Ongoing Exchange Integration Issues

Discussion in '3CX Phone System - General' started by Anonymous, Jun 2, 2007.

  1. Anonymous

    Anonymous Guest

    Hello Everyone. Sorry for the long post, but here it goes.

    Well I have been trying to integrate exchange and 3cx enterprise on the same box for quite some time now and have been failing miserablly...!!

    Here is my setup:
    w2k3 server x64, Exchange2007, 3CX Enterprise, (2) Cisco 7960 Phones

    I can integrate to them perfectly if I keep 3CX on one box and put Exchange 2007 on another, but I am stubborn in my thoughts that this should work if the settings are right. I also feel it would be a much cleanner setup to have the Voicemail and the PBX in one turnkey solutions.

    So in order to have them on the same box, I was told that the SIP port on 3CX had to be something other than 5060. So I am using 5070 for the PBX SIP port. However, I can't get my phones to fully communicate with 3CX when the SIP port is 5070. If I switch the port to 5060, everything works great.

    I am using 2 cisco 7960 phones (8.2) with real simple configs:
    SIPDefault.cnf
    image_version:p0S3-08-2-00

    #Proxy server address
    proxy1_address: 192.168.0.2
    proxy1_port: 5070

    # Proxy Registration (0-disable (default), 1-enable)
    proxy_register: 1

    # Phone Registration Expiration [1-3932100 sec] (Default - 3600)
    timer_register_expires: 60

    # Codec for media stream (g711ulaw (default), g711alaw, g729)
    preferred_codec: g711alaw

    SIP(MacAddress).cnf
    # SIP Configuration Generic File (start)

    image_version: POS3-08-2-00

    line1_name : 101
    line1_authname : 101
    line1_password :

    In my troubleshooting I've turned off all exchange integration and am using 3cx as the voicemail and IVR system. When I call ext105 from ext101, I hear no audio, neither phone rings and the phone shows Proceeding(in 100).

    The server log shows this:
    16:46:05.015 MediaServerReporting::RTPReceiver from 'ippbx:0/MediaServer':No RTP packets were received on 0000004F@:remoteAddr=192.168.0.103:20370,extAddr=0.0.0.0:0,localAddr=192.168.0.2:7156

    16:46:05.000 StratInOut::eek:nHangUp Call(C:A): Call from Ext.101 to 999 has been terminated

    16:45:32.812 CallLegImpl::eek:nConnected Call(C:A): Created audio channel for Ext.101 :)20370) with Media Server (192.168.0.2:7156)

    16:45:17.406 CallConf::eek:nIncoming Call(C:A): Incoming call from Ext.101 to sip:105@192.168.0.2

    16:45:07.515 StratInOut::eek:nHangUp Call(C:9): Call from Ext.101 to 101 has been terminated

    16:45:00.890 CallConf::eek:nIncoming Call(C:9): Incoming call from Ext.101 to sip:101@192.168.0.2

    16:45:00.062 MediaServerReporting::RTPReceiver from 'ippbx:0/MediaServer':No RTP packets were received on 0000004C@:remoteAddr=192.168.0.103:27396,extAddr=0.0.0.0:0,localAddr=192.168.0.2:7150

    16:45:00.046 StratInOut::eek:nCancel Call(C:8): Call from Ext.101 to 999 has been terminated

    16:44:50.359 gt;>:Illegal message rejected: Missing parameter tag

    16:44:50.234 StratInOut::eek:nHangUp Call(C:6): Call from Ext.101 to 800 has been terminated

    16:44:44.750 CallLegImpl::eek:nConnected Call(C:8): Created audio channel for Ext.101 :)27396) with Media Server (192.168.0.2:7150)

    16:44:44.703 CallConf::eek:nIncoming Call(C:8): Incoming call from Ext.101 to sip:999@192.168.0.2
     
  2. Anonymous

    Anonymous Guest

    Yep, I agree with that. One box with the whole lot. I see no reason why that could not work.

    First, can you list what it is you trying to integrate/do with exchange and 3cx so we can work towards the same goal.

    What stops working if you leave the sip port on 5060? Why can you not use port 5060? I believe you only have to change ports if you run multi channels.

    RTP packets are in general in the 10000 - 20000 range are they not? Do you have a firewall somewhere that blocks the 20370 port?


    Would good to see this fixed, as it is my "dream" config :).
     
  3. Anonymous

    Anonymous Guest

    I am setting up Exchange 2007 Unified messaging to act as my voicemail and digital receptionist. Lite years ahead of 3cx. (No offense 3cx, i love your product to death, just no the voicemail package.)


    Exchange Unified Messaging SIP protocol communicates with 3cx on port 5060 which cannot be changed. So if you set 3cx to use that port as well for it's SIP communication with extensions, either the exchange UM service or the 3cx service will crash.

    Now if exchange07 and 3cx were on differnet boxes, then that wouldn't matter. However I am too stubborn to seperate the two packages.

    What is port 20370? What does that do? I think that using a nonstandard port in 3cx is not just as easy as changing the port in the General Settings page. Although they make it seem that easy, everything goes haywire when I change the SIP port.

    I must say, this is my dream config as well.! Aside from the fact that I have two potential customers interested in having this all in one solution but I'd like to have it running live in my office as well. The exchange UM is the most impressive piece of software that Microsoft has put out since Windows 95. And it's a much cheaper PBX/voicemail solution when compared to all the major phone vendors.

    I just wish I could get these bugs worked out!.

    Brian
     
  4. Anonymous

    Anonymous Guest

    Ok here goes nothing, let me check if MS send me the latest exchange .....


    Yep, ok ill build a box now and see if we can sort this. Would be good :).
     
  5. Anonymous

    Anonymous Guest

    Alright... sounds good.

    I finally found the 3cx SIP phone after looking forever. It should be easier to find on the site. It's a great little program and helps with the debugging.

    Strangely enough, It work a lot better than actual SIP phones. I can hear aduio and everything. What does that tell you? Wrong config on my phones?

    Here is the error when the call is handed off to exhange.

    -------------------------------------------
    22:12:42,861: R: 74.92.90.221:5050
    SIP/2.0 403 Forbidden
    Via: SIP/2.0/UDP 71.255.2.135:50862;branch=z9hG4bK0061828ae50fdc11949a4fa9d498b0b0
    To: <sip:101@www.jaydien.com:5050>;tag=f96e1c37
    From: "Test"<sip:109@www.jaydien.com:5050>;tag=10933
    Call-ID: 0061828A-E50F-DC11-9499-4FA9D498B0B0@71.255.2.135
    CSeq: 2 INVITE
    Warning: 403 "Forbidden"
    Content-Length: 0


    -------------------------------------------
    22:12:42,871: T: 74.92.90.221:5050
    ACK sip:101@www.jaydien.com:5050 SIP/2.0
    Via: SIP/2.0/UDP 71.255.2.135:50862;branch=z9hG4bK0061828ae50fdc11949a4fa9d498b0b0
    From: "Test" <sip:109@www.jaydien.com:5050>;tag=10933
    To: <sip:101@www.jaydien.com:5050>;tag=f96e1c37
    Call-ID: 0061828A-E50F-DC11-9499-4FA9D498B0B0@71.255.2.135
    CSeq: 2 ACK
    Content-Length: 0


    -------------------------------------------
    22:12:42,891: Disconnect Indication: 0E 00 01 00 04 82 24 00 01 01 00 00 95 38
    22:12:42,901: Disconnect Indication: 21:Call rejected
    22:12:42,911: Disconnect Response: 0C 00 01 00 04 83 24 00 01 01 00 00
    -------------------------------------------

    By the way, what was that port that you mentioned in that earlier post?

    Brian
     
  6. Anonymous

    Anonymous Guest

    Interesting, I do not even know where they hide the 3cx SIP phone. I think they have to think about a central download section for the site.

    For example
    Downloads
    -- IP PBX software
    -- 3cx Phone
    -- Templates

    Etc.

    Ok back on topic,

    Might be "proxy" can you configure proxy on your IP phones?
     
  7. Anonymous

    Anonymous Guest

    Just had a go at if, but I have no 64bit servers available and the 32 bit version is eval only so Ill give it a mis till later
     
  8. Anonymous

    Anonymous Guest

    Well. I guess I'llhave to keep tinkering with it. Besides the 32 bit version is such a resource hog that it would be crashing too many times on you. Amazing what a difference the 64bit OS does for you when running exchnge.

    But are going to explain what that port was for me? :lol: The suspense it just too much.

    Brian
    http://www.jaydien.com
     
  9. Anonymous

    Anonymous Guest

    LOL, well it keeps you sharp I guess :).

    remoteAddr=192.168.0.103:27396 (that is an RTP port)
    extAddr=0.0.0.0:0
    localAddr=192.168.0.2:7150 (This is your internal extension Port, see general options).

    I believe you could register your phones? but no sound comes through hence the RTP port. (Realtime Transport Protocol) RTP in general is described in RFC 3550, now do not go out and read that as it is very "dry" material. It comes down to that the following ports where reserved for RTP 16384-32767. So you see that the port 27396 (that is an even port, the rtcp is carried on an odd port (the next one up))is part of that "block".

    So where I would look is:
    Ports and "extAddr=0.0.0.0:0" try to get those resolved.

    But than again what is the softphone doing different???

    For some more info on what ports microsoft is using:
    http://www.microsoft.com/smallbusiness/support/articles/ref_net_ports_ms_prod.mspx
     
  10. Anonymous

    Anonymous Guest

    Ok. I'm starting to get it. Let me try and work with the RTP ports and see if that helps my situation. I'm sure that the NAT transversal is making the situation more difficult too.

    Check this site out. Here is where I eventually found the 3cx softphone.

    http://www.deerfield.com/download/3cx/

    Brian
    http://www.jaydien.com
     

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