;
Stage Method(1/2): ch1-8:1;
Unconditional Call Forward to VOIP: UserID: ch1-8:10012+; SipSever: ch1-81; Sip Destination Port: ch1-8:5060++;

Channels Menu:

Dial Tone, RingBack Tone, Busy Tone and Reorder Tone = Brazil DTMF parameters
DTMF Methods: ch1-8:2; ( RFC2833 )
Local SIP Listen Port: ch1-8:5060++;
Round-robin and/or Flexible: rr:1-8;
Prefix to Specify Port(1 stage dialing method): 99

Dial Plan Menu:
PSTN Outgoing Call Dial Plan: { x+ }

On 3CX i create Outbound Rule:

Calls to numbers starting with (Prefix): 991
Route 1 = GrandStream 4108
Strip Digits = 3
Prepend =

I have one separate network and NIC to all IPPHONES and Gateway and 3cx ( i use 192.168.4.x isolated for voip on 3com switch )

How can resolve this ?

Please HELP me


Best Regards,
Rodrigo Araujo
ITMNETWORKS

 

[leejor]

14:56:25.033 [CM503016]: Call(107): Attempt to reach <sip:[email protected]> failed. Reason: Forbidden

You really want to be using one stage dialling. I would go over the settings in the 4108 again. Is there a log in the 4108 that would indicate why it is forbidding the call?

According to http: //www.3cx.com/voip-gateways/Grandstream-GXW-41044108/ NO LONGER AVAILABLE

the Wait for Dialtone should be set to "Yes". Don't know if that will make a difference.

 

[itmnetworks]

Hello all,

I see on my 4108 on Profile 1 SIP Server using network 192.168.2.x and the correct is 192.168.4.x

Now i create calls from 3cx to outside using GrandStream 4108 + PSTN line and using Stage 1 dialing.

Very thanks for all.

Best regards,
Rodrigo Araujo
ITMNETWORKS