Hi BJ Weschke, thanks but unfortunately Ip address is the correct one.
Do you have S8700 with Asterisk working? using oh323 channel??
Maybe can help you my S8700 configuration...
My S8700 configuration is:
-------------------------------------------------------------------------------
list ip-interfaces clan
IP INTERFACES
Num
Skts Net
ON Slot Code Sfx Node Name/ Subnet Mask Gateway Address Warn Rgn VLAN
IP-Address
-- ---- ---- --- --------------- --------------- --------------- ---- --- ----
.............
y 04A04 TN799 D CLND04A04 255.255.255.0 10.64.108.254 400 2 n
10.64.108.132
-------------------------------------------------------------------------------
change signaling-group 23 Page 1 of 5
SIGNALING GROUP
Group Number: 23 Group Type: h.323
Remote Office? n Max number of NCA TSC: 0
SBS? n Max number of CA TSC: 0
IP Video? n Trunk Group for NCA TSC:
Trunk Group for Channel Selection: 23
Supplementary Service Protocol: a Network Call Transfer? n
T303 Timer(sec): 10
Near-end Node Name: CLND04A04 Far-end Node Name: ASTERISK
Near-end Listen Port: 1720 Far-end Listen Port: 1720
Far-end Network Region: 2
LRQ Required? n Calls Share IP Signaling Connection? n
RRQ Required? n
Media Encryption? n Bypass If IP Threshold Exceeded? n
DTMF over IP: out-of-band Direct IP-IP Audio Connections? n
IP Audio Hairpinning? n
Interworking Message: PROGress
DCP/Analog Bearer Capability: 3.1kHz
---------------------------------------------------------------------------------
display trunk-group 23 Page 1 of 19
TRUNK GROUP
Group Number: 23 Group Type: isdn CDR Reports: y
Group Name: ASTERISK-H323 COR: 1 TN: 1 TAC: #23
Direction: two-way Outgoing Display? n Carrier Medium: IP
Dial Access? y Busy Threshold: 255 Night Service:
Queue Length: 0
Service Type: tie Auth Code? n TestCall ITC: rest
Far End Test Line No:
TestCall BCC: 4
TRUNK PARAMETERS
Codeset to Send Display: 0 Codeset to Send National IEs: 6
Max Message Size to Send: 260 Charge Advice: none
Supplementary Service Protocol: a Digit Handling (in/out): enbloc/enbloc
Trunk Hunt: cyclical QSIG Value-Added? n
Digital Loss Group: 18
Incoming Calling Number - Delete: Insert: Format:
Bit Rate: 1200 Synchronization: async Duplex: full
Disconnect Supervision - In? y Out? n
Answer Supervision Timeout: 0
--------------------------------------------------------------------------------
display trunk-group 23 Page 2 of 19
TRUNK FEATURES
ACA Assignment? n Measured: none Wideband Support? n
Internal Alert? n Maintenance Tests? y
Data Restriction? n NCA-TSC Trunk Member:
Send Name: y Send Calling Number: y
Used for DCS? n
Suppress # Outpulsing? n Format: public
Outgoing Channel ID Encoding: preferred UUI IE Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Send Connected Number: n
Network Call Redirection: none Hold/Unhold Notifications? n
Send UUI IE? y Modify Tandem Calling Number? n
Send UCID? n
Send Codeset 6/7 LAI IE? y
SBS? n Network (Japan) Needs Connect Before Disconnect? n
--------------------------------------------------------------------------------
display ip-network-region 2 Page 1 of 19
IP NETWORK REGION
Region: 2
Location: 1 Authoritative Domain:
Name: ** Pool LR VoIP **
Intra-region IP-IP Direct Audio: yes
MEDIA PARAMETERS Inter-region IP-IP Direct Audio: yes
Codec Set: 1 IP Audio Hairpinning? y
UDP Port Min: 2048
UDP Port Max: 20001 RTCP Reporting Enabled? n
DIFFSERV/TOS PARAMETERS RTCP MONITOR SERVER PARAMETERS
Call Control PHB Value: 46
Audio PHB Value: 46
Video PHB Value: 26
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 6
Audio 802.1p Priority: 6 AUDIO RESOURCE RESERVATION PARAMETERS
H.323 IP ENDPOINTS RSVP Enabled? n
H.323 Link Bounce Recovery? y
Idle Traffic Interval (sec): 20
Keep-Alive Interval (sec): 5
Keep-Alive Count: 5
----------------------------------------------------------------------------------
display ip-codec-set 1 Page 1 of 2
IP Codec Set
Codec Set: 1
Audio Silence Frames Packet
Codec Suppression Per Pkt Size(ms)
1: G.711A n 2 20
2: G.711MU n 2 20
3:
4:
5:
6:
7:
Media Encryption
1: none
2:
3:
2005/11/28, BJ Weschke <[EMAIL PROTECTED]>:
> On 11/28/05, Pablo Chacón <[EMAIL PROTECTED]> wrote:
> > Hi I'm trying to connect Avaya S8700 and Asterisk through H323 trunk
> > (using channel oh323).
> > I can make calls from S8700 H323 extension to Asterisk SIP phone using
> > G711a codec but when I try to make a call from SIP phone to S8700
> > extension I listen one ringing tone and the call is dropped.
> > Can anybody help me???
> >
>
> I've had greater success increasing the number of frames in an RTP
> packet when dealing with the med pro resources on the S8700.
>
> Also, make sure you're sending the call to the IP that is bound to
> the CLAN board that also has the signaling group you're trying to call
> into bound to it. With the connection refused here it seems like you
> might be trying to send the call to the IP of the med pro board
> instead of a CLAN board.
>
> BJ
>
>
> --
> Bird's The Word Technologies, Inc.
> http://www.btwtech.com/
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
