PSTN doesn't Work

[cosmocid]

System as follow:

1. Version 1.3.2.0
2. 1.4.2_06
3. Windows XP, SP2
4. Zyxel P2002, Eusso UTG7104-O (4 Port FXO)
5. Pattern 2
6. No ring at Callee (Call destination) 's phone

Real World values:
ATA A: 8331234567, 192.168.0.226 (registered), g711
ATA B: 8337654321, 192.168.0.227 (registered), g711
OSS : 192.168.0.50
PBX : 192.168.0.50
FXO : 192.168.0.35, g711
PSTN : caller ids : 4843600 and 4843601

Problems:
ATA's can call each other and works well. I left all values in dial plans both in OSS and ONDO PBX as default. I try to call 905353490056, ONDO PBX strips 9 from it and tries to send as sip:05353490056@192.168.0.35:5060 but it doesn't work.

[Janet]

We don't know about the Eusso UTG7104-O.
Does it support one-step dialing?

Can it register with OnDO SIP Server?
If it can, can you call the gateway and dial PSTN number after it answers?

[cosmocid]

Eusso UTG7104-O supports one step dialing,

I can register with OnDO SIP Server (for example 833111222333), i dial from ATA's like 833111222333, i can trace it from sessions, i get PSTN line but can't call anywhere, i guess i arranged the dial/busy/congestion/disconnect tones for my country correctly. All ATA's and FXO are configured to use g711 codec, i've tried with SIP INFO,RFC 2833 and PCM, everytime same, i call other ATA, call is okay, i call FXO but can't call from ATA . I mean i've tried one step and two step dialing seperatly.

extra :
http://download.eusso.com:8080/Manual/VoIP/UTG7102&04_Manual.zip
http://download.eusso.com:8080/Manual/VoIP/ITG_Command_Ref.zip
http://download.eusso.com:8080/Manual/VoIP/SIP_Basic_Setting.pdf
http://download.eusso.com:8080/Manual/VoIP/SIP_ CLI_command.pdf
http://download.eusso.com:8080/Manual/VoIP/SIP_Quick_Installation_Guide.pdf

[cosmocid]

<---
INVITE sip:4843600@192.168.0.35 SIP/2.0
From: "302"<sip:302@192.168.0.50:5060> ;tag=1112611726988-24152206
To: <sip:4843600@192.168.0.35>
Call-ID: 20050404013529.222165595@192.168.0.50
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.0.50:5060 ;received=192.168.0.50 ;branch=z9hG4bK60f8bdd355472.1
Via: SIP/2.0/UDP 192.168.0.50:15062 ;branch=z9hG4bK83b310862acac ;rport=15062
Max-Forwards: 18
Contact: <sip:192.168.0.50:15062>
User-Agent: Brekeke OnDO PBX (1.3.2.0/40)
Allow: INVITE
Allow: ACK
Allow: BYE
Allow: CANCEL
Allow: INFO
Allow: MESSAGE
Record-Route: <sip:192.168.0.50:5060;lr>
Content-Type: application/SDP
Content-Length:178

v=0
o=SYSTEM 34 0 IN IP4 192.168.0.50
s=-
c=IN IP4 192.168.0.50
t=0 0
m=audio 12024 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16


--->
SIP/2.0 606 Not Acceptable
From: "302"<sip:302@192.168.0.50:5060> ;tag=1112611726988-24152206
To: <sip:4843600@192.168.0.35> ;tag=3a4ffb68-7962a1df4
Call-ID: 20050404013529.222165595@192.168.0.50
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.0.50:5060 ;received=192.168.0.50 ;branch=z9hG4bK60f8bdd355472.1
Via: SIP/2.0/UDP 192.168.0.50:15062 ;branch=z9hG4bK83b310862acac ;rport=15062
Content-Type: application/SDP
Content-Length:137

v=0
o=From:sip:192.168.0.35 4213434 1686232756 IN IP4 192.168.0.35
s=eitg SIP call
c=IN IP4 192.168.0.35
t=0 0
m=audio 0 RTP/AVP 0


<---
ACK sip:4843600@192.168.0.35 SIP/2.0
From: "302"<sip:302@192.168.0.50:5060> ;tag=1112611726988-24152206
To: <sip:4843600@192.168.0.35> ;tag=3a4ffb68-7962a1df4
Call-ID: 20050404013529.222165595@192.168.0.50
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.0.50:5060 ;branch=z9hG4bK60f8bdd355472.1
Via: SIP/2.0/UDP 192.168.0.50:15062 ;branch=z9hG4bK83b310862acac ;rport=15062
Max-Forwards: 18
Contact: <sip:192.168.0.50:15062>
User-Agent: Brekeke OnDO PBX (1.3.2.0/40)
Record-Route: <sip:192.168.0.50:5060;lr>
Content-Length:0



--->
REGISTER sip:192.168.0.50:5060 SIP/2.0
From: <sip:301@192.168.0.50:5060> ;tag=3a4ff9dc-18d6a5a2e
To: <sip:301@192.168.0.50>
Call-ID: c0a80023-13c4-3a4ff977-1f6-5973
CSeq: 6 REGISTER
Via: SIP/2.0/UDP 192.168.0.35:5060 ;branch=z9hG4bK-3a4ffb6f-7b130-491d
Max-Forwards: 70
Contact: <sip:301@192.168.0.35:5060>
Expires: 100
Content-Length:0



<---
SIP/2.0 200 OK
From: <sip:301@192.168.0.50:5060> ;tag=3a4ff9dc-18d6a5a2e
To: <sip:301@192.168.0.50> ;tag=1112611734750-12115695
Call-ID: c0a80023-13c4-3a4ff977-1f6-5973
CSeq: 6 REGISTER
Via: SIP/2.0/UDP 192.168.0.35:5060 ;branch=z9hG4bK-3a4ffb6f-7b130-491d
Contact: <sip:301@192.168.0.35:5060>;expires=100;q=1.0
Server: Brekeke OnDO SIP Server (1.2.4.1/101)
Content-Length:0

SIP register phone 301 6 status: SUCCESS

[Janet]

[cosmocid]

here's my console window, so we're sure that coding is set to g711

what can be else ?

--------------------------------------------------------------------------------------
Console>show port 0


Configuration for PORT (TCID) 0:

Pref Voice coding profile: 8
Pref Fax coding profile: 5

Telephony Interface Configuration:
Gain (RX,TX) = (*0 , $0 )
Call Progress Tone Detection Control = Always on
Fax Tone Detection Control = Always on
Call Progress Tone Detection Configuration = Default

Signaling Protocol: FXO Loop Start
Answer After: 1 rings
Caller ID Detection: OFF
Valid Call Timer: 4500 msec
Use user-specified CID info: No

Dial Out Parameters:
Out Type: #tone

Dial In Parameters:
Plar addr: None

Call Timing Parameters:
Call Limit: forever
Answer Wait: forever

Tone Table:
Default

Caller ID Parameters:
Caller ID Number: 201
Caller ID Name: ozan
Call forward addresses:

Console>show coding 8


Configuration for coding profile id 8:
Tx Coding = G.711 MU
Rx Coding = G.711 MU
Coding profile for voice
DTMF Relay = Disabled
--------------------------------------------------------------------------------------

[Janet]

Are there any error logs regarding the call on the gateway?
I don't know why the gateway returned "Not Acceptable".
Can you ask the gateway maker about it?

[bilchenko1]

Hello.

1. UA1 call to PSTN.
2. Called not answer.
3. UA1 hung UP.
4. One session Not delete!!!

Any Idea?
========================================
Call sessions:
========================================
Session ID : 8056
from-url sip:1011@hcsserver.wicklow.herbst.ie
from-ip 172.16.100.15:5060
from-if 192.168.0.102:5060
to-url sip:90879113563@hcsserver.wicklow.herbst.ie
to-ip 127.0.0.1:15060
to-if 192.168.0.102:5060
call-id 0299C52F-4DF3-4AC0-8368-DF1C83970070@172.16.100.15
rule <share> & to PBX
listen-port 5060
sip-packet-total 3
sip-packet-stacked 0
phase Ringing
time-inviting Mon Apr 18 17:45:06 BST 2005
rtp-relay off


Session ID : 8057
from-url sip:192.168.0.102:15062
from-ip 192.168.0.102:15062
from-if 192.168.0.102:5060
to-url sip:media1*268@192.168.0.102
to-ip 127.0.0.1:25060
to-if 192.168.0.102:5060
call-id 20050418084417.6532707634@192.168.0.102
rule <share> & to MediaServer
listen-port 5060
sip-packet-total 5
sip-packet-stacked 0
phase Talking
time-inviting Mon Apr 18 17:45:06 BST 2005
time-talking Mon Apr 18 17:45:06 BST 2005
length-talking 00:00:16
rtp-relay off



from-url sip:192.168.0.102:15062
from-ip 192.168.0.102:15062
from-if 192.168.0.102:5060
to-url sip:media2*268@192.168.0.102
to-ip 127.0.0.1:25060
to-if 192.168.0.102:5060
call-id 20050418084417.6633323834@192.168.0.102
rule <share> & to MediaServer
listen-port 5060
sip-packet-total 10
sip-packet-stacked 0
phase Talking
time-inviting Mon Apr 18 17:45:06 BST 2005
time-talking Mon Apr 18 17:45:06 BST 2005
length-talking 00:00:27
rtp-relay off



Session ID : 8059
from-url sip:1011@192.168.0.102:5060
from-ip 192.168.0.102:15062
from-if 192.168.0.102:5060
to-url sip:0879113563@192.168.0.79
to-ip 192.168.0.79
to-if 192.168.0.102:5060
call-id 20050418084417.6720018811@192.168.0.102
rule <outbound> & <share> & from PBX 2
listen-port 5060
sip-packet-total 9
sip-packet-stacked 0
phase Ringing
time-inviting Mon Apr 18 17:45:06 BST 2005
rtp-relay off

========================================
Session not deleted:
========================================
Session ID : 8059
from-url sip:1011@192.168.0.102:5060
from-ip 192.168.0.102:15062
from-if 192.168.0.102:5060
to-url sip:0879113563@192.168.0.79
to-ip 192.168.0.79
to-if 192.168.0.102:5060
call-id 20050418084417.6720018811@192.168.0.102
rule <outbound> & <share> & from PBX 2
listen-port 5060
sip-packet-total 3
sip-packet-stacked 0
phase Ringing
time-inviting Mon Apr 18 17:45:06 BST 2005
rtp-relay off

[jane]

Hi bilchenko1.

i think you should create a new topic for your question.
anyway you should read this.

http://www.brekeke-sip.com/bbs/viewtopic.php?t=5

[bilchenko1]

Hi.

jane > The title of the topic matches my problem exactly

OSS: 1.2.7.4/101
PBX: OnDO PBX SmallOffice Edition (1.3.2.0)
Java: 1.4.2_03
OS: Windows XP Professional
UA: XLite

Pattern 2.

ARS
out:
Matching patterns Deploy patterns
To sip:9(.+)@ To sip:$1@192.168.0.79

GateWay: MultiTech MVP810ST
a)
1. UA1 call to PSTN.
2. Called not answer.
3. UA1 hung UP.
4. One session Not delete!!!
b)
1. UA1 call to PSTN.
2. Called answer.
3. UA1 hung UP.
4. All session delete. OK

[bilchenko1]

Hi all.

The 5.06.AN firmware should resolve that problem.
GateWay MultiTech MVP810ST work correctly now.

Thanks