5530 How to use SIP

I did all the settings to use SIP, the last step is to make a internetphone link.
How can i make this internet phone link so i can make calls with SIP ?
Thx.
Realblackadder

From your error messages to your output parameters, changing the colors of different PowerShell variables and outcomes is a straightforward and useful way to customize your PowerShell console. You can quickly identify ouputs, error messages, parameters and more with simple color cues – and when you're dealing with a longer script, being able to quickly identify anything is a huge help.In aguide on how to configure and customize your color settingsin PowerShell, Petri walks through each and every step, and even donates some of itsown custom PowerShell color themes scripts. (It should be noted that this only works for PowerShell console and not PowerShell ISE.) You can access your color configuration by entering:Powershell$host.privatedataThis will pull up a list of all your colors as they are currently set.Image credit:Jeff HicksNow...

Similar Messages

  • X5800 how to use SIP and VOIP

    Hello, I have a X5800 music, model 5800d-1 and I can enter all the necessary SIP data. I get the message (settings, connectivity, admin settings, sip settings) registered which is fine. My provider is freephonie. Question: 1) Although on the freebox the SIP is activated, I don't receive incoming call on my nokia. 2) How do I tell my nokia to call on the internet ? Is it possible only using SIP tel numbers which I can enter in the contacts list? Help much appreciated. Pierre.

    SIP and VoIP are not synonymous. VoIP can indeed use SIP (and often does), but other services, such as video sharing for example, also use SIP.
    It so happens that the 5800 has no native support for VoIP, which is why you can't make SIP calls using it.
    Your only solution for SIP is to use third-party services such as fring, for example. Once logged into your fring account you can subscribe to a SIP service. Fring's systems will register to the SIP account and relay the comminucation through your fring account. There's lag and the audio quality isn't as good as a direct SIP connection but at least it works.
    The alternative is to use a phone that supports VoIP/SIP natively.
    Was this post helpful? If so, please click on the white "Kudos!" star below. Thank you!

  • N85-how to use SIP setting and use Voice over IP t...

    Dear Nokia user
    In Nokia N85 setting there is an option foe SIP setting.I want to know how can I use this option to register in a SIP proxy server and have an VOIP converstion over internet by my N85
    small_nokia_supporter

    Dear ngnok( nokia advisor)
    thank you for your reply.I went to the link you had given in your reply to download SIP Voip 3x setting but I were asked  to relog and after I entered my username and password . I saw this bloking  message" The credentialsyou provided can not be determined to be authentic".
    I had not claimed  any proficiency befor based on wich you can tell its authentic or not.Please give me the requirment which I provide you to have access to such technical tools.
    Sincerely yours
    small_nokia_supporter

  • HOW TO: Use SIP Trunking to Keep your Telecoms System Up and Running

    Hello guys,
    Here's the deal:
    We had a major HDD fault in a laptop the other day... Pretty messy, (the laptop
    was our chief accountant's, and she's a paranoid x) ) as most of the data on
    the hard drive was invaluable. Using a recovery software, however, we've
    managed to get out about 80% of the info and transfered it to a mobile HDD.
    Almost everything is alright, except for a few xls/doc files that seem to be
    badly damaged. I want to ask if it's possible to somehow repair these files. I
    did try few repair programs, but none of them seems to even detect that the
    files are damaged. I've attached one of these files, please take a look and
    tell me if there's any hope (personally I think it's irreparable, but
    still...).

    From your error messages to your output parameters, changing the colors of different PowerShell variables and outcomes is a straightforward and useful way to customize your PowerShell console. You can quickly identify ouputs, error messages, parameters and more with simple color cues – and when you're dealing with a longer script, being able to quickly identify anything is a huge help.In aguide on how to configure and customize your color settingsin PowerShell, Petri walks through each and every step, and even donates some of itsown custom PowerShell color themes scripts. (It should be noted that this only works for PowerShell console and not PowerShell ISE.) You can access your color configuration by entering:Powershell$host.privatedataThis will pull up a list of all your colors as they are currently set.Image credit:Jeff HicksNow...

  • How to use TSP with CME9.1 + SIP Phones + EM

    Hi,
    can anyone explain me how to use the following setup:
    - Cisco CME 9.1
    - IPPhone 8961, 9971, 6921 all connected via SIP
    - Extension Mobility
    How can i use the Cisco TSP on my Windows 7 PC to dial via Outlook and so on?
    Kind regards
    Andre Lahl

    Hi there,
    is TSP on CME 10.0 with 8961 still not supported?
    kind regards,
    Norbert

  • WIP310-G2: how to connect to an specific VoIP provider using SIP ?

    Dear all,
    I just received a WIP310-G2 and I am wondering how -or whether- I can configure it to connect to an specific Internet based VoIP provider using SIP
    Cant seem to find how to do it....
    Thanks a lot,
    Alvaro

    You connect your WIP310 to your PC using an USB cable. Then lookup the IP-adresse on the phone and enter that into a browser.
    In the UI select Admin Logon and you can enter all the configuration details for your VoIP provider. I would recommned that you ask your VoIP provider to help you with a guide.
    I used a PAP2T configuration example that worked for my WIP310, which I got from my VoIP provider.

  • How to use linphone - open sip VoIP library in Windows desktop application.?

    I want to develop an VoIP .net application in C# lang for windows desktop. Does anyone know any open sip VoIP library to use.? I know about Linphone. but it is available for Android, iOS and Windows(app developed using GTK- UI builder). If anyone knows how
    to use Linphone for windows then ans also will be appreciated.
    I am using MS visual studio for application development. Tell me its compiling steps as well. How to use that in WPF or WinForms.?
    Thanks in advance.

    Hello,
    The VoIP seems not MS product and can you clarify it? Basically, if it is third party product, you will need to ask for the API writer for details. For example, find a forum here:
    http://www.linphone.org/
    Regards,
    Barry Wang
    We are trying to better understand customer views on social support experience, so your participation in this interview project would be greatly appreciated if you have time. Thanks for helping make community forums a great place.
    Click
    HERE to participate the survey.

  • How to develop VoIP client using SIP in J2ME?

    Hi Everybody,
    I want to develop a VoIP client in J2ME that connects to asterisk server of debian and can call to the registered user of asterisk server and can have a telephonic talk session easily.
    Do anybody have idea regarding the development of the client or having tutorial that teaches the development of VoIP in J2ME or in any other way.?
    PLZ help me to provide the solution.
    Thanks in anticipation.
    with regards,
    KHAKHAR SAGAR

    Hi
    I am interested about developing VoIP application (using SIP) in J2ME platform. But I am stuck with the problem of MMAPI. Without using MMAPI J2ME has no access to mobile media devices, such as speaker or microphone, and without creating a player MMAPI can't play media data, such as sound or video. But its not possible to record voice and play voice data simultaneously using player in J2ME. So it seems almost impossible to implement VoIP application maintaining all its constraints and requirements, specially in case of delay and jitter.
    I am looking for some solution, which will provide the ability to overcome this problem. I come out with two possible solutions, but not sure about their out come. If we can develop a native media application, we can have access to it by using KNI (K Native Interface). In that way we can take some risk to develop VoIP application for J2ME. My another solution is, we can handle the player using MMAPI to record and play voice data in mill second level, so that we can have a real time feeling, though I am not sure if its possible by using RTSP.
    If any one have solution of this problem, please help us.
    Reagards
    Asif Mohammed Adnan

  • Issue with instant ringback when using sip trunk to SP

    Hi all,
    We use CUCM 8.0.2.
    We have a SIP trunk to a SP connected via one of our Cisco 2911 routers configured as a CUBE.
    Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.0(1)M3, RELEASE SOFTWARE (fc2)
    c2900-universalk9-mz.SPA.150-1.M3.bin
    Cisco CISCO2911/K9 (revision 1.0)
    Technology Package License Information for Module:'c2900'
    Technology Technology-package
                      Current       Type
    ipbase        ipbasek9      Permanent
    security      securityk9    Permanent
    uc              uck9            Permanent
    data           None            None
    We also have several ISDN lines that run out via various Cisco routers configured as H323 gateways.
    We use 7945 and CIPC for our phones.
    We're having an issue with calls going via the SIP trunk where we hear ringing instantly after dialling - but before the actual device at the other end starts ringing (considerable difference).
    Using the SIP trunk: If I make a call to my mobile phone - I hear ringing instantly - about 3 rings before my mobile phone actually starts ringing - undesireable.
    Using H323 gateway: If I make a call to my mobile phone - I hear silence for a bit - then ringing when the mobile starts ringing - desired.
    Using SIP trunk: If I make a call to a landline that is ready - it rings instantly for at least 1 ring - before the actual phone I'm calling starts ringing - undesireable.
    Using H323 gateway: There is a momentary pause before hearing ringing on my phone and the phone I dialled - desired.
    Using SIP trunk: If I make a call to a landline that is off-hook (with no call-waiting/etc.) - it rings once and then returns the busy signal (the worst issue) - undesireable.
    Using H323 gateway: There is a momentary pause before hearing busy signal - desired.
    Phone to phone internally (same network): Operates as expected (instantly rings locally and on the phone I'm calling). Between phones that utilise the SIP trunk and phones that utilise the H323 gateways within the same network - communication is instant and as expected.
    Any ideas why this happens and how to stop it?
    I want it to not ring until the situation is known and that it can provide the appropriate feedback (ringing/busy/etc.).
    Some possibly relevant config (note that there is a known bug with this IOS that meant I had to declare the codec in each dial-peer as the voice class would not work):
    voice service voip
    address-hiding
    mode border-element
    allow-connections sip to sip
    sip
      bind control source-interface GigabitEthernet0/0
      bind media source-interface GigabitEthernet0/0
      header-passing error-passthru
      early-offer forced
      midcall-signaling passthru
    interface GigabitEthernet0/0
    ip address x.x.x.x 255.255.255.252
    ip access-group acl.SIP-IN in
    no ip redirects
    no ip unreachables
    ip verify unicast reverse-path
    ip virtual-reassembly
    duplex full
    speed 100
    no cdp enable
    gateway
    timer receive-rtp 1200
    sip-ua
    connection-reuse
    gatekeeper
    shutdown
    dial-peer voice 1 voip
    description *** INBOUND CALLS FROM CARRIER ***
    translation-profile incoming SIPTRUNK-INCOMING
    session protocol sipv2
    incoming called-number #blah blah#
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    dial-peer voice 61 voip
    description **** WA, SA AND NT NUMBERS ****
    destination-pattern 0[8]........
    session protocol sipv2
    session target ipv4:<MY SP's SIP SERVER>
    incoming called-number 0[8]........
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    dial-peer voice 81 voip
    description **** MOBILE NUMBERS ****
    destination-pattern 0[4]........
    session protocol sipv2
    session target ipv4:<MY SP's SIP SERVER>
    incoming called-number 0[4]........
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    dial-peer voice 500 voip
    description *** INBOUND SIP TRUNK TO CUCM PUB ***
    translation-profile outgoing SIPTRUNK-CALLING-ADD-0
    preference 1
    destination-pattern 5[12]..
    session protocol sipv2
    session target ipv4:<OUR CUCM PUBLISHER IP>
    dtmf-relay rtp-nte
    codec g711alaw
    ip qos dscp cs5 media
    no vad
    Any help or a point in the right direction would be greatly appreciated.
    Cheers,
    Brett

    I ended up resolving this issue as follows:
    In CUCM, under Device > Device Settings > SIP Profile.
    I modifed the profile relevant to my SIP trunk, under the "Trunk Specific Configuration", I set "SIP Rel1XX Options" from "Disabled" to "Send PRACK if 1xx Contains SDP".
    Now, I get the expected delay before hearing ringback.
    Solved!

  • PAP2T - how to setup SIP proxy and codec on PAP2T

    Hello
    How to setup SIP Proxy server IP addres and codec settings on PAP2T ?

    Your question here appears related to or a continuation of this previous question — is this the case?
    Based on the previous posting, whatever information you're receiving from the vendor support folks appears rather garbled or confused, particularly based on that "The vendor mentioned that we need to route the DNS and get it working (forwarded) on port 80 and I'd tend to avoid them." comment over there.
    I'd ask the vendor for documentation and details (as on its face, a requirement for this proxy seems, well, somewhat questionable), but yes, you're probably in the Apache config files here, and probably using overrides and .htaccess if the site is low-volume, as that'll keep the changes isolated for testing.
    Out of curiousity, is this proxy project part of an attempt to get this package to accessible, but without the vendor actually having the package running on port 80?  (Got a pointer to the package?  We can check the docs, and see if we can translate the installation requirements into something useful on OS X Server.)
    Are you running 10.5 or 10.6?  Both are pretty old releases.

  • Help!! how to use RTP libraries

    Hi you all,
    i have to implement SIP then i were told you have to download RTP libraries i don't know how to download and how to use ,plz anybody can help me :'(

    http://forum.java.sun.com/thread.jspa?threadID=627137&tstart=0
    http://java.sun.com/products/jain/learning/tutorial/index.html
    search forum!!

  • How to use one email adress for multiple recipients

    Hello,
    I'd like to know how to use one email adress for multiple recipients. 
    this would be very useful or projects. for example;
    if i send one mail to [email protected], all people in this project get an email.
    I will add the people in this project myself. 
    I know it is possible, but I don't know how to do it ;-)
    please help me! 

    Hope this help.
    _http://technet.microsoft.com/en-us/library/cc164331(v=exchg.65) .aspx

  • Can't figure out how to use home sharing

    Since the latest couple iTunes updates, my family and I can not figure out how to use home sharing. Everyone in our household has their own iTunes, and for a long time we would just share our music through home sharing. But with the updates, so much has changed that we can no longer figure out how to use it.
    I have a lot of purchased albums on another laptop in the house, that im trying to move it all over to my own iTunes, and I have spent a long time searching the internet, and everything. And I just can't figure out how to do it. So.... how does it work now? I would really like to get these albums from my moms iTunes, onto mine. I would hate to have to buy them all over again.
    If anyone is able to help me out here, that would be great! Thanks!

    The problem im having is that after I am in another library through home sharing, I can't figure out how to select an album and import it to my library. They used to have it set up so that you just highlight all of the songs you want, and then all you had to do was click import. Now I don't even see an import button, or anything else like it. So im lost... I don't know if it's something im doing wrong, or if our home sharing system just isn't working properly.
    Thanks for the help.

  • How to use the same POWL query for multiple users

    Hello,
    I have defined a POWL query which executes properly. But if I map the same POWL query to 2 portal users and the 2 portal users try to access the same page simultaneously then it gives an error message to one of the users that
    "Query 'ABC' is already open in another session."
    where 'ABC' is the query name.
    Can you please tell me how to use the same POWL query for multiple users ?
    A fast reply would be highly appreciated.
    Thanks and Regards,
    Sandhya

    Batch processing usually involves using actions you have recorded.  In Action you can insert Path that can be used during processing documents.  Path have some size so you may want to only process document that have the same size.  Look in the Actions Palette fly-out menu for insert path.  It inserts|records the current document work path into the action being worked on and when the action is played it inserts the path into the document as the current work path..

  • How to use airport time capsule with multiple computers?

    I'm sure there are some thread about this but i couldn't find it... so sorry for that but hear me out! =)
    I bought the AirPort Time Capsule to back up my MBP
    And so i did.
    then i thought "let give this one a fresh start" so i erased all of it with the disk utility and re-installed the MBP from the recovery disk.
    I dont want all of the stuff i backed up just a few files and some pictures so i brought that back.. so far so good.
    Now i want to do a new back up of my MBP so i open time machine settings, pick the drive on the time capsule and then "Choose" i wait for the beck up to begin, and then it fails.  It says (sorry for my bad english, im swedish haha) "the mount /Volume/Data-1/StiflersMBP.sparsebundle is already in use for back up.
    this is what i want:
    i want the "StiflersMBP.sparsebundle" to just be so i can get some stuf when i need them. it's never to be erased.
    i want to make a new back up of my MBP as if it's a second computer...
    so guys and girls, what is the easiest and best solution?
    Best regards!

    TM does not work like that.
    If you want files to use later.. do not use TM.
    Or do not use TM to the same location. Plug a USB drive into the computer and use that as the target for the permanent backup.
    Read some details of how TM works so you understand what it will do.
    http://pondini.org/TM/Works.html
    Use a clone or different software for a permanent backup.
    http://pondini.org/TM/Clones.html
    How to use TC
    http://pondini.org/TM/Time_Capsule.html
    This is helpful.. particularly Q3.
    Why you don't want to use TM.
    Q20 here. http://pondini.org/TM/FAQ.html

Maybe you are looking for