3241 ISDN Gateway--DTMF Relay Type & local vs. national signaling

Question 1:
Is there a way to configure the 3241 so that it will flag local vs. national call types on the d-channel appropriately?  The local TSP sees all calls flagged national..  They do accept both 7 and 10 digit formats for local calls so we set dialing rules to add the local area code (without a +1) to 7-digit numbers and that seems to work OK.
Querstion 2:
Does anyone know what type of DTMF relay is being used by the Cisco-acquired Tandberg/Codian 3241 ISDN Gateway?  I'm pretty sure the DTMF relay type is not configurable in that box.  Does this fall under the alphanumeric H.245 relay type? 
I have a customer that, for outbound long distance calls (Primarily ISDN voice, but some ISDN video), requires their ISP to provide a DTMF-based billing code service.  We've worked with the local TSP to verifiy what they are seeing on the switch and it looks like all digits are passing appropriately (although I also don't see any way to tell the Cisco 3241 GW to flag the call as Local vs. National
The PRI itself is provided through one local LEC and handed to a secondary TSP to provide the DTMF/billing solution.
The PRI has been works fine for local calls, but long distance (national) calls route throuth the primary provider and then pass to the secondary provider.  We've worked with the local TSP to verifiy what they are seeing on the switch and it looks like all digits are passing appropriately.  Logs from the ISDN gateway do not show opening any audio or h.245 channel, no 'biiling tone-prompt' is heard, and we cannot access/enable the touch tone menu on the endpoint during the time the ISDN GW is attempting to set up the call.  I suspect that either the billing code service is not turned up yet, but I'm hoping its is not a H323/H.245 and/or endpoint cpability that is preventing this from working.
In order to troubleshoot with the service provider I assume I'll need to know what DTMF type we are sending, I am hoping they could also tell me by debuggin the switch for any signaling they are receiving from us.
If anyone has any experience setting up similar service on CUCM i'd love to hear what was required on that system for DTMF relay, and if there is anything else I may need to ask the TSP
Thanks!

DTMF relay for H323 is handled with 3 commands.
cisco-rtp Cisco Proprietary RTP
h245-alphanumeric DTMF Relay via H245 Alphanumeric IE
h245-signal DTMF Relay via H245 Signal IE
Your best bet is h245-alphanumeric. The dtmf-relay rtp-nte command is used for SIP dial-peers.
The DTMF relay feature transports DTMF tones generated after call establishment out of band using either a standard H.323 out-of-band method and a proprietary RTP-based mechanism, or for SIP calls, an NTE RTP packet.
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a00800d62d2.shtml#Solution4
Please rate any helpful posts
Thanks
Fred

Similar Messages

  • Connect to SAP gateway failed Connect_PM  TYPE=B

    Hi experts,
    I get the following error when i ping through JCO destinations on portals.
    com.sap.mw.jco.JCO$Exception: (102) RFC_ERROR_COMMUNICATION: Connect to SAP gateway failed Connect_PM  TYPE=B MSHOST=imtsapdev01 GROUP=CLUST R3NAME=DMI MSSERV=sapmsDMI PCS=1 LOCATION    CPIC (TCP/IP) on local host with Unicode ERROR       partner '127.0.0.1:3310' not reached TIME        Tue Jan 27 16:39:54 2009 RELEASE     700 COMPONENT   NI (network interface) VERSION     38 RC          -10 MODULE      nixxi.cpp LINE        2823 DETAIL      NiPConnect2 SYSTEM CALL connect ERRNO       10061 ERRNO TEXT  WSAECONNREFUSED: Connection refused COUNTER     1
    However my JCO destinations were working perfectly until recently. I checked on the logon group. I have one called CLUST, which i use for this purpose.
    I have also checked both service files and they have the following for the ECC server im connecting to.
    3610 because its system number 10 i am connecting to.
    sapmsDMI 3610/tcp
    From the JCE destinations screen i cannot even ping the server. I can though when i log on to the server through remote desktop.
    thanks and regards
    dilanke
    as a reply to nishants posts.
    goto rz70
    give hostname -->
    service --> sapgw00 (sapgw(instance number)
    execute it
    click on yes
    check rfc call (success or failed)
    I have done this. i left hostname as "blank" gave service as "sapgw10"
    The following is what i  got as the output.
        0: IMTSAPDEV01_DMI_10                        : Execute program: SLDRFC
        0: IMTSAPDEV01_DMI_10                        : Execute program: SLDMSGSRV
        0: IMTSAPDEV01_DMI_10                        : Execute program: SLDIPSERV
        0: IMTSAPDEV01_DMI_10                        : Execute program: SLDINSTSP
        0: IMTSAPDEV01_DMI_10                        : Execute program: SLDINSTSC
        0: IMTSAPDEV01_DMI_10                        : Execute program: SLDGWSRV
        0: IMTSAPDEV01_DMI_10                        : Execute program: SLDDBSYS
        0: IMTSAPDEV01_DMI_10                        : Execute program: SLDCOMPSYS
        0: IMTSAPDEV01_DMI_10                        : Execute program: SLDCLIENT
        0: IMTSAPDEV01_DMI_10                        : Execute program: SLDBCSYS
        0: IMTSAPDEV01_DMI_10                        : Execute program: SLDAPPL_SERV
        0: IMTSAPDEV01_DMI_10                        : Execute program: SLDASSOC
        0: IMTSAPDEV01_DMI_10                        : Collection of SLD data finished
        0: IMTSAPDEV01_DMI_10                        : Data collected successfully
        0: IMTSAPDEV01_DMI_10                        : RFC data prepared
        0: IMTSAPDEV01_DMI_10                        : Used RFC destination: SLD_NUC
        0: IMTSAPDEV01_DMI_10                        : RFC call failed: Error opening an RFC connection.
        0: IMTSAPDEV01_DMI_10                        : Existing periodic jobs removed. Number: 1
        0: IMTSAPDEV01_DMI_10                        : Program scheduled: 20090127 184357
        1: IMTSAPDEV01_DMI_10                        : Event-controlled job already exists; scheduling not necessary
    so i guess its not successful. when i look at rz70 of a server where this works correctly its not succsessful either
    if it is success then technical system name will be in sld
    to restart sld --> /sld --> administration --> stop server --> restart server
    or check in VA --> SLD data supplier value --> Runtime
    I checked this and i do not see aa problem. you have configuration status as valid and send reult as Success
    Services file is locted on OS level of R3 system--> Goto --> Start --> Run -> (type)Drivers(enter) --> etc -> (here is services file)
    check ur r/3 server entry is there or not,
    I have the followng in the service file for the server
    sapmsDMI 3610/tcp
    if not add it

    Hi Experts,
    when ilook at the remote gateway properties in SMGW this is what i see on the R/3 server. ive looked at my other system which i can connect to and there isnt much of a difference in the parameters. How can test this to see if it works?
                                                                                    Details of remote gateway                                                                               
    entry                         = 0                                            
      state                         = CONNECTED                                    
      local                         = 1                                            
      system type                   = REMOTE_GATEWAY                               
      client                        = FALSE                                        
      wait for frag write           = 0                                            
      suspended                     = 0                                            
      read/write socket             = 4                                            
      HANDLE                        = 4                                            
      TIME                          = Sun Oct 19 18:54:14 2008                     
      SOCKET                        = 1352                                         
      STAT                          = NI_CONNECTED                                 
      TYPE                          = STREAM IPv4                                  
      OUT                           = 48225 messages 48225 bytes                   
      IN                            = 1 messages 64 bytes                          
      LOCAL                         = 127.0.0.1:3310                               
      REMOTE                        = 127.0.0.1:1297                               
      OPTIONS                       =                                              
      connect                       = Sun Oct 19 18:52:17 2008                     
      last request                  = Thu Jan 29 13:09:58 2009                     
      conversation no               = 0                                            
    thanks and regards,
    dilanke

  • Connect to SAP gateway failed Connect_PM TYPE=A

    In past 4-5 days, some of messages are not processed in our XI production server.
    I got the following error in the falied messages:
    Seem to be some parameter need to be update for sap gateway.
    <SAP:AdditionalText>com.sap.aii.af.ra.ms.api.DeliveryException: RfcAdapter: receiver channel has static errors: can not instantiate RfcPool caused by: com.sap.aii.af.rfc.RfcAdapterException: error initializing RfcClientPool:com.sap.aii.af.rfc.core.repository.RfcRepositoryException: can not connect to destination system due to: com.sap.mw.jco.JCO$Exception: (102) RFC_ERROR_COMMUNICATION: Connect to SAP gateway failed Connect_PM TYPE=A ASHOST=dbcinpr SYSNR=15 GWHOST=dbcinpr GWSERV=sapgw15 PCS=1 LOCATION CPIC (TCP/IP) on local host with Unicode ERROR connection to host dbcinpr, service sapgw15 timed out TIME Sun Oct 21 09:09:50 2007 RELEASE 640 COMPONENT NI (network interface) VERSION 37 RC -12 MODULE nixxi_r_mt.cpp LINE 1067 DETAIL NiPConnect COUNTER 1</SAP:AdditionalText>
      <SAP:ApplicationFaultMessage namespace="" />
    Pls let us know if it is parameter related prolem or some other issue.
    Regards
    Amar

    Hi Amar!
    Test the status of the RFC destinations in transaction SM59 of XI and R/3 systems.
    Also check this thread, I think they might be helpful:
    Communication channel not working
    RFC Receiver Adapter Error
    JCO Destination creation error
    RFC_ERROR_COMMUNICATION - time out
    Reg Jco connections
    RFC Adaptive Model,JCO destinations error
    Regards,
    Matias
    PS: please award points if helpful.

  • DTMF relay

    Hello,
    I have problem relaying DTMF signals from MCU to H.320 GW.
    I was trying to invite an IVR number via H.320 GW. I was able to call the IVR number. Then the IVR asks for extension number. I provided the number. However, this information was not relayed by the MCU.
    Thanks

    Just go to your SIP dial-peers:
    R01#conf t
    R01#dial-peer voice XX voip    << XX is you SIP dial-peer
    R01(config-dial-peer)#dtmf-relay ?
      cisco-rtp          Cisco Proprietary RTP
      h245-alphanumeric  DTMF Relay via H245 Alphanumeric IE
      h245-signal        DTMF Relay via H245 Signal IE
      rtp-nte            RTP Named Telephone Event RFC 2833
      sip-kpml           DTMF Relay via KPML over SIP SUBCRIBE/NOTIFY
      sip-notify         DTMF Relay via SIP NOTIFY messages
    dial-peer voice 555 voip
     destination-pattern 55555
     session protocol sipv2
     session target ipv4:1.1.1.1
     dtmf-relay sip-notify
    -Terry

  • Dtmf-relay sip-notify

    Evening Cisco techs,
    Having trouble enabling sip-notify
    Using CME 3.3 via (C2691-IPVOICEK9-M), Version 12.4(25d), RELEASE SOFTWARE (fc1)
    Have registered cisco phones set up in CME
    SIP is enabled on router. So is H323
    In conf t mode via voice service voip
    Upon using dtmf-relay sip-notify I get the error of unrecognised command
    If other information is required. Please let me know

    Just go to your SIP dial-peers:
    R01#conf t
    R01#dial-peer voice XX voip    << XX is you SIP dial-peer
    R01(config-dial-peer)#dtmf-relay ?
      cisco-rtp          Cisco Proprietary RTP
      h245-alphanumeric  DTMF Relay via H245 Alphanumeric IE
      h245-signal        DTMF Relay via H245 Signal IE
      rtp-nte            RTP Named Telephone Event RFC 2833
      sip-kpml           DTMF Relay via KPML over SIP SUBCRIBE/NOTIFY
      sip-notify         DTMF Relay via SIP NOTIFY messages
    dial-peer voice 555 voip
     destination-pattern 55555
     session protocol sipv2
     session target ipv4:1.1.1.1
     dtmf-relay sip-notify
    -Terry

  • Do you think isdn and frame-relay expire?

    in new ccnp book i didnt see any thing about isdn and some object
    even in ccna exam i didnt see any question about isdn and if i remember about frame-relay
    what's your opinion?do you think isdn and frame-relay expire?is there any place for this technology on it nowadays?
    thanks

    Hi Reza,
    if you are talking about new CCNP curriculum, then ISDN and Frame relay technologies are no longer included there. ISDN and Frame relay were included in the Remote access paper of CCNP old curriculum but now that paper is replaced by Secure Converged Wide Area Networks paper which includes MPLS, IPSec VPNs and other technologies.
    But as Rick said in earlier post that ISDN and frame relay are still very much in use.
    hope it helps ... rate if it does ....

  • CME/CUE SIP Phones DTMF-Relay

    Hi all,
    Just looking for some clarification on this one.  I'm seeing some conflicting advice about setting the DTMF-Relay on SIP Phones registered to CME with a CUE Module.  I've read some documentation indicating that rtp-nte RFC2833 is the only dtmf-relay supported for SIP Phones registered to CME, however I've also read some documents indicating that sip-notify must be configured as the dtmf-relay on SIP phones when they are communicating to a CUE module.  I'm assuming I'm going to need to configure an MTP on the CME, but just wondering what the official DTMF config should be under the voice register pool for SIP phones.
    Thanks!

    Hi  logan
    When doing lab with cme 7.0 and sip phones .sip phones are  not recognizing the "sip-notify" dtmf-relay method .It can only recognize "rtp-nte" method and it does not matter weather you are using sip-notify or rtp-nte for a dial-peer pointing to cme .
    i configured on cue
    ccn subsystem sip
    dtmf-relay sip-notify
    end
    on cme i configured a dial-peer pointing to cue
    dial-peer v 3888 voip
    destination-pattern 3888
    session target ipv4:177.3.11.10
    codec g711ulaw
    no vad
    session protocol sipv2
    dtmf-relay sip-notiy
    on my sip phones
    voice register pool 1
    dtmf-relay sip-notify  ------> now in this case cue wont recognize dtmf tones
                                        when i change this dtmf-relay method to rtp-nte it recognizes dtmf tones to  when recording  a message

  • ISDN Gateway error message- What does it mean and what could be causing it?

    Hi Folks,
    I have a customer that is getting the following errors in the logs of his Cisco/ Codian ISDN Gateway: Any ideas of what they mean and what could be causing them?
    I know Q931 is affiliated with Call Setup but not sure why an error would occur.  Any help that can be provided is much appreciated, thanks.
    Warning
    parse_q931_header: invalid protocol discrimator
    207
    09:45:50.968 
    ISDN
    Warning
    parse_q931_header: invalid protocol discrimator
    208
    09:45:50.975 
    ISDN
    Warning
    parse_q931_header: invalid protocol discrimator
    209
    09:45:51.120 
    ISDN
    Warning
    parse_q931_header: invalid protocol discrimator
    210
    09:45:51.128 
    ISDN
    Warning
    parse_q931_header: invalid protocol discrimator

    I think I narrowed this down to having a network volume mounted with a  share name of "Users".  The strange thing is even you if unmount the  volume you will still have problems.  However, if you avoid mounting the  volume entirely (from boot) it will be OK.
    The  solution, it appears, is to rename the sharepoint to something other  than "Users".  This worked for us.  I'm not sure if it matters whether  the server is Windows or Mac OS.  In our case the sharepoint was an AFP  volume on Mac OS.
    We had other errors related to Bridge  such as "unable to create folder" and the path in the message appeared  to be /Users/currentUser/.....  which made no sense at all.  However, if  Bridge was internally trying to use /Volumes/Users (the mounted  sharepoint), it would not only not find "currentUser", but it would not  have permissions to create that folder either.
    So, another possible solution could be to grant file creation rights on the network volume "Users".  Another not-so-great idea.
    Hope that helps someone else!

  • CUBE - New Deployment Issue - Not working DTMF Relay

    Hello,
    Scheme:
    Cisco SCCP-based IP Phone > CUCM 9.1 w/ SIP Trunk > CUBE (28XX, 151-4.M7) > SIP ITSP
    CUCM Active Call Proc. Node IP: 10.10.10.9
    CUBE Inside Interface IP: 10.10.10.10
    CUBE Outside Interface IP: 20.20.20.20
    Cisco IP Phone: 10.10.10.8
    ITSP SBC IP: 30.30.30.30
    ITSP SIP domain: itsp.domain
    Calling Pty: 9017654321 (translated in CUCM's route pattern which addresses CUBE)
    Called Pty: 9011234567
    While call was connected calling party dialed consequently 0,1,2,3,4 but far-end IVR does not react :(
    Symptom:
    While outbound call is connected calling party (IP Phone) dials digits which are not detected by any far-end PSTN (non-corporate) IVR at all.
    Thoughts:
    ITSP support only inband relay (RFC2833, Named Telephone Events or NTEs).
    Using NTE provides a standard way to transport DTMF tones in RTP packets.
    Thus rtp-nte is configured for both CUCM and ITSP dial-peers on CUBE.
    While initial troubleshooting found that for the active call inbound CUBE's leg shows rtp-nte, but outbound inband-voice.
    A have an assumption that ITSP doesn't give us 101=rtp-nte payload in 183 Response but I'm not sure.
    m=audio 10318 RTP/AVP 8
    b=AS:64
    a=rtpmap:8 PCMA/8000
    a=ptime:20
    a=maxptime:20
    Questions:
    1.How to make CUBE to successfully relay DTMF in according to ITSP requirement?
    2. Why 'show call act/hist voice brief' doesn't show call id? All my attempts are identified as 2... )
    It is hard to differentiate b/w call active/history records..

     Ayodeji,
    Thanks for your feedback.
    If you look through the already attached output 'show call act voice' of the file 'case-no-dtmf_-_cube-show-20140210-1.txt' you will find the following:
    PeerId=101
    CallOrigin=2
    tx_DtmfRelay=rtp-nte
    CallDuration=00:00:05 sec
    PeerId=201
    CallOrigin=1
    tx_DtmfRelay=inband-voice
    CallDuration=00:00:05 sec
    2    : 230 18:40:38.048 MSK Tue Feb 10 2015.1 +1890 pid:101 Answer 79017654321 active
     dur 00:00:04 tx:234/37440 rx:233/37280
     IP 10.10.10.8:22688 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711alaw TextRelay: off
     media inactive detected:n media contrl rcvd:n/a timestamp:n/a
     long duration call detected:n long duration call duration:n/a timestamp:n/a
    2    : 231 18:40:38.068 MSK Tue Feb 10 2015.1 +1860 pid:201 Originate 79011234567 active
     dur 00:00:04 tx:233/37280 rx:307/49120
     IP 30.30.30.30:10318 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711alaw TextRelay: off
     media inactive detected:n media contrl rcvd:n/a timestamp:n/a
     long duration call detected:n long duration call duration:n/a timestamp:n/a
    This means that the target dial-peers (inbound and outbound are matched as designed).
    dial-peer voice 101 voip
     description -= inbound leg from CUCM to CUBE =-
     session protocol sipv2
     incoming called-number x
     voice-class codec 1  
     voice-class sip bind control source-interface Loopback0
     voice-class sip bind media source-interface Loopback0
     dtmf-relay rtp-nte
     no vad
    dial-peer voice 201 voip
     description -= outbound leg from CUBE to ITSP =-
     translation-profile outgoing cdpn-delete-prefix-00XX7
     max-conn 40
     destination-pattern x
     session protocol sipv2
     session target dns:sbc.itsp.domain
     voice-class codec 1  
     voice-class sip profiles 1
     voice-class sip bind control source-interface Vlan100
     voice-class sip bind media source-interface Vlan100
     dtmf-relay rtp-nte
     no vad
    Now I have an argue with ITSP to make them send me NTE in their 183/200 response..
    I've also:
    1. Tried to disable dtmf-relay at all on dial-peers (trying inband-voice) but this doesn't work and also not recommended AFAIK.
    2. Changed the value for SIP Trunk DTMF Signaling Method from 'No preference' to 'RFC 2833' w/ reset applied recommended by Suresh. No luck.

  • ISDN and FRAME-RELAY

    Hi All,
    I currently have two 2500 routers, to get my hands dirty on ISDN and FRAME-RELAY experiments do i need seperate routers for them?. Because i cannot see where to connect to in regards to the required connections i.e.
    2 x Asynchronous/Synchronous Serial Interfaces (DB60)
    4 x Serial Interfaces (DB60)
    1 x ISDN Interface (RJ45)
    1 x Token Ring interface (DB9 and RJ45)
    THat i normally hear about when refering to ISDN or FRAME RELAY.
    Somebody please put me through.
    Secondly i would like to connect my 2500 to the internet ...please put me through how.
    Many thanks in Advance

    Olu
    There are some challenges in what you want to do with this equipment. If each router has 1 x ISDN interface then you can use that for ISDN - but you will need some other piece of equipment to furnish the ISDN switch functionality. You can not just connect 2 ISDN interfaces back to back.
    You would be able to do some testing of Frame Relay if you connected the routers back to back - if one of the cables is a DCE cable and one cable is a DTE cable. The testing that you can do with 2 routers back to back is somewhat limited. You could do much better testing if you had a third router which could be configured as a frame relay switch.
    Connecting your routers to the Internet will be possible if you connect one of the serial interfaces to an Internet Service provider.
    The routers have token ring LAN interfaces. Unless you have some token ring MAUs to establish the connections you will not be able to do effective testing using the token ring LAN interfaces.
    HTH
    Rick

  • Would like to acquire ISDN Gateway 3200 Series Firmware

    Dear All,
    Currently, I am using an ISDN Gateway 3220 running version 2.1(1.43)N that is out of warranty. However, I am experiencing an issue and would like to try an upgrade of firmware to the highest version for 3200 series which is version 2.2(1.79)N.
    The firmware is no longer supported and cannot be downloaded. I also understand the risk for upgrading, but does anybody have this firmware that they can provide me with?

    Have you looked in the downloads section?  I know it doesn't list 2.1(1.79), but you might be able to get close.  Just be sure to look at the description of each software version, looks like there are several that are for other ISDN Gateway models other than the one you're looking for.

  • Local and National Calls

    Local calls are calls made within your local area. They're untimed and charged at a flat fee. Calls within Australia that fall outside of your local call area are usually charged as National calls.
    Check if a phone number is within your local call area by following the steps below:
    Step 1
    Download our Charging Zones guide, available in either PDF (3.06MB) or Word format (3.02MB)
    Step 2
    Find your local Charging Zone. To do this, view the Charging Zones guide and look for your town or suburb (or nearest major town)
    Step 3
    When you've found your local charging zone, check the District heading to view which district you're located within
    Step 4
    Check which areas are covered by your local call district. To do this, view page 5 of the Charging Zones guide. Look for the district you located in the previous step
    Step 5
    When you've located your district, check the contains the following charging zones section to see which areas are covered by local calls. National calls are long distance calls to fixed line numbers within Australia. If you're calling a fixed line outside of your local call charging zone, you'll be charged National call rates. View call prices for your Home Phone plan.

    Re: Local and National Calls
    I was told that I would get a copy of the contract stating what calls were allowed and what Internet usage I was allowed. I also was told that all calls were local with in a certain radius of exchange and that this was 60klms. I want this all in writing or a statement.

  • How to configure sendmail to act as a mail gateway and relay mails???

    I installed the Solaris (Intel) v.8 (10/01) in my external network. Now, I want to configure it as a mail server and as a mail gateway between my internal mail host and itself.
    I think I have read all documentation which comes in Answerbook2 about mail gateway.
    I added some aliases to the aliases file in /etc/mail and executed the newaliases command.
    The aliases pointed to my mail host which is in my internal network.
    I included the IP address and the name of the mail host in the /etc/hosts
    I can telnet the internal mail host and I can connect to it and verify the internal account thru:
    mconnect "mail host"
    vrfy "account"
    But when I do
    /usr/lib/sendmail -v "alias" < /dev/null
    it translates the alias to the account in the internal mail host but it returns to me
    "account" ... User Unknow.
    When I try to use the mail server from one client in my internal network (I configure my mail client to use as Outgoing server the server in the external network) to send mails I receive the error:
    550 5.7.1 Relaying denied. IP name lookup failed
    I create a file called "access" and create the binary called "access.db" (makemap hash access < access), but it doesn�t work.
    I think that maybe the sendmail.cf doesn�t recognize the access file but I�m not sure. I have worked with other "sendmail" version under others OS (RedHat Linux) but the files are quite differents.
    Would you please help me?

    I found the way to do it and this is the procedure
    How to implement the mail gateway on Solaris 5.8
    You have to make changes in 3 different places
    1.- Files in /etc/mail (sendmail files)
    2.- /etc/hosts file
    3.- DNS files and /etc/nsswitch.conf file
    1.- Files in /etc/mail
    1.1.- Backup your existent sendmail.cf file and copy the main.cf file as
    sendmail.cf (cp /etc/mail/main.cf /etc/mail/sendmail.cf)
    1.2.- If you want to accept mail for entire domain in your mail gateway (your mail
    server is an MX record in the DNS files), you�ll have to modify the sendmail.cf
    file editing the line Dj as Dj$m or if you want to accept just mail for your specific
    server you must put Dj$w.$m (be carefull to avoid spaces).
    1.3.- Change in the sendmail.cf file the DM parameter adding ddn. It should
    looks like DMddn (no spaces anywhere).
    1.4.- Insert a line for the Dm parameter, which should include your domain
    (ex.:Dmyourdomain.com, be care with spaces). If you have defined multiple
    domains, add a line with Cm and put the domains one after other separated by
    spaces. You have to put a space between Cm and the first domain
    1.5.- Add the following rule in the ruleset, don�t use spaces, use tabs as
    separator:
    R$*<@$*.$+>$* $#ddn $@ $2.$3 $:$1<@$2.$3>$4 [email protected]
    1.6.- If you find the following rule in the ruleset you should comment out it.
    R$*<@$*.$+>$* $#$M $@$R $:$1<@$2.$3>$4 [email protected]
    1.7.- If your host is receiving mail under different names for "local" delivery,
    often you need to define those hosts in a "Cw" line and put the names one after
    other separated by spaces. You have to put a space between Cw and the first
    name.
    1.8.- If you want to relay mails to different domains or subnets you can include
    them into the file relay-domains. Each record in this file is a domain name (like
    sun.com) or a subnet (like 192.0.0., be carefull with last dot).
    1.9.- To test your sendmail service you have to stop and start it
    (/etc/init.d/sendmail stop; /etc/init.d/sendmail start) and you can do the following:
    1.9.1- To get basic debug information type:
    /usr/lib/sendmail -bt -d0.1 < /dev/null
    1.9.2- To test the conection:
    mconnect "mail server"
    vrfy "any alias"
    1.9.3- To test the alias you inserted into aliases file (remember run newaliases
    command after insert them).
    /usr/lib/sendmail -v "any alias" </dev/null
    2.- The /etc/hosts file:
    2.1.- You must include in this file the IP address server, its nickname
    (servername), its extended nickname (servername.domain), the word
    "mailhost", the word "mailhost" and your domain (mailhost.domain) and the
    word loghost.
    2.2.- If you wish you can include all the servers that you want in the file. You
    should do this is to avoid the use of DNS service translation.
    2.3.- Check the DNS service thru nslookup servername (ex. Nslookup
    sun.com), then test nslookup to your mail gateway
    3.- DNS files and /etc/nsswitch.conf file
    3.1.- Include a MX record for your mail gateway as the mail host for the entire
    domain and verify that each record into the zone files include the MX record.
    These files are in your DNS server (nameserver).
    3.2.- Check your /etc/nsswitch.conf file. The record host must include files and
    dns. It must look like
    hosts: files dns

  • In OBIEE 11g How to Change  Import Type Local Machine to Remote Machine

    Hi All,
    I Have Installed OBIEE 11g Client tools in my Machine and server is in Other Server.when i tried to Import metadata i am getting Connection Falied error.
    When i tried to import metadata in import type option i can see only one option Local Machine that is also in Disable mode.
    how to make import type option to Remote.
    Thanks and Regards
    Kiran Kumar

    In BI EE 10g, there was no Oracle Client bundled along with BI EE. So, BI EE will use your Oracle DB Home client to connect to the database. So, when you try to connect to the database through the Repository or BI Server, it will try to find the tnsnames entry in the Oracle Client of BI EE 11g instead of your database. There are 2 options to work around this
    Copy your tnsnames.ora and Paste it to
    1. Drive:Oracle\Middleware\Oracle_BI1}network\admin
    2 Drive:Oracle\Middleware\oracle_common\network\admin
    TNS File like,
    DBName =
    (DESCRIPTION =
    (ADDRESS = (PROTOCOL = TCP)(HOST = Your Server IP)(PORT = 1521))
    (CONNECT_DATA =
    (SERVER = DEDICATED)
    (SERVICE_NAME = DBName)
    Thanks,
    Balaa...

  • Can recieve mail but not relayed to local users, can't send mail,

    I am working on setting up a home server for web and email. I have server 10.2 and I can get the web working fine but the mail is giving me fits. The computer is behind by router on my private network. I have a domain name that is set for my router and it forwards all requests to the server. Have done that before with linux. I have set the computers network name to be the same as the domain name(probably a mistake). I can get the mail to be accepted by the server but then it gives a no user error when transfering to the local host for delivery. The only way I can edit using any of the tools is to type in the IP number and not the name.
    Should I have named it something like server.domain.com? It is not too far along to reformat and start over. How do you change the network name? Is the network name my problem? I have tried messing with the DNS but it is not as easy and it sounds for 10.2 as in later versions.
    Any suggetions would be helpful.
    David

    It's been too long since I've seen 10.2 to give you tips on it's mail config....
    The main issue you may have is that your router probably doesn't support loopback. This page has a good explanation:
    http://www.dyndns.com/support/kb/loopback_connections.html
    You can use /etc/hosts as described in the article if the machines stay in the office. Another solution is to change your router to one which does support loopback.
    Jeff

Maybe you are looking for

  • Ardour fails to launch (Solved)

    I just installed ardour-2.0beta11.1 on a clean install of Arch, and get the following message: error while loading shared libraries: libFLAC.so.7: cannot open shared object file: No such file or directory I have FLAC installed (and in my path), so wh

  • Custom Rules in ORACLE 10g

    Hi, I have an Ontology containing some sample triples. for eg:         v1:Item v1:hasCost "100"         v1:Item v1:hasWarrantyForYears "3" There are some instances of Item in the Ontology, with hasColor=100 and some with hasWarrantyForYears=3. Now wh

  • Can MacbookPro7, 1 Version 10.6.8 get an 10.7.5 upgrade?

    I have a MacbookPro 7, 1 purchased in Summer of 2010 that I would like to get the 10.7.5 update.  I bought a game and it only works on operating systems 10.7.5 or higher.  I did all the software updates possible on my macbook and it is still at a 10.

  • Can you suggest a suitable RIP for HP Designjet Z3100.

    The workload on our Designjet Z3100 has grown to the point where we need to increase the productivity of the printer.  I believe adding a RIP to the workflow would help, both in terms of accuracy of setup and minimising paper waste (through nesting).

  • How to planning profile using BPS0 tcode

    Hi all , I have a question about BPS. In BW7.0 system, using bps0, I could't see those standard planning profiles, such as 4CRMMP02, 4CRMMP03, 4CRMMP04, 4CRMMP05, 4CRMMP06, 4CRMMP09. And I could't find planning profiles in BI Content. I want to know