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A
Anonymous
July 4, 2011
Question

Problem receiving calls.

  • July 4, 2011
  • 3 replies
  • 1062 views

Hello!, im new using FMG and FMS and I have problems receiving calls. I create a new profile in the sip.xml, rtm.xml and in the workflow.xml. I can call from sample sipphone to any other phone via my sip server, but i cant receive calls, can somebody help me please. .

sip.xml

        <Profile>
            <profileID> MyCon </profileID>
            <userName> myuser </userName>
            <password>mypass </password>
            <displayName> test </displayName>
            <sipDomain> sip domain </sipDomain >
            <doRegister> 1 </doRegister>
            <remoteSipHost> sip proxy </remoteSipHost>
            <supportedCodecs>
                <codecID> G711u </codecID>
            </supportedCodecs>
            <context> mycontext </context>
        </Profile>

workflow.xml

        <Context name="mycontext">
            <Condition variable="destNum" value="^8888$">
                <AppNode sequence="1" app="playfile" args="moh.raw"/>
                <AppNode sequence="2" app="hangup" args="null"/>
            </Condition>
           
            <!-- call rtmp phone -->
            <Condition variable="destNum" value="^1...$">
                <AppNode sequence="1" app="bridge" args="rtmp|${destNum}@FlashCall"/>
                <AppNode sequence="2" app="hangup" args="null"/>
            </Condition>
           
            <!-- call sip phone 777 -->
            <Condition variable="destNum" value="^777$">
                <AppNode sequence="1" app="bridge" args="sip|${destNum}@sipPhone"/>
                <AppNode sequence="2" app="hangup" args="null"/>
            </Condition>
           
            <!-- call sip phone 888 -->
            <Condition variable="destNum" value="^888$">
                <AppNode sequence="1" app="bridge" args="sip|${destNum}@sipPhone2"/>
                <AppNode sequence="2" app="hangup" args="null"/>
            </Condition>
           
            <!-- call out -->
            <Condition variable="destNum" value="^.*$">
                <AppNode sequence="1" app="bridge" args="sip|${destNum}@MyCon"/>
                <AppNode sequence="2" app="hangup" args="null"/>
            </Condition>
        </Context>

rtm.xml

    <Profiles>
        <EnableAutoProfile>false</EnableAutoProfile>

        <DefaultContext>rtmp</DefaultContext>

        <DefaultPassCode>passcode_default</DefaultPassCode>

        <Profile>
            <ProfileID> profile_default </ProfileID>
            <Server> localhost </Server>
            <Application> telephony </Application>
            <Instance> _definst_ </Instance>
            <PassCode>passcode_default</PassCode>

            <Context> rtmp </Context>
        </Profile>
       
        <Profile>
            <ProfileID> FlashCall </ProfileID>
            <Server> localhost </Server>
            <Application> telephony </Application>
            <Instance> FlashTest </Instance>
            <Context> mycontext </Context>
        </Profile>
       
       
    </Profiles>

Thanks

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3 replies

A
Anonymous
July 5, 2011

Thanks for the quick answer, Im going to try that . I'll let you know if it work.

A
Anonymous
July 5, 2011

Hi,

     Some test should help you. Since the profile you have created mycontext is common for both a) the call originating from flash phone as well as b) the call originating from SIP. You should be able to place calls both ways identically.

A test to confirm (this would help identify if there is something kaput on SIP configuration):

    - Try connecting two flashphones to rtmp://<fms-machine-host>/telephony/FlashTest

   -  Dial number of one flash phone from the another.

    - does the call succeeds ? if not please dig into core.00.log and look for the error message reported by FMG while it was processing this call.

Coming back to your problem:

   - Once you feel confident, try the same process by dialing from SIP phone; here are tthe troubleshooting steps by looking into core00.log file

    1) Confirm FMG is actually processing using "myContext" which you have defined in workflow.xml. In certain cases, FMG might be processing using some other profile which could be due to failed registration of SIP client. In this case FMG would fail to recognise your sip client and would use a defaultContext from sip.xml instead of the one you specified.

    2) If logs show that FMG used correct workflow context; move further in the log timeline and see whether FMG tried creating a connection (fmg calls it a callLeg). was it created successfully or there were some error. In the call timeline this would be the second callLeg which FMG initiates after processing mycontext workflow.

     3) If RTMP callLeg was created, track the states of "second" callLeg via states INIT, RING , SENDRECV, HANGUP. If the call was unsuccessfull, the second callLeg (i.e. the one to destination) would hangup before ever reaching to SENDRECV state. In this case note the error message. which would give the clue of why FMG clouldn't connect the the destination flash phone.

      4) If FMG logs complain that rtmp profile (or RTMP URL) is not connected. Try confirming a connected that you are logged into that flash phone and can place a call.

I hope it helps.

Note: In case above steps doesn't help, if relevant, please mention few things like :

            1) RTMP URL to which flash phone is connected.

             2) The phone id displayed by flashphone UI

            3) Phone number dialled from SIP Phone.

A
Anonymous
July 5, 2011

I think when you say sample SIP phone you means sample flash phone which comes with FMG, if that is the case it is possible that you had not registered the flash phone on the correct instance. If flash phone and FMS are on same machine you should use rtmp://localhost/telephony/FlashTest as server URI. If that doesn't work or my assumption was wrong I suggest you to mail your latest logs and conf files to fmg@adobe.com.

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