<HTML dir=ltr><HEAD>
<META http-equiv=Content-Type content="text/html; charset=unicode">
<META content="MSHTML 6.00.2900.2995" name=GENERATOR></HEAD>
<BODY>
<DIV id=idOWAReplyText65343 dir=ltr>
<DIV dir=ltr><FONT face=Arial color=#000000 size=2><FONT face="Times New Roman">Hi,<BR><BR>I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also.<BR><BR>But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized information regarding asterisk is coming.<BR><BR><BR>I am putting my h323.conf and ooh323.conf<BR><BR>h323.conf<BR><BR><BR>; The NuFone Network's<BR>; Open H.323 driver configuration<BR>;<BR><BR>listenAddress=10.142.17.68<BR>listenPort=1720<BR>connectPort=1720<BR>;TCP<BR>tcpStart=10000<BR>tcpEnd=20000<BR><BR>;UDP<BR><BR>udpStart=10000<BR>udpEnd=20000<BR><BR>[general]<BR>port = 1720<BR>bindaddr = 0.0.0.0 ; this SHALL contain a single, valid IP address for this machine<BR>;tos=lowdelay<BR>;<BR>; You may specify a global default AMA flag for iaxtel calls. It must be<BR>; one of 'default', 'omit', 'billing', or 'documentation'. These flags<BR>; are used in the generation of call detail records.<BR>;<BR>;amaflags = default<BR>;<BR>; You may specify a default account for Call Detail Records in addition<BR>; to specifying on a per-user basis<BR>;<BR>;accountcode=lss0101<BR>;<BR>; You can fine tune codecs here using "allow" and "disallow" clauses<BR>; with specific codecs. Use "all" to represent all formats.<BR>;<BR>;disallow=all<BR>;allow=all ; turns on all installed codecs<BR>;disallow=g723.1 ; Hm... Proprietary, don't use it...<BR>;allow=gsm ; Always allow GSM, it's cool :)<BR>;allow=ulaw ; see doc/rtp-packetization for framing options<BR>;<BR>; User-Input Mode (DTMF)<BR>;<BR>; valid entries are: rfc2833, inband<BR>; default is rfc2833<BR>;dtmfmode=rfc2833<BR>;<BR>; Default RTP Payload to send RFC2833 DTMF on. This is used to<BR>; interoperate with broken gateways which cannot successfully<BR>; negotiate a RFC2833 payload type in the TerminalCapabilitySet.<BR>;<BR>; You may also specify on either a per-peer or per-user basis below.<BR>;dtmfcodec=101<BR>;<BR>; Set the gatekeeper<BR>; DISCOVER - Find the Gk address using multicast<BR>; DISABLE - Disable the use of a GK<BR>; <IP address> or <Host name> - The acutal IP address or hostname of your GK<BR>gatekeeper = DISABLE<BR><BR>;gatekeeper=10.142.17.68<BR>;<BR>;<BR>; Tell As terisk whether or not to accept Gatekeeper<BR>; routed calls or not. Normally this should always<BR>; be set to yes, unless you want to have finer control<BR>; over which users are allowed access to Asterisk.<BR>; Default: YES<BR>;<BR>;AllowGKRouted = yes<BR>;<BR>; When the channel works without gatekeeper, there is possible to<BR>; reject calls from anonymous (not listed in users) callers.<BR>; Default is to allow anonymous calls.<BR>;<BR>;AcceptAnonymous = yes<BR>;<BR>; Optionally you can determine a user by Source IP versus its H.323 alias.<BR>; Default behavour is to determine user by H.323 alias.<BR>;<BR>;UserByAlias=no<BR>;<BR>; Default context gets used in siutations where you are using<BR>; the GK routed model or no type=user was found. This gives you<BR>; the ability to either play an invalid message or to simply not<BR>; use user authentication at all.<BR>;<BR>;context=default<BR>;<BR>; Use this option to help Cisco (or other) gateways to setup backward voice<BR>; path to pass inband tones to calling user (see, for example,<BR>; </FONT><A href="https://webmail.wipro.com/exchweb/bin/redir.asp?URL=http://www.cisco.com/warp/public/788/voip/ringback.html" target=_blank><FONT face="Times New Roman">http://www.cisco.com/warp/public/788/voip/ringback.html</FONT></A><FONT face="Times New Roman">)<BR>;<BR>; Add PROGRESS information element to SETUP message sent on outbound calls<BR>; to notify about required backward voice path. Valid values are:<BR>; 0 - don't add PROGRESS information element (default);<BR>; 1 - call is not end-end ISDN, further call progress information can<BR>; possibly be available in-band;<BR>; 3 - origination address is non-ISDN (Cisco accepts this value only);<BR>; 8 - in-band information or an appropriate pattern is now available;<BR>;progress_setup = 3<BR>;<BR>; Add PROGRESS information element (IE) to ALERT message sent on incoming<BR>; calls to notify about required backwared voice path. Valid values are:<BR>; 0 - don't add PROGRESS IE ( default);<BR>; 8 - in-band information or an appropriate pattern is now available;<BR>;progress_alert = 8<BR>;<BR>; Generate PROGRESS message when H.323 audio path has established to create<BR>; backward audio path at other end of a call.<BR>;progress_audio = yes<BR>;<BR>; Specify how to inject non-standard information into H.323 messages. When<BR>; the channel receives messages with tunneled information, it automatically<BR>; enables the same option for all further outgoing messages independedly on<BR>; options has been set by the configuration. This behavior is required, for<BR>; example, for Cisco CallManager when Q.SIG tunneling is enabled for a<BR>; gateway where Asterisk lives.<BR>; The option can be used multiple times, one option per line.<BR>;tunneling=none ; Totally disable tunneling (default)<BR>;tunneling=cisco ; ; Enable Cisco-specific tunneling<BR>;tunneling=qsig ; Enable tunneling via Q.SIG messages<BR>;<BR>;------------------------------ JITTER BUFFER CONFIGURATION --------------------------<BR>; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a<BR> ; H323 channel. Defaults to "no". An enabled jitterbuffer will<BR> ; be used only if the sending side can create and the receiving<BR> &nbs p; ; side can not accept jitter. The H323 channel can accept jitter,<BR> ; thus an enabled jitterbuffer on the receive H323 side will only<BR> ; be used if the sending side can create jitter and jbforce is<BR> ; also set to yes.<BR><BR>; jbforce = no ; Forces the use of a ji tterbuffer on the receive side of a H323<BR> ; channel. Defaults to "no".<BR><BR>; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.<BR><BR>; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is<BR> ; resynchronized. Useful to improve the quality of the voice, with<BR> ; big jumps in/broken timestamps, usualy sent from exotic devices<BR> ; and programs. Defaults to 1000.<BR><BR>; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a H323<BR> ; channel. Two implementations are currenlty available - "fixed"<BR> ; (with size always equals to jbmax-size) and "adaptive" (with<BR> &nb sp; ; variable size, actually the new jb of IAX2). Defaults to fixed.<BR><BR>; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".<BR>;-----------------------------------------------------------------------------------<BR>;<BR>; H.323 Alias definitions<BR>;<BR>; Type 'h323' will register aliases to the endpoint<BR>; and Gatekeeper, if there is one.<BR>;<BR>; Example: if someone calls time@your.asterisk.box.com<BR>; Asterisk will send the call to the extension 'time'<BR>; in the context default<BR>;<BR>; [default]<BR>; exten => time,1,Answer<BR>; exten => time,2,Playback,current-time<BR>;<BR>; Keyword's 'prefix' and 'e164' are only make sense when<BR>; used with a gatekeeper. You can specify either a prefix<BR>; or E.164 this endpoint is responsible for termina ting.<BR>;<BR>; Example: The H.323 alias 'det-gw' will tell the gatekeeper<BR>; to route any call with the prefix 1248 to this alias. Keyword<BR>; e164 is used when you want to specifiy a full telephone<BR>; number. So a call to the number 18102341212 would be<BR>; routed to the H.323 alias 'time'.<BR>;<BR>;[time]<BR>;type=h323<BR>;e164=18102341212<BR>;context=default<BR>;<BR>;[det-gw]<BR>;type=h323<BR>;prefix=1248,1313<BR>;context=detroit<BR>;<BR>;<BR>; Inbound H.323 calls from BillyBob would land in the incoming<BR>; context with a maximum of 4 concurrent incoming calls<BR>;<BR>;<BR>; Note: If keyword 'incominglimit' are omitted Asterisk will not<BR>; enforce any maximum number of concurrent calls.<BR>;<BR>;[BillyBob]<BR>;type=user<BR>;host=192.168.1.1<BR>;context=incoming<BR>;incominglimit=4<BR>;h245Tunneling=no<BR>;<BR>;<BR>; Outbound H.323 call to Larry using SlowStart<BR>;<BR>;[Larry]<BR>;type=peer<BR>;host=192.168.2.1<BR>;fastStart=no<BR><BR>[preeti]<BR>type=peer<BR>c ontext=testing<BR>host=10.142.17.67<BR><BR>[radhamani]<BR>type=peer<BR>context=testing<BR>host=10.142.17.68<BR><BR><BR><BR><BR><BR>ooh323.conf<BR><BR><BR>; Objective System's H323 Configuration example for Asterisk<BR>; ooh323c driver configuration<BR>;<BR>; [general] section defines global parameters<BR>;<BR>; This is followed by profiles which can be of three types - user/peer/friend<BR>; Name of the user profile should match with the h323id of the user device.<BR>; For peer/friend profiles, host ip address must be provided as "dynamic" is<BR>; not supported as of now.<BR>;<BR>; Syntax for specifying a H323 device in extensions.conf is<BR>; For Registered peers/friends profiles:<BR>; OOH323/name where name is the name of the peer/friend profile.<BR>;<BR>; For unregistered H.323 phones:<BR>; OOH323/ip[:port] OR if gk is used OOH323/alias where alias can be any H323<BR>; &nbs p; alias<BR>;<BR>; For dialing into another asterisk peer at a specific exten<BR>; OOH323/exten/peer OR OOH323/exten@ip<BR>;<BR>; Domain name resolution is not yet supported.<BR>;<BR>; When a H.323 user calls into asterisk, his H323ID is matched with the profile<BR>; name and context is determined to route the call<BR>;<BR>; The channel driver will register all global aliases and aliases defined in<BR>; peer profiles with the gatekeeper, if one exists. So, that when someone<BR>; outside our pbx (non-user) calls an extension, gatekeeper will route that<BR>; call to our asterisk box, from where it will be routed as per dial plan.<BR><BR><BR>[general]<BR>;Define the asterisk server h323 endpoint<BR><BR><BR>listenAddress=0.0.0.0<BR>listenPort=1720<BR>connectPort=1720;wich port to sent traffic OUT<BR><BR><BR>;Configure T CP port range to be used by H.323<BR>;<BR>tcpStart=10000<BR>tcpEnd=20000<BR><BR>; Configure UDP port range to be used by H.323<BR>Note<BR>The port range used by RTP are configured from<BR><BR>; "rtp.conf"<BR>;<BR>udpStart=10000<BR>udpEnd=20000<BR><BR><BR>;gatekeeper=10.142.17.67<BR><BR><BR>;The port asterisk should listen for incoming H323 connections.<BR>;Default - 1720<BR>;port=1720<BR><BR>;The IP address, asterisk should listen on for incoming H323<BR>;connections<BR>;Default - 0.0.0.0: tries to find out local ip address on it's own<BR>bindaddr=0.0.0.0 <BR><BR>;Alias address for for asterisk server<BR>;Default - "Asterisk PBX"<BR>h323id=ObjSysAsterisk<BR>e164=100<BR><BR>;CallerID for the asterisk originated calls<BR>;Default - Same as h323id<BR>callerid=asterisk<BR><BR><BR>;This parameter indicates whether channel driver should register with<BR>;gatekeeper as a gateway or an endpoint.<BR>;Default - no<BR>;gateway=no<BR><BR>;Whether this asterisk server wi ll use gatekeeper.<BR>;Default - DISABLE<BR>;gatekeeper = DISCOVER<BR>;gatekeeper = a.b.c.d<BR>gatekeeper = DISABLE<BR><BR><BR>;Whether asterisk should use fast-start and tunneling for H323 connections.<BR>;Default - yes<BR>;faststart=yes<BR>;h245tunneling=yes<BR><BR>;Whether media wait for connect for fast start call<BR>;Default - no<BR>;mediawaitforconnect=no<BR><BR>;Location for H323 log file<BR>;Default - /var/log/asterisk/h323_log<BR>;logfile=/var/log/asterisk/h323_log<BR><BR><BR>;Following values apply to all users/peers/friends defined below, unless<BR>;overridden within their client definition<BR><BR>;Sets default context all clients will be placed in.<BR>;Default - default<BR>context=default<BR><BR>;Sets rtptimeout for all clients, unless overridden<BR>;Default - 60 seconds<BR>;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity<BR> &nb sp; ; when we're not on hold<BR><BR>;Type of Service<BR>;Default - none (lowdelay, thoughput, reliability, mincost, none)<BR>;tos=lowdelay<BR><BR>;amaflags = default<BR><BR>;The account code used by default for all clients.<BR>;accountcode=h3230101<BR><BR>;The codecs to be used for all clients.Only ulaw and gsm supported as of now.<BR>;Default - ulaw<BR>; ONLY ulaw, gsm, g729 and g7231 supported as of now<BR>disallow=all ;Note order of disallow/allow is important.<BR>allow=gsm<BR>allow=ulaw<BR><BR><BR>; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad<BR>; h245alphanumeric, h245signal.<BR>;Default - rfc 2833<BR>dtmfmode=rfc2833<BR><BR>; User/peer/friend definitions:<BR>; User config options Peer config options<BR>; ------------------ & nbsp; -------------------<BR>; context <BR>; disallow disallow<BR>; allow allow<BR>; accountcode accountcode<BR>; amaflags &nbs p; amaflags<BR>; dtmfmode dtmfmode<BR>; rtptimeout rtptimeout<BR>; ip<BR>; port<BR>; ; h323id<BR>; email<BR>; url<BR>; e164<BR>; &nbs p; <BR>;<BR><BR>;Define users here<BR>;Section header is extension<BR>[myuser1]<BR>type=user<BR>context=context1<BR>disallow=all<BR>allow=gsm<BR>allow=ulaw <BR><BR>[preeti]<BR>type=peer<BR>context=testing<BR>ip=10.142.17.67<BR>port=1720<BR>e164=101<BR><BR>[radhamani]<BR>type=peer<BR>context=testing<BR>ip=10.142.17.67<BR>port=1720<BR>e164=101<BR><BR>[mypeer1]<BR>type=peer<BR>context=context2<BR>ip=10.142.17.67 ; UPDATE with appropriate ip address<BR>port=1720 ; UPDATE with appropriate port<BR>e164=101<BR><BR><BR><BR>[myfriend1]<BR>type=friend<BR>context=default<BR>ip=10.142.17.67 ; UPDATE with appropriate ip address<BR>port=1720 ; UPDATE with appropriate port<BR>disallow=all<BR>allow=ulaw<BR>e164=12345<BR>rtptimeout=60<BR>dtmfmode=rfc2833<BR><BR><BR><BR><BR>Please help me understanding the problem and what error I am doing.<BR><BR><BR>Thanking you,<BR><BR>With regards,<BR>Preeta Pandey<FONT size=3> </FONT></FONT><BR></FONT></DIV></DIV><p class=MsoNormal><strong><span style='font-size:24.0pt;font-family:Webdings;
color:green'>P</span></strong><strong><span style='font-size:10.0pt;font-family:
"Palatino Linotype","serif";color:green'> Please do not print this email unless it is absolutely necessary. Spread environmental awareness.</span></strong><span style='font-family:"Arial","sans-serif"'><o:p></o:p></span></p>
<p> The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. </p>
<p>WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. </p>
<p>
www.wipro.com
</p>
</BODY></HTML>