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