[Asterisk-Users] Help connecting Avaya S8700 and Asterisk through
H.323 trunk
Pablo Chacón
pchacong at gmail.com
Mon Nov 28 13:54:19 MST 2005
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 <bweschke at gmail.com>:
> On 11/28/05, Pablo Chacón <pchacong at gmail.com> 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/
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