MB usage for SIP-VOIP ?
Does anyone have any experience on :-
1. How much data (GPRS mega-byte) used per minute for Skype over FRING,
2. How much data used for Google-talk
3. Similar apps.
I am using E71, with SIP-VOIP, GRPS, FRING, NIMBUZZ.
Thanks for your help !!
Harry
Hello giliadw,
Don’t wait for a built in function because that’s no where soon to be seen in near future. You should certainly contact a hosted PBX provider they definitely provide VoIP solutions for mobiles. You can simply function all the SIP tools through their application. I can recommend you my PBX VoIP provider The RealPBX, they have a decent phone application and services
Similar Messages
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RV042 protocol binding for SIP and RTP (VoIP)
Hello everybody,
I have a RV042 with a DSL (WAN1) and cable (WAN2) internet connection in Load Balance Mode. The DSL provider also provides internet telephony when registered via his line. When I disable the WAN2 port, my IP phone successully registers with the registration server of the DSL provider. I also defined protocol bindings for SIP (port 5060) and RTP (ports 5004 to 5020) to be bound to WAN1. My IP phone is set up to listen on only these ports.
The rules are in detail:
SIP(UDP/5060~5060) -> "myPhoneIP"~"myPhoneIP" ("RegistrationServerP"~"RegistrationServerIP") WAN1 [Enabled]
SIP(UDP/5060~5060) -> "RegistrationServerIP"~"RegistrationServerIP" ("myPhoneIP"~"myPhoneIP") WAN1 [Enabled]
RTP(UDP/5004~5020) -> "myPhoneIP"~"myPhoneIP" ("RegistrationServerP"~"RegistrationServerIP") WAN1 [Enabled]
RTP(UDP/5004~5020) -> "RegistrationServerIP"~"RegistrationServerIP" ("myPhoneIP"~"myPhoneIP") WAN1 [Enabled]
With these protocol bindings in place when I re-enable WAN2, then after some time the phone reports "registration failed".
Do I need to set something else apart from protocol binding to force the VoIP traffic to go via WAN1?
Thanks for your help
FelixPardon my memory if I am mistaken, when configuring the protocol bind for the WAN port, there are 4 or 5 options. Service, which of course is 1~65535, source IP, in this scenario it should be the phone or PBX, whatever you're using. The destination IP should be 0.0.0.0 and interface is your desired WAN, WAN 1 or 2.
Example:
Wan 1- Cable Wan 2 - Dsl
| |
| ________________ |
|
RV042-----------
____| |
| Computer 192.168.10.100
Tele/PBX 192.168.10.250
On this example to route the Telephone / PBX to WAN 1
All services 1~65535
Source IP 192.168.10.250
Destination IP 0.0.0.0
Interface WAN 1
Please correct me if I am mistaken, I'm currently not at work due to the US holiday -
Looking for a SIP VOIP with some free trial credit
Hi...
Looking for a SIP VOIP with some free trial credit
Can anyone suggest any ?
HarryHi...
Looking for a SIP VOIP with some free trial credit
Can anyone suggest any ?
Harry -
E52 - Nokia SIP VoIP application - help needed
Hi there,
I am pretty new here. I need the Nokia SIP VoIP application for my E52, something like SIP_VoIP_3_1_Settings_S60_3.... .sis
Where can I get it?
My problem is, that I can set up easily an VoIP service with my phone and it connects perfectly. However, I can't use it with my E52 since I can't activitate it on my phone. I would need the option: > connection > Internet Tel. Settings which is not there.
I guess I need the Nokia SIP VoIP application, however I can't find a compatible version, and the SIP_VoIP_3_1_Settings_S60_5_x_v1_0_en.sis does not work.
Thanx
CaristeoAlthough I've never used it, I was able to install the SIP Settings program for my E73 (same OS and Feature Pack as your phone) from http://www.developer.nokia.com/info/sw.nokia.com/id/d476061e-90ca-42e9-b3ea-1a852f3808ec/SIP_VoIP_Se... . (You need to make an account if you don't have one already.) Download the "SIP VOIP 3.x" version. I do have a Settings -> Connection -> SIP Settings option on my phone, as well as Control Panel -> Net Settings -> Advanced VOIP Settings. Hope that helps.
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Error downloading SIP VOIP settings application
Hello!
I have a n85 and I've tried to download the SIP voip app 3.x from
http://www.forum.nokia.com/info/sw.nokia.com/id/d476061e-90ca-42e9-b3ea-1a852f3808ec/SIP_VoIP_Settin...
and it always give me an error.
Is it possible that someone who downloaded it correctly could send it to me?
Thanks!If your office network is behind a firewall that blocks the ports that are necessary for correct operation of SIP then there's nothing you can do to get it working properly. At least not directly.
What you can try is using a 3rd-party application such as fring. Fring allows you to set up a generic SIP session with the VoIP provider of your choice. You connect to fring's systems over a single-port, proprietary connection, and fring connects to your SIP provider instead of you.
It's obviously not as good as a direct connection between you and the VoIP provider but at least it works. I'm more or less forced to do that now with the N96 having no other real SIP support.
Was this post helpful? If so, please click on the white "Kudos!" star below. Thank you! -
VOIP: SIP VoIP 3.1 Nokia Suite begins install, E71...
I have a Nokia E71 and would like to begin using our wifi to make calls, (our carrier signal is week/dropped calls/connection on-hold). When I try to install the file, SIP_VoIP_3_1_Settings_Symbian_3_v1_0_en, Nokia Suite begins the installation, says to continue on phone, phone pauses and then a red screen appears, "File Corrupted".
Can you help?@oneofmarysboys
You are trying to install a version for Symbian S^3 devices, click on the blue upturned arrow head to reveal the other available versions of which you require SIP_VoIP_3x_Settings_v2.0_en.sis (164 kb).
Once installed you may like to look at this resource:http://mywebexperience.com/voip/sip-call-settings-for-e71-voip-calls-through-your-nokia-mobile/
Happy to have helped forum with a Support Ratio = 42.5 -
Nokia e52 WLAN/SIP/VoIP auto connects. Plz help.
Hi,
WLAN/SIP/VoIP on my E52 auto connects evey time it finds a wireless network. I have turned off WLAN monitor but problem is still there.
Any ideas how can i make it manual?
Thanks
David
Solved!
Go to Solution.disable automatic internet connection from those apps which needs internet and by default they have automatic connection option like world traveler, world mate if u have installed it and mail for exchange.
¨Arm yourself because no one else here will save you¨ -
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.
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BlackBerry SIP VoIP Client / fgmicrotec
We are beta testing our BlackBerry SIP VoIP client und would like to invite everyone to join the trial phase. The registration is quick and simple and can be found here:
http://www.fgmicrotec.com/BlackBerry-VoIP.html
Here are a few facts:
- the client comes preconfigured with a test account on our system
- you can reconfigure the client to use any VoIP provider
- use it with your own SIP capable PBX (i.e Asterisk)
- dial from anywhere (call history, contacts, emails, SMS etc.)
- trial users with good bug reports get commercial license for free
- quick and reliable support via our forum http://www.fgmicrotec.com/forum/index.phpI had the similiar problem and finally I found solution:
www.sipVoipSdk.com -
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. -
hi, i am having difficulties making internet calls to mobile/landline/international numbers. I have no difficulties in registering my sip account and making internet calls to other sip account holders. What I notice is that for any internet calls domain name is also appended, for ex [email protected]. I am using nokia E63.
Any suggestions pleaseHi eveybody, an add on to my post.
Regards.
Attachments:
Compare results energy SIP VoIP.pdf 54 KB -
I recently added my brother's phone to my plan, he had Verizon already and we transferred his number to my plan from another person's plan. He's been on my plan since Aug. 2nd, and he's the 4th phone line on my plan now. He isn't being responsible by telling me what his most frequently called numbers are so I may add them to my 10 F&F. I tried to view his top numbers dialed in the usage section, and his number doesn't show up. It only shows me usage for the original 3 lines. I am able to view his calls and details under the billing section, but I have no way of knowing which numbers are most at risk for going over, it won't tell me which of his numbers have VZW or not. He's completely on my plan now, so why wouldn't it show me his heavy hitters? His usage is higher than the other three lines put together, so I need to be able to monitor his usage frquently to ensure no overages. And by the way, I do NOT want him authorized on my account, so hopefully the answer will not be having him call in and access my account to update something... Thanks in advance!
I was getting ready to reply...and saw the issue has resolved itself.
For others who may be reading, and logging in later doesn't work....you can sort the columns when you are viewing the bill online by clicking on the column header. Sort by number, sort by length of call, etc - not ideal but if the Verizon report features are not working, it will give you the information you seek.
if you are good with Excel, you can download the data to a spreadsheet and do your calculations and manipulations there as well. -
How can I monitor my monthly data usage for all 3 computers in my house? I have an Airport base station and it seems there should be software to monitor it from that point rather than monitoring the usage for each computer and then adding it up.
The following example was one of dozens that showed up on a simple Google search of.....
monitor Internet data use on a Mac
Watch your Internet usage with NetUse Monitor | Macworld
Most service providers have an application for their users as well. -
How can I find out my data usage for the past 3 months?
Greetings all, I have been a long time Verizon customer, and as such I, along with my husband, was grandfathered into the unlimited data plan. When I last spoke to a representative, they informed me that the only way we were going to be able to keep our unlimited data was by paying the full retail price ($600+) to get a new phone. So we were considering upgrading and changing to one of the newer data plans so we could still get the discounted price on the new phone, but I would like to make sure I choose one with enough data, without paying for a bunch of data that I don't use, however MyVerizon only shows me the amount I have used since the start of the billing cycle. Is there a way to find out the data usage for both my husband and myself for the last 3 months so I can choose the right option?
Thanks in advance for your help!Hello sweetpea1221! How exciting that you're considering upgrading with us. mrhelper is helpful, indeed. You can view your data usage for each line under the View Bill option via your My Verizon account. You can also view your 3 month average data usage with our Account Analysis feature. Here's a link that will be helpful in doing so http://bit.ly/vnLjqO. I'm confident that we have a great plan to match your data needs. Please let us know if you have further questions.
TanishaS1_VZW
Follow us on Twitter @VZWSupport -
How could i get the kernel and user cpu usage for each process
Hi all,
In order to monitor the system CPU usage, I would like write a script to gather the kernel and user CPU usage for each process, like the prstat or top does. As always missing the shortlived kernel usage, prstat or top cann't get the precise CPU usage. I checked with the dtrace syscall, proc and fbt provider, but don't get which one is useful.
Please provide your comments and suggestion.
Thanks in advmail2sleepy wrote:
As I've studied the "dtrace" for a while, and seems Sun gives a pretty high score on this new feature.....I do want to know whether there's some probe can work for it, like writing a "dtrace" version prstat.You can write a prstat without dtrace. Because that's just polling at specific intervals and reading some process structures from /proc. You could have dtrace fire a probe every 5 seconds and read the same thing, but it wouldn't really be using any features of dtrace. Trhying to write it "in dtrace" doesn't make much sense.
What you could do that would be harder via other methods is to fire a probe at process exit that displayed the process information including total CPU time. They could print exactly when processes exited. Doing that without dtrace would be very difficult.
Darren
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