SIP: Choosing sip account from SIP URI

It is possible to store arbitrary strings in vcards, for example sip:[email protected], but unfortunately the inbuilt SIP client seems not to parse the URI and tries to reach myfriend at someregistrar.com, but uses the default account for all calls.
Is it possible to select the SIP account for outgoing calls on a call by call basis? Using IP adresses for calling should be possible as well, e.g. sip:192.168.1.1.
Is it possible to reach the phone via SIP even when it is not registered - most SIP clients are reachable via their FQDN/IP address
Thanks!
Nokia E51

Flodis79,
No need to delete/add anything repeatedly. You can do that but as you mentioned it's not easy.
I'm not doing this under "Settings". I only use settings to define the two SIP accounts (VoipBuster and TruPhone). For example Menu->Tools->Settings->Connection takes me to where my "SIP Settings" and "Internet tel. settings" can be changed. Not talking about modifying these at all. However, your SIP Settings for both providers needs to be set to Registration == "always on" if you really want this to work as I stated in my original post.
Where I'm switching is:
Menu->Connect->Internet tel
When you do this it may say "Discovering services" and takes a few seconds. When that's done you will see a message pop up if you are already connected. In my case I see "Already connected to TruPhone and VoipBuster". The message will go away and there is an "opions" button in the lower left corner.
Select it and you should get a menu that says:
Connect to service or Disconnect from serv
Change service
Refresh
Settings
Help
Exit
The "Change service" is what I use to do what you are asking about. The "Connect to service" or "Disconnect from serv" option just depends on you current state.
So in my case I just click change service and a list will pop up with the Net call service to use:
TruPhone
VoipBuster
Just pick the one you want and that's it. Go to your phone book and make your call. Switch between the two providers between calls.
You don't need to "Disconnect from serv" to change it. Just another option if you already have an active profile want to stop using a profile. Also, if you select Connect to service and your SIP profile Registration is set to "Always on" it will get changed to "When Needed" for some reason? Took me a while to figure that out.
Does that help?
Also, you don't need to store two versions of the same phone number for each SIP account. Just put the phone number in your address book and the client will add the @sipProvder.com to the phone number when you make an Internet call.

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    timeouts wait-release 1
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    connection plar 600
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    line 1/0 1/15
    line vty 0 4
    login local
    transport input telnet
    scheduler allocate 20000 1000
    end

    Having looked at your spreadsheet I see you're failing H323 transfers back to your ISDN system, but only under certain circumstances. Quite why, I'm not sure, possibly because you haven't codec defined on your H323 dial peers. or it could be something else
    I think you may be able to work around the problem by adding
    " supplementary-service h450.12 " under voice service voip on your CME router as a quick fix.
    reference
    http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetrans.html#wpxref44614
    worth a try
    Adam

  • Help needed in receiving  two consecutive msges from SIP Ser in UDP server!

    Hi everyone,
    I am Mitul Gogoi,from Assam,North-east part of India.
    I am writing a SIP proxy server,which is simly a UDP server and which will sit between the Client(X-Lite softphone) and Brekeke SIP Server and just receive and send messages.
    Client---------------->My UDP Server-------------------------->SIP Server (This is for requests)
    again,
    SIP Server---------->My UDP Server-------------------------->Client (This is for responses)
    (couldnot draw the arrows together)
    My server will receive any msg coming from Client in port 7000 from Client and My server will send the msg to SIP Server to its IP address and port 5060.So, I just wanted to change the port number to which Client will send msges;i.e. instead of earlier port 5060,it will now send to port 7000.
    The msg sending and receiving scenerio:
    1.The first msg from Client is received by My Server.
    2.My server sends the msg to SIP Server.
    3.My server then waits for response from SIP server.
    4.One msg comes to My server and received successfully. and it closes the socket.
    5.This msg is sent to Client.
    6.Second msg comes from SIP server to My server.But My server is unable to receive this second msg.maybe because My server closes the socket to receive the next msg.
    I am using ResponseHandler Thread in main which will listen to any msges that may come from SIP server to My server.
    My question is :
    Is it not possible to receive two consecutive msg from SIP server ?
    If one msg comes,it then closes the socket.
    My code for sending and receiving :
    udpClSocket = new DatagramSocket();
    packet = new DatagramPacket(buf, buf.length,InetAddress.getByName(server), port);
    udpClSocket.send(packet);//it will send to SIP Server as well as Client,by changing the server and port
    byte resbuf[] = new byte[msgSize];
    packet = new DatagramPacket(resbuf, resbuf.length);
    udpClSocket.receive(packet);//it will receive msg from SIP server only
    packet = null;
    udpClSocket.close();
    The first msg comes to My server successfully,but the second msg is not being received.
    I have tried in so many ways,such as taking two different ports :one for receiving for Client and My server ,and another port for My server and SIP server.
    If anyone can help me in this ,then I will be highly grateful.
    regards,
    Mitul

    Why? Throw all this code away, and use the NIST reference implementation of JAIN-SIP. They've done all the hard work for you. All you have to do to write a stateless proxy is a little mild header processing.
    Thank you ejp for your reply.Mine is a Outbound SIP proxy server,which uses a port other than default 5060 SIP server port.The client will REGISTER SIP server via my server.and moreover,I want to build my own server at least once for learning purpose.
    maybe because My server closes the socket to receive the next msg.
    Why?
    I donot know why.may be I am very new to network programming or did not do much research in networking.But,my code should work;couldn't find out any fault.
    Here is my code:
    package com.ef;
    import java.net.DatagramPacket;
    import java.net.DatagramSocket;
    import java.net.InetAddress;
    import java.util.Date;
    * It sends UDP packets to following destinations-
    * 1. SIP Server
    * 2. SIP UACs
    public class OBUDPClient extends Thread {
         private DatagramSocket udpClSocket;
         private DatagramPacket packet;
         private String server = "192.168.1.2";//IP of Brekeke SIP server
         private int port = 5060;
    //SIP Message Size
         private int msgSize = 2048;
         private byte buf[];
         private boolean serFlag = true;
         private OBMessage m;
         * Send Request to SIP Server
         * @param in_packet
         * @throws Exception
         public OBUDPClient(DatagramPacket in_packet) throws Exception{
              super("Client-"+String.valueOf((new Date()).getTime()));
              this.buf = in_packet.getData();
              m = new OBMessage();
              * Keep track of IP and Port of the source of this packet. Response
              * from SIP Server will be redirected to this IP and Port
              m.setTargetIP(in_packet.getAddress().getHostAddress());
              m.setTargetPort(in_packet.getPort());
              start();
         * Send Request to SIP UAC (). This constructor gets call from OBResponseHandler.java
         * @param server
         * @param port
         * @param buf
         * @throws Exception
         public OBUDPClient(String server, int port, byte buf[]) throws Exception{
              super(String.valueOf((new Date()).getTime()));
              this.server = server;
              this.port = port;
              this.buf = buf;
              serFlag = false;
              start();
         public void run(){
              try{
              System.out.println("OBUDPClient Redirecting packet");
              udpClSocket = new DatagramSocket();
              //Send Request
              packet = new DatagramPacket(buf, buf.length,InetAddress.getByName(server), port);
              udpClSocket.send(packet);
              //Receive Response
              byte resbuf[] = new byte[msgSize];
              packet = new DatagramPacket(resbuf, resbuf.length);
              udpClSocket.receive(packet); //could not receive two consecutive msg from SIP server
              if(serFlag){
              * If request is sent to SIP Server we are interested for the
              * response otherwise not
              System.out.println("<----------------Handle Response----------------->");
         System.out.println("++++++++++++++++++++++++++++++++++++++++++++++++++");
         System.out.println("Response from SIP Server: "+new String(packet.getData()).trim());
         System.out.println("++++++++++++++++++++++++++++++++++++++++++++++++++");
         //Read SIP reseponse sent by SIP Server
         m.setMessage(new String(packet.getData()).trim());
         //Store the Response message in Queue
         OBMain.q.push(m);
         }catch(Exception ex){
         ex.printStackTrace();
         packet = null;
         udpClSocket.close();
    Note:Everything starts from the Client.First,Client makes a REGISTER request;it is passed through Outbound server to SIP server.Then,SIP server responds with 100 Trying;this is received successfully by my outbound server and sent to Client.Then,again,SIP server responds with 200 OK;this is not received by my outbound server;hence cannot reach Client,as a result of which Registration fails.
    regards,
    mitul

  • SIP stops when upgrading from ASA from 8.4.1 to 8.4(2)8 w/ out config change? Why?

    I have to be missiong something small in my config.
    If I upgrade my ASA 5510 which I am routing and NATing off of, from 8.4.1 to 8.4.2.8, SIP stops. All phones go dead.
    If I roll bck to 8.4.1, SIP comes up.,... Go bck to 8.4(2)8 nd SIP goes down..... 
    This is without mking any config changes.
    I have looked at it so long, I must be overlooking something simple, simple, simple...

    Have spent sIx hours in past 24 w/ Cisco TAC and they have a tin of caps as have I but can't figure out why there is a denial of SIP from inside outside and outside inside to/from sip providers three IP addresses. Have created new access lists, new access groups to allow all 3 ip's in & out, increased timeout, bypassed IPS, have both sip UDP & tcp allowed in/out, specified inspection to approve any any for all sip protocols in/out to/from Lync & mediation and nada.
    To answer another question, yes I'm certain config doesn't change... I reloaded tge same running config from a bkup just to make sure.....
    What I see in the logs coming in/out is the call does make it all the way through the SSM to the ASA..
    What happens there is the head scratcher...
    SiP even though allowed and even though I've specified it to push through inspection On ASA side is denied based on inspection rule...
    I also tried using another one of my (unused) public IPs for only SIP thinking that maulybe there was a core conflict with multiple services NATd to the same public IP but that also did nothing.
    On topology I only have a single location so I'm using my 5510 to route as well...
    Have 1 IIS web server l, SQL, (ports clised except to obe vendor and am allowing via access list by their IP and ipsec,) Exchange, Lync, Ironport, Endpoint and everything else is 80/80...
    Everything is on Server 08r2 w/ exception of web server and two boxes ( one stand-alone & one VM on hyper-v)  I am running Server8 for Microsoft TAP engineering / validation airlift. Neither of those are attached to UC/UM at all...
    I'm using dynect from dyndns for outside network web services and just piggybacking on time Warner metro e for internal (no physical DNS server)
    When I look at caps everything is identical in the tcp and UDP trace even on sip except for the denial...
    Which caps/logs would 'y'all like to see and I'll post em when I get home....
    Is there a link to bug notes Jullio? Is it sip specific? Any possibility of it being just a name/cosmetic big I can force a work around to?
    I recall when Asa first was released I had to specify port 25  allow instead of being able to simply say allow smtp .. That took 2 weeks but it allowed for a work around so whatever I can do/try I'm willing!! Someone may wanna tell TAC if it's a bug because after 6 hours yesterday they are saying there's not a bug... :)
    Thanks all!!!!

  • Calls forwarded from sip device

    I am setting up a callmanager with an h323 gateway.
    for incoming and outgoing calls , the customer wants the calls passed to the H323 gateway and then  passed to another gateway within the Lan, which has a SIP trunk to the outside world.
    Is it just a simple case of having a  H323 dial peer on each gateway to pass calls to and from callmanager?

    Yes, you need to have the ff dial-peers
    On h323 gateway
    1. inbound h323 dial-peer from cucm to h323 gateway
    2. outbound h323 dial-peer to cucm from h323 gateway
    3. Outbound sip-dial-peer from h323 gateway to the sip gateway
    4. inbound sip dial-peer from sip gateway to h323 gateway
    On SIP gateway
    5. inbound sip dial-peer for calls from h323 gateway to sip gateway
    6. outbound sip dial-peer to h323 gateway
    7. outbound dial-peer for PSTN calls to ITSP
    8. You may also have inbound dial-peer from ITSP for inbound PSTN calls

  • Calls from Sip Trunk to UC540 and then to CUE returned ** Service Unavailable**

    Hi to all
    i have something strange here and i need your assistance
    Call Flow:
    Sip trunk-->UC540--> CUE
    When calls coming to UC540 from outside and then going to cue then we send back service unavailable.I made a translation and i sent directly the incoming calls to CUE
    The same behavior is also if i send the calls to dummy number and then from there set forward all to voice mail.
    Incoming voicemail is working fine
    Incoming calls to phones also ok
    Uc540: 8.6
    CUE: 8.6.5
    A number: 99999999
    B number: 22777777
    Voice Mail Number:111
    Attached is the trace
    i see that we hit the correct dial peers .
    I have enable only trancoder since MTP is not register ( don't know why , but i don't think also that is necessary..
    voice service voip
     ip address trusted list
      ipv4 172.16.80.0 255.255.255.0
      ipv4 172.16.81.0 255.255.255.0
     allow-connections sip to sip
     supplementary-service h450.12
     no supplementary-service sip moved-temporarily
     no supplementary-service sip refer
     supplementary-service media-renegotiate
     sip
      no update-callerid
    dial-peer voice 1000 voip
     description **SIP TRUNK**
     translation-profile incoming SIP-INCOMING
     translation-profile outgoing SIP-OUTGOING
     destination-pattern 9T
     modem passthrough nse codec g711alaw
     session protocol sipv2
     session target sip-server
     incoming called-number .T
     voice-class codec 2  
     voice-class sip dtmf-relay force rtp-nte
     dtmf-relay rtp-nte
     fax-relay ecm disable
     no fax-relay sg3-to-g3
     fax rate 9600
     fax protocol pass-through g711alaw
     no vad
    dial-peer voice 2001 voip
     description ** cue voicemail pilot number **
     destination-pattern 111
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number 111
     no voice-class sip outbound-proxy   
     dtmf-relay sip-notify
     codec g711ulaw
     no vad
    Regards
    chrysostomos

    Hi
    Interface                  IP-Address      OK? Method Status                Protocol
    FastEthernet0/0            unassigned      YES NVRAM  up                    up
    FastEthernet0/0.10         192.168.0.10    YES DHCP   up                    up   ----> For internet
    FastEthernet0/0.20         10.151.5.130    YES NVRAM  up                    up  ------> For sip trunk
    In0/0                      10.1.10.2       YES unset  up                    up    --------> default gw for cue
    Vlan1                      unassigned      YES unset  up                    up
    Vlan100                    unassigned      YES unset  up                    up
    Vlan200                    unassigned      YES unset  up                    down
    Vlan300                    unassigned      YES unset  up                    down
    NVI0                       10.1.10.2       YES unset  up                    up
    BVI1                       192.168.20.1    YES NVRAM  up                    up
    BVI100                     10.1.1.1        YES NVRAM  up                    up   ---------> ip for cme
    Loopback0                  10.1.10.2       YES NVRAM  up                    up   ------> default gw for cue
    dial-peer voice 2001 voip
     description ** cue voicemail pilot number **
     destination-pattern 111
     b2bua
     session protocol sipv2
     session target ipv4:10.1.10.1
     incoming called-number 111
     no voice-class sip outbound-proxy
     voice-class sip bind control source-interface BVI100
     voice-class sip bind media source-interface BVI100
     dtmf-relay sip-notify
     codec g711ulaw
     no vad
    interface FastEthernet0/0.10
     description **FOR INTERNET**
     encapsulation dot1Q 10
     ip address dhcp
     ip access-group 105 in
     ip nat outside
     ip inspect SDM_LOW out
     ip virtual-reassembly in
    interface FastEthernet0/0.20
     description **FOR SIP TRUNK WITH ISP**
     encapsulation dot1Q 20
     ip address 10.151.5.130 255.255.255.240
    ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
    ping 10.1.10.1 source bvi100
    Type escape sequence to abort.
    Sending 5, 100-byte ICMP Echos to 10.1.10.1, timeout is 2 seconds:
    Packet sent with a source address of 10.1.1.1
    Success rate is 100 percent (5/5), round-trip min/avg/max = 1/1/1 ms
    I have bind the interface of cme ( 10.1.1.1) but the call fails again
    Attached is the trace
    Anything to advice?

  • 3rd Party SIP phone to CUCM via SIP Proxy

    Hi all,
    This is the scenario i'm currently working on :
    3rd party SIP phone <--> Internet <--> SIP Proxy <--> LAN <--> CUCM
    The SIP proxy basically terminates everything (REGISTER, INVITE, etc), including the RTP stream.
    I can register the 3rd party SIP phone to CUCM and in CUCM and  i can see SIP Proxy IP Address as the registered address of the phone.
    Calls from the 3rd party SIP phone to internal Cisco or internal 3rd party SIP phone and vice versa work like charm.
    The only (fatal) problem is i can only register 1 3rd party SIP phone to CUCM via this SIP proxy.
    Since this SIP Proxy always use its internal IP Address and port 5060 (TCP) as its source of registration, CUCM sees multiple registrations for multiple extensions (users) come from a single IP and port, and rejects the second registration request.
    It seems that CUCM binds a digest user to an IP address and port, therefore cannot accept multiple registrations from a single IP and port.
    Can anyone clarify this?  Or is there any way around this?
    I'm using CUCM 8.6.2 and CUCM 9.X (both do not work).
    Regards,
    Christian

    This is most likely because of the following...
    Because third-party SIP phones do not send a MAC address, they must identify themselves by using digest authentication.
    The REGISTER message includes the following header:
    Authorization: Digest username="xxxxxxxxxx",realm="ccmsipline",nonce="GBauADss2qoWr6k9y3hGGVDAqnLfoLk5",uri="sip:172.18.197.224",algorithm=MD5,response="126c0643a4923359ab59d4f53494552e"
    The username, xxxxxxxxxxx, must match an end user that is configured in the End User Configuration window of Cisco Unified CallManager Administration. The administrator configures the SIP third-party phone with the user; for example, swhite, in the Digest User field of Phone Configuration window.
    See the following document.
    http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/5_1_3/ccmcfg/b09sip3p.html
    Also Try this bug CSCef88775

  • Sip 503 service unavailable and sip 500 internal server error

    Hi guys,could any one help me in the following.
    ITSP-->Voice gateway configured as CUBE-->CUCM-->UCCX
    I am moving a system from cme and aa enviroment to cucm and uccx
    The VGW is configured as CUBE and also is added as h323 gateway on cucm.
    When i tested the debug ccsip messages shows
    Sip 503 service unavailable or
    sip 500 internal server error.
    I can't now provide any debugs cause i am not on site,only on Saturday.
    As i read in previous discussion that could be the bind source address problem but i had this configured.
    Also i tried to configure the gateway instead of h232 to use sip trunk from cucm,but after this the incoming calls didn't even reach the router,the debug ccsip messages showed nothing.
    For now can any one advice me to what these 2 errors related to.
    What could be missing?
    Thanks in advance.

    Hi there : can some one explain the reason that i am getting this sip error with itsp:
    here is the debug of ccsip messages:
    Received:
    INVITE sip:[email protected];user=phone SIP/2.0
    Via: SIP/2.0/UDP 188.254.68.66:9298;branch=z9hG4bK-6110d60075a24c0f-a3c000c-1
    Call-ID: isbc6994325518768294927-1385194135-11717
    From: [email protected];user=phone>;tag=sbc09106994325518768294927
    To:
    CSeq: 1 INVITE
    Min-SE: 90
    Session-Expires: 3600;refresher=uac
    Contact:
    Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO,PRACK
    Supported: timer,100rel
    Diversion: [email protected]>;privacy=off;screen=no;reason=unknown,[email protected]>;privacy=off;screen=no;reason=unknown
    Max-Forwards: 70
    User-Agent: VCS 5.8.2.56-03
    Content-Length: 394
    Content-Type: application/sdp
    v=0
    o=- 87852 198805 IN IP4 188.254.68.67
    s=SBC call
    c=IN IP4 188.254.68.67
    t=0 0
    m=audio 23682 RTP/AVP 8 0 18 98 96 97 101
    a=rtpmap:98 G.729a/8000
    a=rtpmap:96 G.729ab/8000
    a=rtpmap:97 G.729b/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=fmtp:18 annexb=no
    a=ptime:10
    a=X-vrzcap:vbd Ver=1 Mode=FaxPr ModemRtpRed=0
    a=X-vrzcap:identification bin=DSR2866 Prot=mgcp App=MG
    00:43:23: //11/FDB448CE8020/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 188.254.68.66:9298;branch=z9hG4bK-6110d60075a24c0f-a3c000c-1
    From: [email protected];user=phone>;tag=sbc09106994325518768294927
    To:
    Date: Sat, 23 Nov 2013 08:06:29 GMT
    Call-ID: isbc6994325518768294927-1385194135-11717
    CSeq: 1 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    00:43:23: //11/FDB448CE8020/SIP/Msg/ccsipDisplayMsg:
    Sent:
    SIP/2.0 503 Service Unavailable
    Via: SIP/2.0/UDP 188.254.68.66:9298;branch=z9hG4bK-6110d60075a24c0f-a3c000c-1
    From: [email protected];user=phone>;tag=sbc09106994325518768294927
    To:
    c2801#er=phone>;tag=27BA64-1DAE
    Date: Sat, 23 Nov 2013 08:06:29 GMT
    Call-ID: isbc6994325518768294927-1385194135-11717
    CSeq: 1 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Reason: Q.850;cause=38
    Content-Length: 0
    00:43:23: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    ACK sip:[email protected];user=phone SIP/2.0
    Via: SIP/2.0/UDP 188.254.68.66:9298;branch=z9hG4bK-6110d60075a24c0f-a3c000c-1
    Call-ID: isbc6994325518768294927-1385194135-11717
    From: [email protected];user=phone>;tag=sbc09106994325518768294927
    To: ;tag=27BA64-1DAE
    CSeq: 1 ACK
    Max-Forwards: 70
    Content-Length: 0
    show run:
    voice service voip
    ip address trusted list
      ipv4 87.226.136.164 255.255.255.255
      ipv4 172.16.24.0 255.255.255.0
      ipv4 188.254.68.66 255.255.255.255
      ipv4 188.254.68.67 255.255.255.255
      ipv4 188.254.69.66 255.255.255.255
      ipv4 188.254.69.67 255.255.255.255
      ipv4 46.38.52.68 255.255.255.255
    address-hiding
    allow-connections h323 to h323
    allow-connections h323 to sip
    allow-connections sip to h323
    allow-connections sip to sip
    supplementary-service h450.12
    no supplementary-service sip moved-temporarily
    no supplementary-service sip refer
    redirect ip2ip
    fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback cisco
    sip
    voice class codec 1
    codec preference 1 g729br8
    codec preference 2 g729r8
    codec preference 3 g711alaw
    codec preference 4 g711ulaw
    voice class codec 2
    codec preference 1 g711ulaw
    codec preference 2 g711alaw
    codec preference 3 g729r8
    codec preference 4 g729br8
    voice translation-rule 1
    rule 1 /XXX5397962/ /1999/
    voice translation-rule 2
    rule 1 /XXX55317577/ /1999/
    voice translation-rule 3
    rule 1 /5555317884/ /1999/
    voice translation-profile ROS
    translate called 1
    voice translation-profile ROS2
    translate called 2
    voice translation-profile ROS3
    translate called 3
    interface FastEthernet0/0
    ip address 178.208.129.221 255.255.255.248
    ip access-group INBOUND in
    no ip unreachables
    ip verify unicast reverse-path
    ip nat outside
    ip inspect IPFW in
    ip inspect IPFW out
    ip virtual-reassembly in
    duplex auto
    speed auto
    no cdp enable
    interface FastEthernet0/1
    no ip address
    ip nat inside
    ip virtual-reassembly in
    duplex auto
    speed auto
    interface FastEthernet0/1.1
    encapsulation dot1Q 1 native
    ip address 10.110.0.200 255.255.255.0
    ip nat inside
    ip virtual-reassembly in
    interface FastEthernet0/1.2
    encapsulation dot1Q 2
    ip address 172.16.24.254 255.255.255.0
    ip nat inside
    ip virtual-reassembly in
    h323-gateway voip interface
    h323-gateway voip bind srcaddr 172.16.24.254
    ip dns server
    ip nat inside source list NAT interface FastEthernet0/0 overload
    ip route 0.0.0.0 0.0.0.0 178.208.X.X
    ip route 192.168.0.0 255.255.0.0 Null0 254
    sccp local FastEthernet0/1.2
    sccp ccm 172.16.24.101 identifier 1 version 7.0
    sccp
    sccp ccm group 1
    associate ccm 1 priority 1
    associate profile 1 register XCODE123456
    keepalive retries 1
    keepalive timeout 10
    switchover method immediate
    switchback method immediate
    dspfarm profile 1 transcode 
    codec g711ulaw
    codec g711alaw
    codec g729ar8
    codec g729abr8
    codec g729r8
    codec g729br8
    maximum sessions 6
    associate application SCCP
    dial-peer voice 10000 voip
    tone ringback alert-no-PI
    description ROSTELECOM Incoming
    translation-profile incoming ROS
    destination-pattern 74955397962
    session protocol sipv2
    session target ipv4:87.226.136.164
    session transport udp
    incoming called-number XXXX5397962
    dtmf-relay rtp-nte
    codec g711ulaw
    dial-peer voice 10010 voip
    tone ringback alert-no-PI
    description ROSTELECOM Incoming
    translation-profile incoming ROS2
    destination-pattern XXX55317577
    session protocol sipv2
    session target ipv4:87.226.136.164
    session transport udp
    incoming called-number 75555317577
    dtmf-relay rtp-nte
    codec g711ulaw
    dial-peer voice 10020 voip
    tone ringback alert-no-PI
    description ROSTELECOM Incoming
    translation-profile incoming ROS3
    preference 1
    destination-pattern 5555317884
    session protocol sipv2
    session target ipv4:188.254.68.66
    session transport udp
    incoming called-number 5555317884
    dtmf-relay rtp-nte
    codec g711ulaw
    dial-peer voice 10021 voip
    tone ringback alert-no-PI
    description ROSTELECOM Incoming
    translation-profile incoming ROS
    preference 2
    destination-pattern 5555317884
    session protocol sipv2
    session target ipv4:188.254.69.66
    session transport udp
    incoming called-number 5555317884
    dtmf-relay rtp-nte
    codec g711ulaw
    dial-peer voice 2 voip
    tone ringback alert-no-PI
    description to CUCM_PUB
    destination-pattern 1...
    session target ipv4:172.16.24.101
    voice-class codec 2 
    dtmf-relay rtp-nte
    I see in the debug that the itsp over g729 family codecs but not g711 at all
    This system was working with this dialpeers before with same provider ,just i have added the dial-peer 2 .
    I have changed the codec to match what is offered by itsp but no difference,still getting the same message.
    PLZ help ASAP.

  • HTTP Authentication Digest for SIP messages in a trunk SIP CUCME

    Hello,
    we would like to implement HTTP Authentication Digest for SIP messages in a trunk SIP between a Cisco 2851 and an Asterisk server.
    We are using CUCM Express with 15.1(4)M (CME 8.6) as voice gateway to connect to PSTN.
    According to Cisco documentation:
    "To configure a gateway to use HTTP Authentication Digest, give the following command in each dial peer or SIP-UA configuration mode:
    authentication username username password password [realm realm]."
    The problem is that when call is from CISCO to ASTERISK, Asterisk sends a challenge to Cisco to do Authentication:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.0.70.11:5060;branch=z9hG4bK3E205D
    Remote-Party-ID: "DN1001" <sip:[email protected]>;party=calling;screen=no;privacy=off
    From: "DN1001" <sip:[email protected]>;tag=5317D4-2271
    To: <sip:[email protected]>
    Date: Thu, 20 Feb 2014 10:55:56 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces,sdp-anat
    Min-SE: 1800
    Cisco-Guid: 1679566433-2572423651-2156454406-1292596908
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1392893756
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 208
    <--- Reliably Transmitting (no NAT) to 10.0.70.11:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 10.0.70.11:5060;branch=z9hG4bK3E205D;received=10.0.70.11
    From: "DN1001" <sip:[email protected]>;tag=5317D4-2271
    To: <sip:[email protected]>;tag=as665c9410
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Server: Asterisk PBX 11.7.0
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="559bd1d2"
    Content-Length: 0
    However, when call is for ASTERISK to Cisco, there is no challenge sent.
    INVITE sip:[email protected] SIP/2.0
    Via: SIP/2.0/UDP 10.1.32.70:5060;branch=z9hG4bK0c57d67c
    Max-Forwards: 70
    From: "JOSE MANUEL" <sip:[email protected]>;tag=as2f789a9f
    To: <sip:[email protected]>
    Contact: <sip:[email protected]:5060>
    Call-ID: [email protected]:5060
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 11.7.0
    Date: Thu, 20 Feb 2014 09:58:27 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 282
    <--- SIP read from UDP:10.0.70.11:60829 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.1.32.70:5060;branch=z9hG4bK0c57d67c
    From: "JOSE MANUEL" <sip:[email protected]>;tag=as2f789a9f
    To: <sip:[email protected]>
    Date: Thu, 20 Feb 2014 10:58:27 GMT
    Call-ID: [email protected]:5060
    CSeq: 102 INVITE
    Allow-Events: telephone-event
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 10.1.32.70:5060;branch=z9hG4bK0c57d67c
    From: "JOSE MANUEL" <sip:[email protected]>;tag=as2f789a9f
    To: <sip:[email protected]>;tag=556830-757
    Date: Thu, 20 Feb 2014 10:58:27 GMT
    Call-ID: [email protected]:5060
    CSeq: 102 INVITE
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    Allow-Events: telephone-event
    Remote-Party-ID: "DN1001" <sip:[email protected]>;party=called;screen=no;privacy=off
    Contact: <sip:[email protected]:5060>
    Server: Cisco-SIPGateway/IOS-12.x
    Content-Length: 0
    My configuration in Cisco device is:
    dial-peer voice 1 voip
    description **Calls to ASTERISK **
    destination-pattern 9T
    session protocol sipv2
    session target sip-server
    codec g711ulaw
    sip-ua
    keepalive target ipv4:10.1.32.70
    authentication username CCME password 7 070E234F4A realm asterisk
      sip-server ipv4:10.1.32.70:5060
    To avoid that the ASTERISK is blocked by Cisco TOLLFRAUD_APP I have added:
    voice service voip
    ip address trusted list
      ipv4 10.1.32.70 255.255.255.255
    allow-connections sip to sip
    sip
      registrar server
    The issue is that I would like that Cisco also send a challenge to asterisk server to authenticate SIP messages.
    Any ideas?.
    Regards.

    Hello,
    yes, but credentials command configure credentials that are used when Cisco UA must register in a server.
    I do not need register Cisco into Asterisk server. What I want is that Cisco authenticate SIP messages that receive. I know
    that can be enough with TOLLFRAUD_AP where remote IP is checked, but I want to do something like others routing
    protocols (as OSPF, BGP) where every message must be authenticated.
    Thanks.
    Regards.

  • I have 4 email accounts in apple Mail.  My mail is sending from the wrong account.  even on "reply" it sends from a different account.  I have 3 gmail accounts and one exchange server account. I choose which account to send from and it still sends from a

    I have 4 email accounts in apple Mail.  My mail is sending from the wrong account.  even on "reply" it sends from a different account.  I have 3 gmail accounts and one exchange server account. I choose which account to send from and it still sends from the same gmail account.  Help.

    I HAD two accounts because of this problem.  I completely removed the problem account from the Apple Mail client.  Guess what is happening??  That's right - Mail is still sending from the other account that no longer exists on my computer, and I have absolutely no idea how this is happening.  This is incredibly frustrating.  When a recipient chooses to reply to my message, quite often I won't get it now because it is going to the other account that has now been deleted from my system.  COME ON APPLE!!! WHAT'S THE DEAL WITH THIS???

  • Script to choose which account mail is to be sent from in Mail?

    Does anyone have an AppleScript which will automatically set the default Mail account to an account chosen in script?
    I use FileMaker to send mail automatically, but have to use Eudora because it makes it easy to auto reset the default account from which the mail to be sent before FileMaker actually sends the mail.
    Mail always uses either the first account in list of accounts (if none chosen) or the chosen account if one is chosen. What I need is a script that "chooses the account" before I have FileMaker send mail.

    Hello, I have a similar problem. When Thunderbird is set as the Default Email program in the Preferences (instead of Apple Mail) and when I click "send email" in Filemaker Tasks or Meetings, nothing happens. No email windows. No email programs started. Zippo. However, when Apple Mail is set as the default email program, then it works fine (aside from the fact that I lose all formatting from my message that exists in Filemaker). I find this annoying. Any ideas?
    Thanks, Jeff

  • There are an option that let us on choosing an browser in that will be open the Adobe Account  from  app-desktop Creative Cloud in the windows operating system?

    Hello,
    There are an option that let us on choosing an browser in that will be open the Adobe Account  from  app-desktop Creative Cloud in the windows operating system?
    ...in the case this option  not exist...
    Please add an option (maybe in next update schedule) that will let us to choose a different browser not the default.
    Adobe Cloud ,Behance also Prosite are very weak implemented on the Internet Explorer.
    All the best,
    Cristian

    Hi Jeff,
    Thank you for the attention given! I know this ... CCloud,Behance,Prosite go very well with Chrome;Firefox browsers(I use also on this browsers).
    In case that will set Chrome or Firefox as the default browser I'll lose modern (metro) Internet Explorer... I don't want to loose this browser because is using on different stuff...a taste option.
    I try question more like an suggestion to Adobe (maybe a future thinking option) but my English writes are weak ..
    All the beast,
    Cristian

  • Another Error 8 - but a test account from Ralph's site works....any Ideas??

    iChat Video or Audio: This has yet to work since buying this iMac on the release date 8/7/2007. Person on the other end is a AIM account. The funny thing is I can connect to a test account that Ralph had turned me onto on this forum. Very strange. I would think if Video works for one I should be able to connect with another AIM user.
    Date/Time: 2007-08-22 11:21:23.714 -0400
    OS Version: 10.4.10 (Build 8R4031)
    Report Version: 4
    iChat Connection Log:
    AVChat started with ID 0.
    0x153e8740: State change from AVChatNoState to AVChatStateWaiting.
    [email protected]: State change from AVChatNoState to AVChatStateInvited.
    0x153e8740: State change from AVChatStateWaiting to AVChatStateConnecting.
    [email protected]: State change from AVChatStateInvited to AVChatStateConnecting.
    0x153e8740: State change from AVChatStateConnecting to AVChatStateEnded.
    Chat ended with error -8
    [email protected]: State change from AVChatStateConnecting to AVChatStateEnded.
    Chat ended with error -8
    Video Conference Error Report:
    @:0 type=4 (00000000/22)
    [VCSIP_INVITEERROR]
    [19]
    @SIP/SIP.c:2448 type=4 (900A0015/22)
    [SIPConnectIPPort failed]
    @SIP/SIP.c:2448 type=4 (900A0015/22)
    [SIPConnectIPPort failed]
    Video Conference Support Report:
    @SIP/Transport.c:1218 type=1 (00000000/0)
    [INVITE sip:[email protected]:5061 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK4983d4066c6878c0
    Max-Forwards: 70
    To: "u0" <sip:[email protected]:5061>
    From: "[email protected]" <sip:[email protected]>;tag=2125523722
    Call-ID: 5277c4a2-50c3-11dc-8b12-ea927bfc13c4@lip
    CSeq: 1 INVITE
    Contact: <sip:[email protected]>;isfocus
    User-Agent: Viceroy 1.2
    Content-Type: application/sdp
    Content-Length: 512
    v=0
    o=vlebass 0 0 IN IP4 192.168.1.3
    [email protected]
    c=IN IP4 192.168.1.3
    b=AS:2147483647
    t=0 0
    a=hwi:1056:2:2400
    a=bandwidthDetection:YES
    a=iChatEncryption:NO
    m=audio 16386 RTP/AVP 12 3 0
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:-1395227391
    m=video 16384 RTP/AVP 126 34
    a=rtpmap:126 X-H264/90000
    a=rtpmap:34 H263/90000
    a=fmtp:34 imagesize 1 rules 30:352:288
    a=framerate:20
    a=RTCP:AUDIO 16387 VIDEO 16385
    a=pogo
    a=fmtp:126 imagesize 0 rules 20:640:480:640:480
    a=rtpID:-1827866006
    @SIP/Transport.c:1218 type=1 (00000000/0)
    [INVITE sip:[email protected]:5061 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK4983d4066c6878c0
    Max-Forwards: 70
    To: "u0" <sip:[email protected]:5061>
    From: "[email protected]" <sip:[email protected]>;tag=2125523722
    Call-ID: 5277c4a2-50c3-11dc-8b12-ea927bfc13c4@lip
    CSeq: 1 INVITE
    Contact: <sip:[email protected]>;isfocus
    User-Agent: Viceroy 1.2
    Content-Type: application/sdp
    Content-Length: 512
    v=0
    o=vlebass 0 0 IN IP4 192.168.1.3
    [email protected]
    c=IN IP4 192.168.1.3
    b=AS:2147483647
    t=0 0
    a=hwi:1056:2:2400
    a=bandwidthDetection:YES
    a=iChatEncryption:NO
    m=audio 16386 RTP/AVP 12 3 0
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:-1395227391
    m=video 16384 RTP/AVP 126 34
    a=rtpmap:126 X-H264/90000
    a=rtpmap:34 H263/90000
    a=fmtp:34 imagesize 1 rules 30:352:288
    a=framerate:20
    a=RTCP:AUDIO 16387 VIDEO 16385
    a=pogo
    a=fmtp:126 imagesize 0 rules 20:640:480:640:480
    a=rtpID:-1827866006
    @SIP/Transport.c:1218 type=1 (00000000/0)
    [INVITE sip:[email protected]:5061 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK4983d4066c6878c0
    Max-Forwards: 70
    To: "u0" <sip:[email protected]:5061>
    From: "[email protected]" <sip:[email protected]>;tag=2125523722
    Call-ID: 5277c4a2-50c3-11dc-8b12-ea927bfc13c4@lip
    CSeq: 1 INVITE
    Contact: <sip:[email protected]>;isfocus
    User-Agent: Viceroy 1.2
    Content-Type: application/sdp
    Content-Length: 512
    v=0
    o=vlebass 0 0 IN IP4 192.168.1.3
    [email protected]
    c=IN IP4 192.168.1.3
    b=AS:2147483647
    t=0 0
    a=hwi:1056:2:2400
    a=bandwidthDetection:YES
    a=iChatEncryption:NO
    m=audio 16386 RTP/AVP 12 3 0
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:-1395227391
    m=video 16384 RTP/AVP 126 34
    a=rtpmap:126 X-H264/90000
    a=rtpmap:34 H263/90000
    a=fmtp:34 imagesize 1 rules 30:352:288
    a=framerate:20
    a=RTCP:AUDIO 16387 VIDEO 16385
    a=pogo
    a=fmtp:126 imagesize 0 rules 20:640:480:640:480
    a=rtpID:-1827866006
    @SIP/Transport.c:1218 type=1 (00000000/0)
    [INVITE sip:[email protected]:5061 SIP/2.0
    Via: SIP/2.0/UDP m.0;branch=z9hG4bK2d8a7796024f218a
    Max-Forwards: 70
    To: "u0" <sip:[email protected]:5061>
    From: "[email protected]" <sip:[email protected]>;tag=279800261
    Call-ID: 5146a436-50c3-11dc-8b12-bb2eb7eb13c4@lip
    CSeq: 1 INVITE
    Contact: <sip:[email protected]>;isfocus
    User-Agent: Viceroy 1.2
    Content-Type: application/sdp
    Content-Length: 514
    v=0
    o=vlebass 0 0 IN IP4 m.0
    [email protected]
    c=IN IP4 m.0
    b=AS:2147483647
    t=0 0
    a=hwi:1056:2:2400
    a=bandwidthDetection:YES
    a=iChatEncryption:NO
    m=audio 16386 RTP/AVP 12 3 0
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:-1395227391
    m=video 16384 RTP/AVP 126 34
    a=rtpmap:126 X-H264/90000
    a=rtpmap:34 H263/90000
    a=fmtp:34 imagesize 1 rules 30:352:288
    a=framerate:20
    a=RTCP:AUDIO 16387 VIDEO 16385
    a=pogo
    a=fmtp:126 imagesize 0 rules 20:640:480:640:480
    a=rtpID:-1827866006
    @SIP/Transport.c:1218 type=1 (00000000/0)
    [INVITE sip:[email protected]:5061 SIP/2.0
    Via: SIP/2.0/UDP m.0;branch=z9hG4bK2d8a7796024f218a
    Max-Forwards: 70
    To: "u0" <sip:[email protected]:5061>
    From: "[email protected]" <sip:[email protected]>;tag=279800261
    Call-ID: 5146a436-50c3-11dc-8b12-bb2eb7eb13c4@lip
    CSeq: 1 INVITE
    Contact: <sip:[email protected]>;isfocus
    User-Agent: Viceroy 1.2
    Content-Type: application/sdp
    Content-Length: 514
    v=0
    o=vlebass 0 0 IN IP4 m.0
    [email protected]
    c=IN IP4 m.0
    b=AS:2147483647
    t=0 0
    a=hwi:1056:2:2400
    a=bandwidthDetection:YES
    a=iChatEncryption:NO
    m=audio 16386 RTP/AVP 12 3 0
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:-1395227391
    m=video 16384 RTP/AVP 126 34
    a=rtpmap:126 X-H264/90000
    a=rtpmap:34 H263/90000
    a=fmtp:34 imagesize 1 rules 30:352:288
    a=framerate:20
    a=RTCP:AUDIO 16387 VIDEO 16385
    a=pogo
    a=fmtp:126 imagesize 0 rules 20:640:480:640:480
    a=rtpID:-1827866006
    @SIP/Transport.c:1218 type=1 (00000000/0)
    [INVITE sip:[email protected]:5061 SIP/2.0
    Via: SIP/2.0/UDP m.0;branch=z9hG4bK2d8a7796024f218a
    Max-Forwards: 70
    To: "u0" <sip:[email protected]:5061>
    From: "[email protected]" <sip:[email protected]>;tag=279800261
    Call-ID: 5146a436-50c3-11dc-8b12-bb2eb7eb13c4@lip
    CSeq: 1 INVITE
    Contact: <sip:[email protected]>;isfocus
    User-Agent: Viceroy 1.2
    Content-Type: application/sdp
    Content-Length: 514
    v=0
    o=vlebass 0 0 IN IP4 m.0
    [email protected]
    c=IN IP4 m.0
    b=AS:2147483647
    t=0 0
    a=hwi:1056:2:2400
    a=bandwidthDetection:YES
    a=iChatEncryption:NO
    m=audio 16386 RTP/AVP 12 3 0
    a=rtpmap:3 GSM/8000
    a=rtpmap:0 PCMU/8000
    a=rtpID:-1395227391
    m=video 16384 RTP/AVP 126 34
    a=rtpmap:126 X-H264/90000
    a=rtpmap:34 H263/90000
    a=fmtp:34 imagesize 1 rules 30:352:288
    a=framerate:20
    a=RTCP:AUDIO 16387 VIDEO 16385
    a=pogo
    a=fmtp:126 imagesize 0 rules 20:640:480:640:480
    a=rtpID:-1827866006
    @:0 type=2 (00000000/22)
    [VCVIDEO_OUTGOINGATTEMPT]
    [4]
    Video Conference User Report:
    Binary Images Description for "iChat":
    0x1000 - 0x17dfff com.apple.iChat 3.1.8 (445) /Applications/iChat.app/Contents/MacOS/iChat
    0x14c6e000 - 0x14c77fff com.apple.IOFWDVComponents 1.9.0 /System/Library/Components/IOFWDVComponents.component/Contents/MacOS/IOFWDVComp onents
    0x14cfc000 - 0x14d38fff com.apple.QuickTimeFireWireDV.component 7.2 /System/Library/QuickTime/QuickTimeFireWireDV.component/Contents/MacOS/QuickTim eFireWireDV
    0x14d44000 - 0x14d74fff com.apple.QuickTimeIIDCDigitizer 7.2 /System/Library/QuickTime/QuickTimeIIDCDigitizer.component/Contents/MacOS/Quick TimeIIDCDigitizer
    0x14d7e000 - 0x14dc7fff com.apple.QuickTimeUSBVDCDigitizer 2.0.5 /System/Library/QuickTime/QuickTimeUSBVDCDigitizer.component/Contents/MacOS/Qui ckTimeUSBVDCDigitizer
    0x14df1000 - 0x14df6fff com.apple.audio.AppleHDAHALPlugIn 1.3.6 (1.3.6a8) /System/Library/Extensions/AppleHDA.kext/Contents/PlugIns/AppleHDAHALPlugIn.bun dle/Contents/MacOS/AppleHDAHALPlugIn
    0x14e05000 - 0x14f6efff com.apple.opengl 1.5.1 /System/Library/Frameworks/OpenGL.framework/Resources/GLEngine.bundle/GLEngine
    0x14f9c000 - 0x151cdfff com.apple.ATIRadeonX2000GLDriver 1.5.6 (5.0.6) /System/Library/Extensions/ATIRadeonX2000GLDriver.bundle/Contents/MacOS/ATIRade onX2000GLDriver
    0x15215000 - 0x15232fff com.apple.opengl 1.5.1 /System/Library/Frameworks/OpenGL.framework/Versions/A/Resources/GLRendererFloa t.bundle/GLRendererFloat
    0x1657b000 - 0x1657efff com.apple.iokit.IOQTComponents 1.4 /System/Library/Components/IOQTComponents.component/Contents/MacOS/IOQTComponen ts
    0x1676d000 - 0x16770fff com.apple.audio.AudioIPCPlugIn 1.0.2 /System/Library/Extensions/AudioIPCDriver.kext/Contents/Resources/AudioIPCPlugI n.bundle/Contents/MacOS/AudioIPCPlugIn
    0x1678c000 - 0x167b6fff com.apple.audio.SoundManager.Components 3.9.2 /System/Library/Components/SoundManagerComponents.component/Contents/MacOS/Soun dManagerComponents
    0x173bb000 - 0x173c7fff com.apple.airport.diskmenu 1.2.1 (121.4) /System/Library/Contextual Menu Items/AirPort Disk Menu.plugin/Contents/MacOS/AirPort Disk Menu
    0x173cc000 - 0x173cefff com.apple.AutomatorCMM 1.0.1 (87) /System/Library/Contextual Menu Items/AutomatorCMM.plugin/Contents/MacOS/AutomatorCMM
    0x173e1000 - 0x173e5fff com.apple.FolderActionsMenu 1.3.1 /System/Library/Contextual Menu Items/FolderActionsMenu.plugin/Contents/MacOS/FolderActionsMenu
    0x173ea000 - 0x173eafff com.apple.SpotLightCM 1.0 (121.36) /System/Library/Contextual Menu Items/SpotlightCM.plugin/Contents/MacOS/SpotlightCM
    0x8fe00000 - 0x8fe4afff dyld /usr/lib/dyld
    0x90000000 - 0x90171fff libSystem.B.dylib /usr/lib/libSystem.B.dylib
    0x901c1000 - 0x901c3fff libmathCommon.A.dylib /usr/lib/system/libmathCommon.A.dylib
    0x901c5000 - 0x90202fff com.apple.CoreText 1.1.2 (???) /System/Library/Frameworks/ApplicationServices.framework/Versions/A/Frameworks/ CoreText.framework/Versions/A/CoreText
    0x90229000 - 0x902fffff com.apple.ApplicationServices.ATS 2.0.6 (???) /System/Library/Frameworks/ApplicationServices.framework/Versions/A/Frameworks/ ATS.framework/Versions/A/ATS
    0x9031f000 - 0x90774fff com.apple.CoreGraphics 1.258.77 (???) /System/Library/Frameworks/ApplicationServices.framework/Versions/A/Frameworks/ CoreGraphics.framework/Versions/A/CoreGraphics
    0x9080b000 - 0x908d3fff com.apple.CoreFoundation 6.4.7 (368.28) /System/Library/Frameworks/CoreFoundation.framework/Versions/A/CoreFoundation
    0x90911000 - 0x90911fff com.apple.CoreServices 10.4 (???) /System/Library/Frameworks/CoreServices.framework/Versions/A/CoreServices
    0x90913000 - 0x90a07fff libicucore.A.dylib /usr/lib/libicucore.A.dylib
    0x90a57000 - 0x90ad6fff libobjc.A.dylib /usr/lib/libobjc.A.dylib
    0x90aff000 - 0x90b63fff libstdc++.6.dylib /usr/lib/libstdc++.6.dylib
    0x90bd2000 - 0x90bd9fff libgcc_s.1.dylib /usr/lib/libgcc_s.1.dylib
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    0x91ce2000 - 0x91d70fff com.apple.vImage 2.5 /System/Library/Frameworks/Accelerate.framework/Versions/A/Frameworks/vImage.fr amework/Versions/A/vImage
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    0x9271a000 - 0x92797fff com.apple.DesktopServices 1.3.6 /System/Library/PrivateFrameworks/DesktopServicesPriv.framework/Versions/A/Desk topServicesPriv
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    0x92b49000 - 0x92ba1fff com.apple.opengl 1.5.1 /System/Library/Frameworks/OpenGL.framework/Versions/A/Libraries/libGLU.dylib
    0x92bb5000 - 0x92bb5fff com.apple.Carbon 10.4 (???) /System/Library/Frameworks/Carbon.framework/Versions/A/Carbon
    0x92bb7000 - 0x92bc7fff com.apple.ImageCapture 3.0.4 /System/Library/Frameworks/Carbon.framework/Versions/A/Frameworks/ImageCapture. framework/Versions/A/ImageCapture
    0x92bd6000 - 0x92bdefff com.apple.speech.recognition.framework 3.6 /System/Library/Frameworks/Carbon.framework/Versions/A/Frameworks/SpeechRecogni tion.framework/Versions/A/SpeechRecognition
    0x92be4000 - 0x92beafff com.apple.securityhi 2.0.1 (24742) /System/Library/Frameworks/Carbon.framework/Versions/A/Frameworks/SecurityHI.fr amework/Versions/A/SecurityHI
    0x92bf0000 - 0x92c81fff com.apple.ink.framework 101.2.1 (71) /System/Library/Frameworks/Carbon.framework/Versions/A/Frameworks/Ink.framework /Versions/A/Ink
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    0x92c9c000 - 0x92cbafff com.apple.openscripting 1.2.5 (???) /System/Library/Frameworks/Carbon.framework/Versions/A/Frameworks/OpenScripting .framework/Versions/A/OpenScripting
    0x92ccc000 - 0x92cd2fff com.apple.print.framework.Print 5.2 (192.4) /System/Library/Frameworks/Carbon.framework/Versions/A/Frameworks/Print.framewo rk/Versions/A/Print
    0x92cd8000 - 0x92d3bfff com.apple.htmlrendering 66.1 (1.1.3) /System/Library/Frameworks/Carbon.framework/Versions/A/Frameworks/HTMLRendering .framework/Versions/A/HTMLRendering
    0x92d62000 - 0x92da3fff com.apple.NavigationServices 3.4.4 (3.4.3) /System/Library/Frameworks/Carbon.framework/Versions/A/Frameworks/NavigationSer vices.framework/Versions/A/NavigationServices
    0x92dca000 - 0x92dd8fff com.apple.audio.SoundManager 3.9.1 /System/Library/Frameworks/Carbon.framework/Versions/A/Frameworks/CarbonSound.f ramework/Versions/A/CarbonSound
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    0x92de9000 - 0x930defff com.apple.HIToolbox 1.4.9 (???) /System/Library/Frameworks/Carbon.framework/Versions/A/Frameworks/HIToolbox.fra mework/Versions/A/HIToolbox
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    0x931f5000 - 0x93210fff com.apple.DirectoryService.Framework 3.3 /System/Library/Frameworks/DirectoryService.framework/Versions/A/DirectoryServi ce
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    0x93262000 - 0x93918fff com.apple.AppKit 6.4.9 (824.44) /System/Library/Frameworks/AppKit.framework/Versions/C/AppKit
    0x93c99000 - 0x93d14fff com.apple.CoreData 91 (92.1) /System/Library/Frameworks/CoreData.framework/Versions/A/CoreData
    0x93d4d000 - 0x93e07fff com.apple.audio.toolbox.AudioToolbox 1.4.5 /System/Library/Frameworks/AudioToolbox.framework/Versions/A/AudioToolbox
    0x93e4a000 - 0x93e4afff com.apple.audio.units.AudioUnit 1.4.3 /System/Library/Frameworks/AudioUnit.framework/Versions/A/AudioUnit
    0x93e4c000 - 0x9400dfff com.apple.QuartzCore 1.4.12 /System/Library/Frameworks/QuartzCore.framework/Versions/A/QuartzCore
    0x94053000 - 0x94094fff libsqlite3.0.dylib /usr/lib/libsqlite3.0.dylib
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    0x94d5c000 - 0x94ea2fff com.apple.AddressBook.framework 4.0.5 (487) /System/Library/Frameworks/AddressBook.framework/Versions/A/AddressBook
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    0x94fb7000 - 0x94fd4fff libresolv.9.dylib /usr/lib/libresolv.9.dylib
    0x94fdb000 - 0x95041fff com.apple.Bluetooth 1.9.1 (1.9.1f18) /System/Library/Frameworks/IOBluetooth.framework/Versions/A/IOBluetooth
    0x95300000 - 0x95393fff com.apple.WebKit 419.3 /System/Library/Frameworks/WebKit.framework/Versions/A/WebKit
    0x953ed000 - 0x9546dfff com.apple.JavaScriptCore 418.6.1 /System/Library/Frameworks/WebKit.framework/Versions/A/Frameworks/JavaScriptCor e.framework/Versions/A/JavaScriptCore
    0x954a6000 - 0x95785fff com.apple.WebCore 418.23 /System/Library/Frameworks/WebKit.framework/Versions/A/Frameworks/WebCore.frame work/Versions/A/WebCore
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    0x96ba0000 - 0x96ba0fff com.apple.vecLib 3.3.1 (vecLib 3.3.1) /System/Library/Frameworks/vecLib.framework/Versions/A/vecLib
    0x97087000 - 0x970a9fff com.apple.speech.LatentSemanticMappingFramework 2.5 /System/Library/PrivateFrameworks/LatentSemanticMapping.framework/Versions/A/La tentSemanticMapping
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    0x97286000 - 0x97286fff com.apple.MonitorPanelFramework 1.1.1 /System/Library/PrivateFrameworks/MonitorPanel.framework/Versions/A/MonitorPane l
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    0x98216000 - 0x98218fff com.apple.DisplayServicesFW 1.8.2 /System/Library/PrivateFrameworks/DisplayServices.framework/Versions/A/DisplayS ervices
    0x98445000 - 0x992abfff com.apple.QuickTimeComponents.component 7.2 /System/Library/QuickTime/QuickTimeComponents.component/Contents/MacOS/QuickTim eComponents
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    0x9a012000 - 0x9a17efff com.apple.MessageFramework 2.1.1 (752.3) /System/Library/Frameworks/Message.framework/Versions/B/Message

    The log does suggest the problem lies at the other end as the INVITE is not answered.
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    5:20 PM Thursday; August 23, 2007

  • How do I unlink an iCloud account from an iTunes one?

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  • Project server 2010 don't respect the 'Choose an account' option

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