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  • VoIP Phone Services: Things You must Know

    Moving along with the trend will not give you the desired profits automatically just for the reason that every organization has its own set of structures and requirements. Below Listed are a few basi... [Author: Dennis Smith - Computers and Internet - March 25, 2010]

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  • Use Business VoIP Service To Maximize Your Profits

    The very basic principle for each and Every business concern is to generate or to maximize their profits. There are a several factors which show the accurate results of the company whether it is maki... [Author: Dennis Smith - Computers and Internet - May 17, 2010]

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  • How can I filter packets from a port monitor?

    - by engineerchuan
    I have some data going from Point A to Point B. I have a SPAN monitor set up to a monitoring device C. To recreate some real world scenarios, I want to filter out all traffic which is a certain type (H.323 VoIP Signaling Packets) so that C sees a subset of the information that is flowing from A to B. What would the easiest way to do this be? I assume I would need a computer with 2 NIC cards and some software to examine each packet and chuck out the H.323 VoIP packets? Thanks!

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  • DrayTek 2820 configuration using public IP addresses

    - by Kev
    I have a /29 range of public IP addresses assigned to me by my ISP. I'm trying to configure a SIP VOIP handset to register with my VOIP provider who recommend using public IP addresses rather than NAT. I have a DrayTek 2820 router flashed with the latest firmware and have configured my router as per DrayTek's FAQ at: How do I use a public subnet on the LAN (non-NAT operation ) ? My IP range is: xx.xx.94.16 -> xx.xx.94.23 This gives a usable range of: xx.xx.94.17 -> xx.xx.94.22 My router's public IP address is: xx.xx.94.17, the SIP VOIP handset is allocated xx.xx.94.18. I have a second internet connection and via that I can ping the handset. However for some reason I can't seem to get it to register with the provider. I tried adding a new Firewall filter to pass through from WAN to LAN: Source: ANY, Destination: xx.xx.94.18, UDP - Ports 1024 -> 65535 Out of interest I also tried opening port 80 to see if I could browse to the phone's admin web interface but no joy. I know that my ISP aren't blocking inbound service ports because I NAT Port Forwarded port 80 to one of my internal web servers and it rendered a test page I had set up. All the NAT settings are reset to factory defaults, i.e. there are no Port Redirection, DMZ Host, Open Ports or Address Mappings configured. The handset I'm using is a GrandStream GXP-2000. Is there anything else I should be doing?

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  • Bandwidth Suggestion

    - by Campo
    I have been asked to analyze the bandwidth usage of a company and make a recommendation for upgrading their Internet connection(s). Here is the layout 3 DLS lines so it is 3x(6 Down, 1 Up Each) into a load balancer out to the office's network. 30 VOIP phones run on a T1 (1.5 Down, 1.5 Up) The users at the company are heavily uploading. It is my suspicion that the issue in slowdown is being cause by multiple people uploading and others not being able to get requests out for even simple http requests. My initial idea is to get them a fiber line with a 10 down and 10 up. What do others think on this plan? Will that be enough to host their network traffic? What do I do about the VOIP line afterward? The fiber is expensive and I know the T1 does a great job for their VOIP so I do not want to suggest a DSL line because I know it may not be sufficient. I would also like to save them some money if I can. Maybe even get a faster fiber line and forgo the T1. Though I know their load balance/switch can only handle 20MB/S throughput. Looking for some confirmation/suggestions on my plan. I am planning on going in to get some real diagnostic numbers. Any suggestions on software to use for that? Preferably Windows software.

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  • Why do I need a managed switch and which one should I buy?

    - by ascanio1
    I bought a 2nd router and I want both routers to have direct WAN access to the modem. One of the 2 routers directs VOIP traffic to a telephone line port. This VOIP service is provided by the cable carrier which also leases the modem & the router. The cable company technician told me that this VOIP line uses IPv6 addressing and therefore I must employ an IPv6 capable/compliant Giga Hub/Switch or my telephone line won't work anymore. Pls advise me (brand/model) an IPv6 compliant, 2 port, switch to purchase. Pls educate me: By reading this forum I thought that hubs broadcast traffic to all ports, regardless of which input/output is being used and so, theoretically, they have nothing to do with IP. Correct? Same story for unmanaged switches, where the only difference is that these latter devices route traffic only to those ports which are detected to be in use. Correct? I also understood that unmanaged switches route traffic simply by detecting hardware use and not by selecting specific IP traffic. Correct? Finally, there are managed switches which DO select traffic based on IP and, therefore, only these managed switches are involved with IPv6... Why would my cable company explicitly tell me, over and over, that I must use an IPv6 compliant switch? Why would they need a managed switch instead of an unmanaged one? Thanks in advance for helping me understand!

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  • Is there an alternative to the skype service?

    - by Moonwalker
    On ubuntu 13.04 Skype segfaults constantly (I've read a couple of threads about fixing the issue and it is kind of works now expect it segfaults every time chat message comes in) so I'm thinking it is time to find it a replacement. Which one should I choose? Ok, I've seen previous post, yet only one answer in it highlights some alternatives. Also I want no the alternative skype client, but the whole ecosystem. The one alternative presented ooVoo does not support linux and other goober shows unresolved dependency: libglew1.5

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  • Implementing the transport layer for a SIP UAC

    - by Jonathan Henson
    I have a somewhat simple, but specific, question about implementing the transport layer for a SIP UAC. Do I expect the response to a request on the same socket that I sent the request on, or do I let the UDP or TCP listener pick up the response and then route it to the correct transaction from there? The RFC does not seem to say anything on the matter. It seems that especially using UDP, which is connection-less, that I should just let the listeners pick up the response, but that seems sort of counter intuitive. Particularly, I have seen plenty of UAC implementations which do not depend on having a Listener in the transport layer. Also, most implementations I have looked at do not have the UAS receiving loop responding on the socket at all. This would tend to indicate that the client should not be expecting a reply on the socket that it sent the request on. For clarification: Suppose my transport layer consists of the following elements: TCPClient (Sends Requests for a UAC via TCP) UDPClient (Sends Requests for a UAC vid UDP) TCPSever (Loop receiving Requests and dispatching to transaction layer via TCP) UDPServer (Loop receiving Requests and dispatching to transaction layer via UDP) Obviously, the *Client sends my Requests. The question is, what receives the Response? The *Client waiting on a recv or recvfrom call on the socket it used to send the request, or the *Server? Conversely, the *Server receives my requests, What sends the Response? The *Client? doesn't this break the roles of each member a bit?

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  • How can I tell if ZRTP is enabled in a Twinkle SIP call?

    - by komputes
    I recently attended a talk about GNU Telephony. I was informed that Twinkle supports ZRTP for encrypted SIP calls. I went into Edit User Profile Security and made sure that ZRTP was enables and that all boxes were checked. I asked a friend to do the same and then we called each other. There is no immediate indication that I can see that the call is secure. How can I tell if ZRTP is enabled in a Twinkle SIP call?

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  • Skype locking up, and microphone "lagging"

    - by Svendbenno
    Hi. I've always had this problems with skype and pulseaudio on ubuntu. Whenever i start up skype, i have to call someone, then hang up 4-5 times, before the other person can hear my voice. When i, or the other person hang up, skype tends to lock up. I can't kill it with "killall skype" or a logout, so i have to restart my computer. Have anyone else encountered this problem, and if so solved this? I'm using 10.10 btw.

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  • asterisk/freeswitch in nat/no-nat setup

    - by pQd
    hi, my current setup - i use bunch of sip hard-phones around few offices. all devices have two sip accounts configured - one on internal sip proxy [for calls between the branches], another - at 3rd party voip providers [ since it's in different countries - those are different providers, but that's irrelevant ]. i was thinking about terminating sip calls on something like asterisk/freeswitch server and having all sip-devices log on just once to such server[s] - mostly to provide things like voicemail, groupcalls, redirections etc. it seems perfectly doable but there is one problem - i cannot find examples how to prepare for nat/no nat. for calls routed to from/to 3rd party voip operator - i'll need handling for nat/stun etc, but for handling of internal calls - i do not want any nat, all traffic should go via vpns to different branches. can you provide me some hints how to configure it? any tutorials? thanks!

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  • maemo - n900 - SIP call quality

    - by Walter White
    Hi all, I have been using SIP / VoIP on my n900 to make calls and my problem is after about 15 minutes of talk time, more recently 18 minutes exactly, my connection dies and I can no longer hear them or them me. I have tested this with various VoIP providers to confirm that it is not specific to any one provider, but instead my phone. I also have tested this on my laptop. I sent my phone to be tested at some place that tests hardware and no problems were found with the hardware. What can I do to rectify the 15 minute call barrier with SIP on my phone? The other problem I have too is that for the wireless broadband to start working again, I need to restart the phone, it appears the network driver gets overloaded. The one thing that appears to work fine is making cellular calls. I have yet to have call quality drop off after 15 minutes over a cellular connection. Thanks, Walter

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  • add presence for a remote user to a legacy telephone system?

    - by niko
    we have a small call center that uses an old nortel phone system with analog lines. one of our sales people works from home so her calls do not go through the phone system. this creates a problem at time as the receptionist does not know if she is on the phone or not. we can easily get around this by using instant messenger status but i wanted to ask if there is another way that we can do it so that calls can also be forwarded to her when she is not on the phone. i realize that we can do this with a voip system but we're not planning on upgrading to voip until next year. does anyone know if there is an inexpensive way to add this capability today?

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  • How do I setup a sip account with pennytel in empathy or ekiga in ubuntu 10.04?

    - by Glen
    Hi, I am trying to get my PennyTel VOIP account working in ubuntu 10.04. I know ubuntu use to use ekiga for sip calls. I also know that ubuntu now uses empathy for sip calls. However I can't get either program to work with my VOIP provider. In empathy I have no idea what details to enter. How does it know I want to use pennytel? What is my user name? I tired [email protected] but it did not work. In Ekiga I can't figure out how to use a sip provider. It looks like I can only use the Ekiga sip provider, which I don't want to use because I already have a PennyTel account. Any help would be appreciated. Thanks, Glen.

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