Choppy voip connection

I have 1MB wireless link from my ISP. I am using Cisco VOIP phones in my company. Over the months my VOIP quality is not good(on/off connectivity),though my internet is somehow working fine. When I talked to my ISP , they told me to use Cisco switch / router instead of Nortal. Moreover ISP also saying that the switch / router can be configured to full duplex/ half duplex because ISP equipment can be switched to half/duplex.
-->My questions.
1-what cisco router/switch should I use with voip phones ?
2-What duplex setting should be on the router/switch so voip quality be remains optimum.
3-Is 1MB link can be used for voip?
4-Is it possible to configure voip phones to full/half duplex ?
Please help. Thanks.

Hi,
2. If you're currently running your phones over a wireless connection, this can cause problems. Wireless doesn't have guaranteed bandwidth and it's half duplex.
If you were to switch this out with a router and switch, you could make every link full duplex.
Full duplex is what you should run everywhere. The only modern technology you should be running half-duplex on is wireless, because that's just the way it works.
1. The router depends on your company size and what you want to use it for.
3. 1 MB link may be alright, but again, it depends on how many active phone calls, what your data usage is, and what codec you're running.
4. Phones are auto-negotiating, and they will use either.

Similar Messages

  • Nokia E52 choppy VoIP due to WLAN interruptions an...

    I love my Nokia E52, but one of the main reasons I bought it (the combination of WiFi Wireless Lan support and fully integrated Voice over IP client) is not usable due to very laggy, unsteady WLAN connection (while having perfect signal strength - E52 shows 100% (same room)).
    Because of this I get choppy VoIP (cracks and interruptions up to several words in my VoIP conversations).
    My E52:
    Software version: 031.012
    Modell: E52-1
    Type: RM-469
    This is not a problem of the VoIP Client, audio-codec, or SIP provider/internet connection.
    It is the Wireless Lan connection between my E52 and my Router. I tried 3 different access points, with and without wireless encryption, 802.11b (1M...11M data-rate) and 802.11g - problem always the same.
    I can track it down by pinging my E52 (idle without any programs running) from my Linux box:
    192.168.0.100 is the IP of my E52 assigned by DHCP.
    1.) 64byte pings, once per second:
    PING 192.168.0.100 (192.168.0.100) 56(84) bytes of data.
    64 bytes from 192.168.0.100: icmp_seq=1 ttl=69 time=6.36 ms
    64 bytes from 192.168.0.100: icmp_seq=2 ttl=69 time=57.9 ms
    64 bytes from 192.168.0.100: icmp_seq=3 ttl=69 time=80.9 ms
    64 bytes from 192.168.0.100: icmp_seq=4 ttl=69 time=203 ms
    64 bytes from 192.168.0.100: icmp_seq=5 ttl=69 time=128 ms
    64 bytes from 192.168.0.100: icmp_seq=6 ttl=69 time=50.0 ms
    64 bytes from 192.168.0.100: icmp_seq=8 ttl=69 time=324 ms
    64 bytes from 192.168.0.100: icmp_seq=9 ttl=69 time=79.7 ms
    64 bytes from 192.168.0.100: icmp_seq=10 ttl=69 time=37.5 ms
    64 bytes from 192.168.0.100: icmp_seq=11 ttl=69 time=60.0 ms
    64byte pings using a good wireless 802.11g connection should be around 1ms and constant!!
    It's not because of bandwith limit of E52 wlan/cpu, because the higher ping times stay the same, even if I increase ping-packet-size 150x:
    2.)10000byte pings (150x times bigger packages!), once per second [still ping times look quite similar]:
    PING 192.168.0.100 (192.168.0.100) 10000(10028) bytes of data.
    10008 bytes from 192.168.0.100: icmp_seq=1 ttl=69 time=175 ms
    10008 bytes from 192.168.0.100: icmp_seq=2 ttl=69 time=89.8 ms
    10008 bytes from 192.168.0.100: icmp_seq=3 ttl=69 time=115 ms
    10008 bytes from 192.168.0.100: icmp_seq=4 ttl=69 time=234 ms
    10008 bytes from 192.168.0.100: icmp_seq=5 ttl=69 time=153 ms
    10008 bytes from 192.168.0.100: icmp_seq=6 ttl=69 time=83.7 ms
    3.) Small 64byte pings, but now ten pings per second:
    Now it gets even more strange:
    # ping 192.168.0.100 -i 0.1
    PING 192.168.0.100 (192.168.0.100) 56(84) bytes of data.
    64 bytes from 192.168.0.100: icmp_seq=1 ttl=69 time=200 ms
    64 bytes from 192.168.0.100: icmp_seq=2 ttl=69 time=95.0 ms
    64 bytes from 192.168.0.100: icmp_seq=3 ttl=69 time=2.25 ms
    64 bytes from 192.168.0.100: icmp_seq=4 ttl=69 time=5.93 ms
    64 bytes from 192.168.0.100: icmp_seq=5 ttl=69 time=4.34 ms
    64 bytes from 192.168.0.100: icmp_seq=6 ttl=69 time=8.45 ms
    64 bytes from 192.168.0.100: icmp_seq=7 ttl=69 time=5.91 ms
    64 bytes from 192.168.0.100: icmp_seq=8 ttl=69 time=2.69 ms
    64 bytes from 192.168.0.100: icmp_seq=9 ttl=69 time=2.31 ms
    64 bytes from 192.168.0.100: icmp_seq=10 ttl=69 time=5.90 ms
    64 bytes from 192.168.0.100: icmp_seq=72 ttl=69 time=8.42 ms
    64 bytes from 192.168.0.100: icmp_seq=73 ttl=69 time=7.25 ms
    64 bytes from 192.168.0.100: icmp_seq=74 ttl=69 time=7.28 ms
    64 bytes from 192.168.0.100: icmp_seq=75 ttl=69 time=7.25 ms
    64 bytes from 192.168.0.100: icmp_seq=76 ttl=69 time=7.24 ms
    64 bytes from 192.168.0.100: icmp_seq=77 ttl=69 time=7.92 ms
    64 bytes from 192.168.0.100: icmp_seq=78 ttl=69 time=7.24 ms
    64 bytes from 192.168.0.100: icmp_seq=79 ttl=69 time=8.69 ms
    64 bytes from 192.168.0.100: icmp_seq=80 ttl=69 time=10.9 ms
    64 bytes from 192.168.0.100: icmp_seq=81 ttl=69 time=10.6 ms
    64 bytes from 192.168.0.100: icmp_seq=150 ttl=69 time=213 ms
    64 bytes from 192.168.0.100: icmp_seq=151 ttl=69 time=105 ms
    64 bytes from 192.168.0.100: icmp_seq=152 ttl=69 time=3.09 ms
    64 bytes from 192.168.0.100: icmp_seq=153 ttl=69 time=2.23 ms
    64 bytes from 192.168.0.100: icmp_seq=154 ttl=69 time=2.14 ms
    64 bytes from 192.168.0.100: icmp_seq=155 ttl=69 time=3.78 ms
    64 bytes from 192.168.0.100: icmp_seq=156 ttl=69 time=3.02 ms
    64 bytes from 192.168.0.100: icmp_seq=157 ttl=69 time=5.48 ms
    64 bytes from 192.168.0.100: icmp_seq=158 ttl=69 time=2.15 ms
    64 bytes from 192.168.0.100: icmp_seq=159 ttl=69 time=2.13 ms
    ^C
    --- 192.168.0.100 ping statistics ---
    159 packets transmitted, 30 received, 81% packet loss, time 16866ms
    rtt min/avg/max/mdev = 2.130/25.327/213.218/54.102 ms, pipe 2
    Note: My Linux Ping Command doesn't say something like "Request timed out." for lost packets (like Windows Ping does). So if there is a jump from icmp_seq=81 to icmp_seq=150 it is because everything in between got lost.
    You see, in 3.) it has the slow pings once in a while and in between almost normally fast pings, but suddenly a long lag of 6-7sec (this is deadly for voip) and then the circle repeats.
    Is my E52 wlan chip defect, or is this some kind of Power-Management system? How can I turn it off?
    I don't need a fast wlan connection on my E52. 1MBit/s is more than enough. But it has to be steady, without lags.
    Ideas, anybody? Please!

    Well, it has nothing to do with the VoIP service itself for sure. As razvan_t has already noted earlier this problem affect all network services like internet radio.
    It behaves exactly as E52affair described in the very first post. If I ping the phone through my home LAN using ping with normal settings, i.e. 1 packet per second nothing much wrong happens. It finishes with some 1% packet loss. But when I increase the ping frequency up to 10 packets/s the same test ends with terrible >80% packet loss as you can see below. And the following pattern can be observed during this test: 10 packets goes through well then some 60-70 packets is lost and it repeats again and again. Lags in VoIP audio stream correspond to the lags in ping. The wifi conditions are excellent, phone is located some 2 meters far from the router´s wifi antenna.
    I tried watch some video on Youtube. The video stream was prefetched in jumps appearing with the same frequency as lags in fast ping were observed during the test.
    Before I found this thread I obviously thought it was a VoIP problem. Therefore I tried various VoIP settings, different codecs, nothing has helped. Now it is clear why.
    I even did a phone hard reset. It did not help neither.
    Because I observe almost the same lags in VoIP even when the phone is connected using the packet data through the GSM network, I suppose the problem is not limited to wifi connection only. But I cannot confirm this theory because I´m not able to ping the phone throughout the GSM data network.
    jirka@pc-jirka:~$ sudo ping 192.168.100.14 -i 0.1
    PING 192.168.100.14 (192.168.100.14) 56(84) bytes of data.
    64 bytes from 192.168.100.14: icmp_req=1 ttl=69 time=64.6 ms
    64 bytes from 192.168.100.14: icmp_req=2 ttl=69 time=7.58 ms
    64 bytes from 192.168.100.14: icmp_req=3 ttl=69 time=7.61 ms
    64 bytes from 192.168.100.14: icmp_req=4 ttl=69 time=7.62 ms
    64 bytes from 192.168.100.14: icmp_req=5 ttl=69 time=4.36 ms
    64 bytes from 192.168.100.14: icmp_req=6 ttl=69 time=7.64 ms
    64 bytes from 192.168.100.14: icmp_req=7 ttl=69 time=7.93 ms
    64 bytes from 192.168.100.14: icmp_req=8 ttl=69 time=7.57 ms
    64 bytes from 192.168.100.14: icmp_req=9 ttl=69 time=7.58 ms
    64 bytes from 192.168.100.14: icmp_req=10 ttl=69 time=7.60 ms
    64 bytes from 192.168.100.14: icmp_req=69 ttl=69 time=568 ms
    64 bytes from 192.168.100.14: icmp_req=70 ttl=69 time=461 ms
    64 bytes from 192.168.100.14: icmp_req=71 ttl=69 time=354 ms
    64 bytes from 192.168.100.14: icmp_req=72 ttl=69 time=248 ms
    64 bytes from 192.168.100.14: icmp_req=73 ttl=69 time=140 ms
    64 bytes from 192.168.100.14: icmp_req=74 ttl=69 time=33.0 ms
    64 bytes from 192.168.100.14: icmp_req=75 ttl=69 time=7.58 ms
    64 bytes from 192.168.100.14: icmp_req=76 ttl=69 time=4.48 ms
    64 bytes from 192.168.100.14: icmp_req=77 ttl=69 time=7.68 ms
    64 bytes from 192.168.100.14: icmp_req=78 ttl=69 time=7.65 ms
    64 bytes from 192.168.100.14: icmp_req=147 ttl=69 time=57.9 ms
    64 bytes from 192.168.100.14: icmp_req=148 ttl=69 time=5.26 ms
    64 bytes from 192.168.100.14: icmp_req=149 ttl=69 time=7.68 ms
    64 bytes from 192.168.100.14: icmp_req=150 ttl=69 time=7.60 ms
    64 bytes from 192.168.100.14: icmp_req=151 ttl=69 time=7.67 ms
    64 bytes from 192.168.100.14: icmp_req=152 ttl=69 time=8.07 ms
    64 bytes from 192.168.100.14: icmp_req=153 ttl=69 time=7.67 ms
    64 bytes from 192.168.100.14: icmp_req=154 ttl=69 time=4.24 ms
    64 bytes from 192.168.100.14: icmp_req=155 ttl=69 time=7.62 ms
    64 bytes from 192.168.100.14: icmp_req=156 ttl=69 time=7.53 ms
    64 bytes from 192.168.100.14: icmp_req=225 ttl=69 time=99.5 ms
    64 bytes from 192.168.100.14: icmp_req=226 ttl=69 time=3.79 ms
    64 bytes from 192.168.100.14: icmp_req=227 ttl=69 time=20.3 ms
    64 bytes from 192.168.100.14: icmp_req=228 ttl=69 time=3.91 ms
    64 bytes from 192.168.100.14: icmp_req=229 ttl=69 time=4.57 ms
    64 bytes from 192.168.100.14: icmp_req=230 ttl=69 time=4.70 ms
    64 bytes from 192.168.100.14: icmp_req=231 ttl=69 time=7.95 ms
    64 bytes from 192.168.100.14: icmp_req=232 ttl=69 time=5.35 ms
    64 bytes from 192.168.100.14: icmp_req=233 ttl=69 time=5.65 ms
    64 bytes from 192.168.100.14: icmp_req=234 ttl=69 time=5.63 ms
    ^C
    --- 192.168.100.14 ping statistics ---
    294 packets transmitted, 40 received, 86% packet loss, time 31394ms
    rtt min/avg/max/mdev = 3.795/56.194/568.719/126.270 ms, pipe 6

  • Broken VoIP connection E61 to asterisk server afte...

    After upgrading and restoring settings from backed up data the mobile phone finds it's WLAN connection and shows an established VoIP connection (as before).
    But no incoming or outgoing calls are possible over VoIP. At server side nothing was changed.
    Asterisk logs the following at session initiation:
    -- Registered SIP 'XXX' at 192.168.1.184 port 5060 expires 3600
    -- Got SIP response 400 "Bad Request" back from 192.168.1.184
    When trying to call the phone:
    -- Got SIP response 488 "Not Acceptable Here" back from 192.168.1.184
    -- SIP/XXX-69d3 is circuit-busy
    == Everyone is busy/congested at this time (1:0/1/0)
    Anbody any ideas?
    Thanks in advance.
    Nazdravi

    as far as i know, you'd need xsan nodes for all machines involved and san storage shared among all nodes. ideally, you'd need primary and secondary xsan metadata controllers just to deal with san metadata, and they should not be your od master or replica.
    you could probably skimp on hardware and double up services on each machine, but it probably wouldn't save you much in the short term, at least.
    how about moving to more power efficient processors for your current setup? your network as it is now sounds exactly like what some of my clients would love to have, but it would take a lot of cash to build with apple hardware/software.

  • VOIP connectivity with BIAMP Tesira DSP

    Hello All,
    We are looking at hooking a BIAMP Tesira DSP into our Lync 2013 server to handle VOIP calls in our conference room.  The unit has an add in module specific for handling VOIP connectivity (SVC-2) but I haven't been able to find any documentation on how
    to go about configuring it for a Lync server. 
    Has anyone ever worked with one of these or have any ideas on where to start?
    Thanks!

    Hi,
    I agree with Holger Bunkradt. You need a Microsoft Certified Media Gateway or a SBC between Lync Mediation Server and the BIAMP Tesira DSP.
    You can choose from the link below:
    https://technet.microsoft.com/en-us/office/dn788945
    Best Regards,
    Eason Huang  
    Please remember to mark the replies as answers if they help, and unmark the answers if they provide no help. If you have feedback for TechNet Support, contact [email protected]
    Eason Huang
    TechNet Community Support

  • Caller ID problem between 2 PBX's across Cisco VoiP connection

    Hi all,
    I have two Cisco 2811 Routers connecting 2 Avaya Index PBX's via T1/E1 cards (configured for E1)across a LES 10 circuit. Everything works fine but the problem i have is when phone1 (John) at site A dials phone2 (James) a site B, the incoming number is shown but not the name of the person. When 2 calls are made within a site, the name of the calling party is shown (not the directory number). Can someone please shed some light on if this is possible across the VoiP connection and how i would configure it. I have attached the configs to help.
    Thanks in advance,
    Gary.

    Ideally there should not be any line code violations on the data T1 at all.
    Violations are at the data T1 and the voice T. And they shud be avoided.

  • QSIG VoIP Connection CCBS feature

    We are connecting three PBXs wit VoIP.
    Using 1760-V with 12.3(4)T11 images, VWIC-1MFT-E1 interfaces and isdn Q.SIG.
    Our PBXs vendor and our customer wants to use CCBS
    (call completion to busy subsriber).
    But there we could not find any documentation, if this feature is supported or not.
    And which IOS version to use.
    Many thanks in advance.
    August

    Hi there,
    QSIG with supplementary services is supported only with MGCP and with PRI backhaul in conjunction with ccm (this means the voice gateway passes the information to the end pbx).
    http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803edb19.html#wp1139777
    an example between avaya, ccm and how to configure 3745:
    http://www.cisco.com/en/US/partner/tech/tk652/tk701/technologies_tech_note09186a00803d22cd.shtml

  • Multipoint VOIP Connection

    Good Afternoon,
    I want to connect through VOIP 15 different points.
    Each point has 1 POTS Line and 1adsl Line. Each Point has Static IP address.
    I want to use VOIP only for calls between those 15 points.
    I am thinking to use SPA3102 to each point with analog devices.
    All external calls will be redirected to the POTS line and all calls between the 15 points will be redirected through VOIP to the other point.
    Is this possible to be implemented?
    Do I need any extra Hardware to implement this? Like a SPA900 somewhere?
    Is SPA3102 the appropriate device to do this?
    The router each point is using is a CISCO B107. Are there any issues continue using this hardware?
    Thank you very much in advance for your interest
    dvachtse

    I think the only way to do this and keep it manageable is to setup a PBX at one site. You could probably pull it off with an SPA9000, but you are right at the maximum number of connections and routing would be a pain. If it were me I would setup a trixbox at one location.

  • VoIP connection to PABX

    Essentially, I'm trying to extend a analog line from a Nortel PABX across a data network to a slightly out of reach phone.
    I have 2 routers (2801) equipped with FXO and FXS cards each. I want to extend a analog line (RJ11) from the PABX to the router with the FXO card and from that router to another router with a FXS card, and from there, to a POTS phone. The PABX would provide an extension to the phone, and would route calls to that extension to the connected port on the router. On the other hand, the router with the FXS card would capture whatever number the phone dials and just pass it straight alot to the PABX.
    Is this possible? I've been reading reference material all week trying to figure it out.

    Hi,
    for the HQ router with 4 FXO's i have:
    voice-port 0/3/0
    input gain 2
    output attenuation -4
    timing hookflash-out 500
    connection plar opx 200
    impedance complex1
    voice-port 0/3/1
    input gain 2
    output attenuation -4
    timing hookflash-out 500
    connection plar opx 201
    impedance complex1
    voice-port 0/3/2
    input gain 2
    output attenuation -4
    timing hookflash-out 500
    connection plar opx 202
    impedance complex1
    voice-port 0/3/3
    input gain 2
    output attenuation -4
    cptone ZA
    timing hookflash-out 500
    connection plar opx 203
    impedance complex2
    dial-peer voice 5572 pots
    destination-pattern 5572
    port 0/3/0
    dial-peer voice 5852 pots
    destination-pattern 5852
    port 0/3/1
    dial-peer voice 5815 pots
    destination-pattern 5815
    port 0/3/2
    dial-peer voice 5853 pots
    destination-pattern 5853
    port 0/3/3
    dial-peer voice 200 voip
    destination-pattern 20.
    session target ipv4:10.61.156.1
    incoming called-number .
    dtmf-relay h245-signal
    no vad
    and for the remote site with 4 FXS i have:
    voice-port 0/2/0
    input gain 2
    output attenuation -4
    cptone ZA
    connection plar 5572
    voice-port 0/2/1
    input gain 2
    output attenuation -4
    cptone ZA
    connection plar 5852
    voice-port 0/3/0
    input gain 2
    output attenuation -4
    cptone ZA
    connection plar 5815
    voice-port 0/3/1
    input gain 2
    output attenuation -4
    cptone ZA
    connection plar 5853
    dial-peer voice 200 pots
    destination-pattern 200
    port 0/2/0
    dial-peer voice 201 pots
    destination-pattern 201
    port 0/2/1
    dial-peer voice 202 pots
    destination-pattern 202
    port 0/3/0
    dial-peer voice 203 pots
    destination-pattern 203
    port 0/3/1
    dial-peer voice 100 voip
    max-conn 4
    destination-pattern 5...
    session target ipv4:10.59.1.70
    incoming called-number .
    dtmf-relay h245-signal
    no vad
    Hope this helps

  • VoIP connection to company server

    Hi there,
    When reading the specs of the IPhone 3G I was interrested in its capability to connect to VoIP servers. I noticed there are lots of apps around which let it connect to skype like services. In other words: services for which you need an account and you place calls via the Internet.
    This is not what I am looking for.
    What I need is a replacement for the Nokia E series.
    With the Nokia E series I can make a VoIP profile connecting to our company VoIP server. The profile becomes an intergral part in the phones methodes of placing and receiving calls.
    This way I can place and receive calls through our company VoIP server when in range of our Wifi network. When out of range it switches to the network of the telecom provider (like Vodafone).
    Is this posible with Apple IPhone?
    Peter

    Seems it is called ISip nowadays.... looks good.
    Now all I need is an IPhone to actually test it .... thanks

  • "Choppy" audio connection

    Howdy folks,
    I've read the FAQs and the various other threads in this forum and I haven't been able to solve this problem of mine on my own. I apologise if I have missed something!
    My girlfriend and I are trying to use an audio connection in iChat. Connecting to one another is not a problem, and the audio is typically crystal-clear and uninterrupted *at first*. Soon, however, it will cut out on my side (her side is more or less fine). Sometimes very briefly, sometimes for a second or two. Often it will happen once and be fine for ten seconds or so (never much longer than that, though); other times it will happen in rapid succession. The net effect is that it is difficult to carry a conversation with these interruptions. I'm hoping, however, that there is something I can do. Any ideas?
    We are both running the version of iChat that came with Tiger. We also both have PowerBook G4s. My inet connection is to a university network via ethernet. My download and upload rates are, respectively: approximately 200 KB/sec and app. 45 KB/sec. My girlfriends inet connection is wireless with a download rate of app. 320 KB/sec and upload rate of app. 125 KB/sec.
    Thanks for your help!
    Charles Dobson

    Hello, Charles
    My hightest recommendation: You should upgrade to 10.4.2. It corrects several issues (including iChat AV issues) that were problems in the previous versions of Tiger.
    Try this and any other relevant suggestions from Help for iChat AV 3 Problems.
    The testing and tutorial suggestions in Using iSight with iChat AV might also give you some trouble shooting ideas if you are still having difficulty after implementing the above suggestions.
    Also, being at a university, you might try at different times of day to see whether busy network traffic is causing problems for you at certain times.
    If necessary, you can even take your PowerBook to an off-campus network to help determine whether the problem is in your Mac no matter which network you use to connect.

  • Nokia E51 VOIP connection

    I have E51 mobile phone and appreciate if anyone in the forum knows how to configure SIP settings for internet telephone using
    VOIP. Can this be done with EQO or Sype services?

    what do u mean by recording?
    T18>6210>8310>6800>6210>7610>6680>6210>E50>3110>2610>E65>E61i

  • Choppy airport connection after OS X reinstall

    I recently put a new hard drive in my 12" 1.33 GHZ iBook G4, which of course required reinstallation of everything. After I got everything back up and running, I've found that my airport connection, while showing full signal strength, gets little to no connectivity(pages take minutes to load, if not failing first). I'm using an 802.11b linksys router. The internet works fine when I plug an ethernet cable in to my iBook. Furthermore, a windows laptop right next to my computer works just fine with the wireless signal. Are there any setting that I might have forgot to change back after reinstalling OS X, or any updates/programs that may cause this kind of thing? Thanks alot in advance.
    iBook G4   Mac OS X (10.4.3)  

    Tried it. No luck again. I reformatted the drive and did a new installation of the OS. I was welcomed to Tiger and asked if I wanted to transfer info from another Mac. I said no, and I was then able to enter my basic info (name, address, etc.) myself. But halfway through the setup of the OS (right when I selected my wireless network) the computer screen went blue and that was it. No further activity for the next hour or so until I shut it off and gave up.
    I am wondering if the problem might have something to do with a faulty Airport card or something. The computer goes to access the wireless and hangs.
    Maybe.
    Macbook Pro 2.16 15-inch   Other OS   Not booting.
    AlBook 1.67GHz 15-inch   Mac OS X (10.4.10)   1GB RAM, 80GB HD, Superdrive
    Macbook Pro 2.16 15-inch   Mac OS X (10.4.5)   Not booting or allowing OS X to be fully installed/set up.

  • U-Verse VOIP into house wiring: "Line-in-Use"

    All,       ATT converted our DSL to U-Verse high-speed Internet + VOIP 5 months ago. This included VOIP connection to original phone-wiring in the house (built in 1989), supporting 3 extensions.  All worked fine until last week, when all extensions show "Line-in-Use" and are inoperable. VOIP constantly trying to "Connect" == flashing green light on Modem/VOIP.  Disconnecting all phones has no effect.  Disconnecting line from Modem/VOIP to house wiring immediately enables VOIP connection.  Plugging phone directly into Modem/VOIP provides excellent phone service.       Obvious conclusion is that there is now a problem with house wiring.  Odd since the wires haven't been touched in months, and we have had no rain.  Any suggestions on where to start looking at the house wiring?Appreciate any advice.-Joe in San Diego

    A couple of examples of house wiring problems out of the blue -
    The wireis behind kitchen cabinets. Every time the spoon drawer is closed it beats up the wire til it breaks or shorts.
    The wire gets crimped in the jack mounting plate & eventually shorts out.
    Obviously, it is difficult to troubleshoot hidde wires. First determine if you are daisy chained from jack to jack or star connected to a bunching block.
    The daisy chain - work out from the RG to open the next segment til you bracket the segment.
    The star - check the feed leg. Remove all legs & add them back one at atime til it fails.
    Note that the tel connection is on thetwo center prongs of the RJ-11 jack. IT is standard to connect these to the red/green wires. With one phone the yellow/black are not used. If you find a problem with the red-green that you cannot fix because you cannot get to it, you could use the yellow -black for that segment.
    When you had DSL the tel house wiring was fed from outside. When you switched to u-verse feed on the RG the connecton to the outside should have been opened up. If it was not it may  still have worked. But, now the short is outside. Make sure you are not still connected to the outside.

  • VoIP Phones - Testing Latency, Jitter, and Packet Loss

    I am having big problems with my VoIP phone connection and I'll try to lay it out clearly here.
    The main telephone system resides at Location A (static IP address - see below - xxx.xxx.206.19), which has a network connection of 50MB down/20MB up (i.e., very fast).  The VoIP phone configured for that system resides at Location B, which has a network connection of 10MB down/1MB up (i.e., also fast, or at least fast enough "on paper" for a quality VoIP connection).  The LAN at Location A uses an Airport Extreme router, which does not have QOS or EF capability. The LAN at Location B uses a D-Link DIR-655 router which does have QOS that is configured properly to direct all traffic to the VoIP phone's IP address.
    The VoIP phone at Location B is having intermittent call quality problems with skipping of words, hollowing out noises, jittery conversations, etc.  All the inquiries I've made to the ISPs and phone system manufacturer (ESI) suggest that my base Internet speeds are not the problem.
    I'm told, instead, that the problem might be latency, jitter, or packet loss between Location A and Location B.  This leads to several questions:
    (1)     Is there any Mac software that can test latency, jitter, and packet loss? I've looked at Network Utility and it seems to only measure a few things. 
    (2)     Does anyone see anything in the following Traceroute and Ping results (done twice from Location B to Location A) that looks problematic to VoIP quality?:
    Traceroute:
    First run: Traceroute has started…
    traceroute to xxx.xxx.206.19 (xxx.xxx.206.19), 64 hops max, 72 byte packets
    1  alfirving (192.168.0.1)  0.569 ms  0.363 ms  0.302 ms
    2  10.72.28.1 (10.72.28.1)  27.567 ms 18.161 ms  22.288 ms
    3  70.125.216.150 (70.125.216.150)  9.841 ms  10.346 ms  9.497 ms
    4  24.164.209.116 (24.164.209.116)  11.042 ms 8.298 ms  9.433 ms
    5  70.125.216.108 (70.125.216.108)  21.068 ms  20.657 ms  12.045 ms
    6  te0-8-0-2.dllatxl3-cr01.texas.rr.com (72.179.205.48)  11.154 ms  11.540 ms  24.495 ms
    7  107.14.17.136 (107.14.17.136)  11.994 ms  14.217 ms  15.816 ms
    8  ae-3-0.pr0.dfw10.tbone.rr.com (66.109.6.209) 14.566 ms  32.670 ms  15.947 ms
    9  ix-0-3-2-0.tcore2.dt8-dallas.as6453.net (209.58.47.105)  11.647 ms  12.260 ms  12.386 ms
    10  if-2-2.tcore1.dt8-dallas.as6453.net (66.110.56.5) 10.023 ms  12.285 ms  12.338 ms
    11  209.58.47.74 (209.58.47.74)  17.641 ms 16.741 ms  16.372 ms
    12  0.ae2.xl3.dfw7.alter.net (152.63.97.57)  11.584 ms  12.315 ms  12.890 ms
    13  0.so-6-1-0.dfw01-bb-rtr1.verizon-gni.net (152.63.1.90)  13.812 ms
        0.ge-3-0-0.dfw01-bb-rtr1.verizon-gni.net (152.63.1.17)  18.831 ms
        130.81.23.164 (130.81.23.164)  14.189 ms
    14  p14-0-0.dllstx-lcr-05.verizon-gni.net (130.81.27.40) 14.561 ms  13.621 ms  15.544 ms
    15  * * *
    16  static-xxx.xxx.206.19.dllstx.fios.verizon.net (xxx.xxx.206.19)  23.125 ms  24.136 ms  22.411 ms
    Second run: Traceroute has started…
    traceroute to xxx.xxx.206.19 (xxx.xxx.206.19), 64 hops max, 72 byte packets
    1  alfirving (192.168.0.1)  0.603 ms  0.420 ms  0.324 ms
    2  10.72.28.1 (10.72.28.1)  40.494 ms 26.625 ms  14.152 ms
    3  70.125.216.150 (70.125.216.150)  9.431 ms  9.660 ms  9.018 ms
    4  24.164.209.116 (24.164.209.116)  16.293 ms  12.339 ms  19.252 ms
    5  70.125.216.108 (70.125.216.108)  15.801 ms  11.438 ms  12.068 ms
    6  te0-8-0-2.dllatxl3-cr01.texas.rr.com (72.179.205.48)  23.221 ms  30.459 ms  17.519 ms
    7  107.14.17.136 (107.14.17.136)  14.611 ms  15.696 ms  15.775 ms
    8  ae-3-0.pr0.dfw10.tbone.rr.com (66.109.6.209) 17.643 ms  14.812 ms  16.294 ms
    9  ix-0-3-2-0.tcore2.dt8-dallas.as6453.net (209.58.47.105)  11.169 ms  12.374 ms  9.849 ms
    10  if-2-2.tcore1.dt8-dallas.as6453.net (66.110.56.5) 16.453 ms  12.168 ms  12.384 ms
    11  209.58.47.74 (209.58.47.74)  18.015 ms 14.867 ms  16.432 ms
    12  0.ae2.xl3.dfw7.alter.net (152.63.97.57)  11.471 ms  11.993 ms  12.395 ms
    13  0.ge-6-3-0.dfw01-bb-rtr1.verizon-gni.net (152.63.96.42)  14.077 ms  29.153 ms
        0.ge-3-0-0.dfw01-bb-rtr1.verizon-gni.net (152.63.1.17) 17.962 ms
    14  p14-0-0.dllstx-lcr-05.verizon-gni.net (130.81.27.40)  14.629 ms  12.297 ms  12.839 ms
    15  * * *
    16  static-xxx.xxx.206.19.dllstx.fios.verizon.net (xxx.xxx.206.19)  24.976 ms  22.170 ms  22.376 ms
    Ping:
    First Run: Ping has started…
    PING xxx.xxx.206.19 (xxx.xxx.206.19): 56 data bytes
    64 bytes from xxx.xxx.206.19: icmp_seq=0 ttl=242 time=22.814 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=1 ttl=242 time=24.621 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=2 ttl=242 time=24.711 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=3 ttl=242 time=24.109 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=4 ttl=242 time=23.336 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=5 ttl=242 time=25.644 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=6 ttl=242 time=27.755 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=7 ttl=242 time=25.135 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=8 ttl=242 time=22.443 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=9 ttl=242 time=24.635 ms
    --- xxx.xxx.206.19 ping statistics ---
    10 packets transmitted, 10 packets received, 0.0% packet loss
    round-trip min/avg/max/stddev = 22.443/24.520/27.755/1.448 ms
    Second Run: Ping has started…
    PING xxx.xxx.206.19 (xxx.xxx.206.19): 56 data bytes
    64 bytes from xxx.xxx.206.19: icmp_seq=0 ttl=242 time=27.183 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=1 ttl=242 time=24.629 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=2 ttl=242 time=22.511 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=3 ttl=242 time=39.620 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=4 ttl=242 time=26.722 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=5 ttl=242 time=23.183 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=6 ttl=242 time=25.171 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=7 ttl=242 time=24.412 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=8 ttl=242 time=23.837 ms
    64 bytes from xxx.xxx.206.19: icmp_seq=9 ttl=242 time=23.785 ms
    --- xxx.xxx.206.19 ping statistics ---
    10 packets transmitted, 10 packets received, 0.0% packet loss
    round-trip min/avg/max/stddev = 22.511/26.105/39.620/4.713 ms
    (3) Any other ideas on what my call quality problem might be, or how I can tweak it?  For example, would putting a DIR-655 router at Location A and enabling QOS really make a difference?
    Thanks to everyone, and I hope this is not too long or difficult to understand.

    Hey thanks for your reply  Yeah im only getting 1 ro sometimes 2 bars reception so hopefully the antenna will beef things up but I think it is what it is perhaps.  

  • Making a VoIP call with the Cisco 837 ADSL router

    I would greatly appreciate if could please provide some technical assistance to my questions below:
    Is it possible to make a VoIP call between two 837 ADSL Cisco routers over a 1Mbps ADSL broadband connection?
    If so, can I configure this VoIP connection using either a PPPoE or ATM WAN link?
    Is it possible to make a VoIP call using a Cisco 837 Router while simultaneously surfing the Internet? In other words do I need two public IP addresses i.e. one for accessing the internet and one for making the VoIP call or is one static IP address obtained from my ISP sufficent.
    It is possible to configure QoS parameters (e.g. RSVP, Voice precedence, Voice codec selection) on this 837 router using PPoE or can it only be done using an ATM WAN interface?
    Does the Cisco 837 router support both the H.323 and SIP communication protocols? Do I need to purchase a certain IOS operating system version for VoIP calling?
    Does the VoIP dial peers need to be configured with both a POTS and VoIP phone numbers or is only one number required?
    Do I need to obtain a special VoIP number from my VoIP service provider? or can I use existing POTS numbers or made up numbers within the dial peers as this situation involves making a private VoIP call between two branch offices using 837 ADSL routers and not via a VoIP service provider.
    Finally, can I use POTS ordinary telephones with the Cisco 837 for making VoIP calls or do I strictly need to purchase VoIP phones?
    My apologies for the number of questions asked here but I currently need to know the technical ability of the Cisco ADSL 837 as I am thinking of employing these routers in my company organisation.
    I await your feedback in due course.
    Thanks,
    Martin Healy

    Hi,
    I give you a sample config of my router.
    class-map voice
    match access-group 101
    policy-map mypolicy
    class voice
    priority 128
    class class-default
    fair-queue 16
    ip subnet-zero
    gateway
    interface Ethernet0
    ip address 20.20.20.20 255.255.255.0
    no ip directed-broadcast (default)
    ip route-cache policy
    ip policy route-map data
    interface ATM0
    ip address 10.10.10.20 255.255.255.0
    no ip directed-broadcast (default)
    no atm ilmi-keepalive (default)
    pvc 1/40
    service-policy output mypolicy
    protocol ip 10.10.10.36 broadcast
    vbr-nrt 640 600 4
    ! 640 is the maximum upstream rate of ADSL
    encapsulation aal5snap
    bundle-enable
    h323-gateway voip interface
    h323-gateway voip id gk-twister ipaddr 172.17.1.1 1719
    h323-gateway voip h323-id gw-820
    h323-gateway voip tech-prefix 1#
    router eigrp 100
    network 10.0.0.0
    network 20.0.0.0
    ip classless (default)
    no ip http server
    access-list 101 permit ip any any precedence critical
    route-map data permit 10
    set ip precedence routine
    line con 0
    exec-timeout 0 0
    transport input none
    stopbits 1
    line vty 0 4
    login
    voice-port 1
    local-alerting
    timeouts call-disconnect 0
    voice-port 2
    local-alerting
    timeouts call-disconnect 0
    voice-port 3
    local-alerting
    timeouts call-disconnect 0
    voice-port 4
    local-alerting
    timeouts call-disconnect 0
    dial-peer voice 10 voip
    destination-pattern ........
    ip precedence 5
    session target ras
    dial-peer voice 1 pots
    destination-pattern 5258111
    port 1
    dial-peer voice 2 pots
    destination-pattern 5258222
    port 2
    dial-peer voice 3 pots
    destination-pattern 5258333
    port 3
    dial-peer voice 4 pots
    destination-pattern 5258444
    port 4
    end

Maybe you are looking for

  • Can I watch my slideshows on my television via a USB cable and if so how

    I'm new to Apple so still learning! I wold like to view my slideshows on my television via an USB cable is this possible and how do I do it ?

  • Strange behaviour in JEditorPane

    Hi all, I'm using JEditorPane for displaying html content. Following is a sample class which initialize JEditorPane, set html content type and finally set some text into the editorPane. public class TestComponent extends JPanel{ JPanel rootPanel = ne

  • Simple Question: Building a basic results page

    I've followed the Dreamweaver CS3 help file: "Building a basic results page" in building a simple search and results page set for a very simple MySQL query. After restarting multiple times, I'm not getting a results page, but rather "HTTP 500" -no pa

  • Why is osx lion no longer available?

    today is 8/17/12 and I want to download lion so i can then download mountain lion. Anyone know why lion is no longer available?

  • Production order - Backflush items

    Hi, How to reversal the Production Order compoent Backflush items we are cancel the confirmation for that all operation in that production order, Backflush items not reversed. All material set Backflush for material master MRP view 2 pls advise how r