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...
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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! -
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Dear Nokia user
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small_nokia_supporterDear 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".
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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
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AlvaroYou 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.
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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
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I am using MS visual studio for application development. Tell me its compiling steps as well. How to use that in WPF or WinForms.?
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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/
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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.
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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.
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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,
BrettI 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! -
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Hello
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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.
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Hi you all,
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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.
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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.
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