Voice Calls dropping over WLAN link

Hi,
The problem I am having is that I have 2 x 350 bridges set up with a Data Vlan & a Voice Vlan connecting 2 buildings. The Data works ok but the voice drops out occasionally. Cisco IP phones at both sites but connected through 3com switches. I have other sites exactly the same and working no problem. Any ideas

This indicates a loss of RF signal for approximately 1.5 to 2 seconds. Because the phone was not in motion and there were no obstacles in the radio path from the AP to the phone, the conclusion is that another wireless device interfered. The device can be another AP on the same channel

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  • Welcome to the Voice and Video over WLAN Discussion

    Welcome to the Cisco Networking Professionals Connection Wireless Forum. This conversation will provide you the opportunity to discuss issues surrounding Voice and Video over WLAN. We encourage everyone to share their knowledge and start conversations on issues such as bandwidth intensive wireless applications, and any other topic concerning Voice and Video over WLAN.
    Remember, just like in the workplace, be courteous to your fellow forum participants. Please refrain from using disparaging or obscene language or posting advertisements.
    We encourage you to tell your fellow networking professionals about the site!
    If you would like us to send them a personal invitation simply send their names and e-mail addresses along with your name to us at [email protected]

    This is easily done with dial peer statements . The dial peer in your originating router must route the calls to the terminating router first. That would look like :
    dial-peer voice xxxxx voip ( the xxxxx is just a tag)
    destination-pattern 45... (that would route any 5-digit calls beginning with 45)
    session-target ipv4:xxx.xxx.xxx.xxx (ip address of the terminating router)
    If digitones are to be dialed after the connection is established, use the statement:
    dtmf-relay-h.245-alphanumeric
    You could also use a statement to specify the codec to be used:
    codec g711ulaw
    You would need multiple voip dial peers if the calls were going to different routers based on the dialed digits. If all calls are sent to the same terminating router, use all wild cards in the dest-pattern statement.
    At the terminating router configure pots dial peers:
    dial-peer voice xxxxx pots
    dest-pattern 45...
    port x/x (whichever port the call is to be terminated on)
    prefix 45 (this re-inserts matched digits which are stripped off by the pots dp)
    Repeat for other ports which will receive calls.
    Paul

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    Try SMS & CallerID Speech Alert from https://market.android.com/details?id=com.webcipe.SpeechAlertLegacy&feature=search_result
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                       Hello,
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    Sahar Hanna

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    Is the Pearl antenna inherently weak?
    Is there a way to boost antenna power on the phone?

    Its not my intention to resurrect dead threads but I just had this very same issue with my 8120 and Rogers in Canada. I thought I would share my solution as I came upon this thread while looking for a solution.
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    So if your phone is functioning and can browse the internet through wifi then you have satisfied 2 of the 3 requirements.
    To test this I used the "Manage Connections" icon to turn off my mobile network. Then I verified that I could browse the internet through wifi and send and receive email but I could not make calls. In my case that just left the phone company.
    SO I called Rogers and bitched them out that my UMA stopped working and had them monkey around with the account settings until UMA came back on. Problem fixed.

  • External and internal mikes are not automatically switching over either recording or on voice calls

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    Sounds like you need to upgrade to the iPhone 5s
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    No swiping to unlock either, just touch the home button
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    Here is a video of it in action
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    Stephen

    HI,
    Describe in more detail please what you mean by "Crash"
    Can you Video Chat ?
    Can You Audio Only chat ?
    These both use the same ports when making a connection and during the actual Chat
    A Screen Share is  an Audio Chat+ and uses  a random port for the Screen Sharing side.
    As this port is random you cannot set up Port Forwarding or Port triggering (If the router part does that) and you need to use UPnP (Universal Plug and Play).
    UPnP is normally On by default in the later Home Hubs (2 and 3)
    Early versions
    If she happens to be using iChat 3 or earlier though there can be issues as the Hub is capable of VoIP and using the phone over the Internet.
    VoIP uses a protocol over the internet called SIP (Session Initiation Protocol)
    This has internationally agreed ports  of 5060-5063 (it uses one and will try one at time if not available)
    Early version of iChat  also used port 5060 to make connections.
    However some ISPs started to Block these ports and then charge extra for this "extra" service (sometimes without notification)
    Some Modems that can "route' the Phone over the internet also need to have setting changed to allow iChat to work,  (Basically the device has a SIP filter regardless of the addresses on the data packages meaning they don't get to the computer and get "lost" as they are not addressed to the Phone either).
    Later versions of iChat moved away from the 5060 port(s) and used 16402 as the first choice.
    Skype that also uses SIP to make connection have never used the 5060 range.
    If there is  a error message about Unexpectedly Quit we could do with seeing the "Crash Log" from her Home Folder/Library/Logs/CrashReporter/iChat.
    11:15 PM      Friday; April 6, 2012
    Please, if posting Logs, do not post any Log info after the line "Binary Images for iChat"
      iMac 2.5Ghz 5i 2011 (Lion 10.7.3)
     G4/1GhzDual MDD (Leopard 10.5.8)
     MacBookPro 2Gb (Snow Leopard 10.6.8)
     Mac OS X (10.6.8),
    "Limit the Logs to the Bits above Binary Images."  No, Seriously

  • I had to change my sim card due to call drop outs and now i have lost important voice messages. How do i get the messages back?

    Please help. I lost very important voicemail when i changed my sim card due to call drop out. How do i get my voicemail back??

    If you mean that Find My Phone is asking for a password to a different Apple ID to your current Apple ID and that ID is a previous version of your current ID, not an entirely different one.
    This feature has been introduced to make stolen phones useless to those that have stolen them.
    However it can also arise when the user has changed their Apple ID details with Apple and not made the same changes to their iCloud account/Find My Phone on their device before upgrading to iOS 7, or if you restore from a previous back up made before you changed your details and some other circumstances.
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    You should then change your Apple ID back to its current state, save it once again and then log back in using your current Apple ID. Finally, turn "find my phone" back on once again.
    This article provides more information about Activation Lock.
    This is answer is provided from my own database of boilerplate responses and the content was last reviewed and tested on: 2014/05/23

  • Route ISDN Voice calls over LAN/WAN

    Is it possible within the relms of normal Cisco LAN  to have an ISDN Line coming into one building, Connect it to a LAN/WAN so we could pick up the voice calls on the ISDN on the other end, In a different location?

    There are no CCM logs to tell you that, CUCM has nothing to do with the RTP, this is usually QoS issues, you always hear "nothing has changed" but they often overlook things they think are not worth mentioning, or that they think would have nothing to do. Make sure really nothing has changed.
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  • In Siri, I can call by my voice, but why I can not use Siri voice call in my country (Laos), just can call only us phone number; my country we use like 3 number for option call, 8 numbers for call friend but Siri cannot use this please help us, thanks

    In Siri, I can call by my voice, but why I can not use Siri voice call in my country (Laos), just can call only us phone number; my country we use like 3 number for option call, 8 numbers for call friend but Siri cannot use this please help us, thanks
    And please help me can type Laos font in it like andrio phone.

    Hi Cozumel,
    Thanks for posting. I'm sorry you're having problems with your bills. I can take a look at this for you. Drop me an email with your account details and a link to this thread for reference. You'll find the address in my profile.
    Cheers
    David
    BTCare Community Mod
    If we have asked you to email us with your details, please make sure you are logged in to the forum, otherwise you will not be able to see our ‘Contact Us’ link within our profiles.
    We are sorry but we are unable to deal with service/account queries via the private message(PM) function so please don't PM your account info, we need to deal with this via our email account :-)

  • Does iOS 5 has features such as Block certain caller, record voice call, 3G Video call (GSM), Personalized Text Ringtone, Voice Conference call, etc?

    Does anyone know whether iOS5 has features such as:
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    - Record Voice Call
    - 3G Video Call (GSM)
    - Personalize Text Ringtone
    - Conference Call

    According to what's been published so far:
    Leewing wrote:
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    - Record Voice Call
    - 3G Video Call (GSM)
    - Personalize Text Ringtone
    - Conference Call
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    Unknown, probably NO- Record Voice Call
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  • FXO traffic is going over Wan link after enabling QoS

    Hi,
    We have CCM 5.0,2811 voice gateway,1 4FXO and 1 g.shdsl card on it. 4 Telco lines are on 4FXO card.CCM is at the other side of g.shdsl line.Topology like that,
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    4fxo g.shdsl
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    Thank you

    Check your H323 gateway config in CCM to be sure the Media Termination Point Required box is unchecked. Using MTP forces the call to terminate on the closest MTP resource which is likely at your CCM site.
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  • Is it possible that Exchange UM could be configure with two call managers over the same sip?

    Hi,
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    Thank you,
    Peter

    Hi,
    I have Cisco call manager 8.2 integrated with Microsoft Exchange Server 2010 Unified Messaging.
    Call manager has primary and secondary server. I created a sip trunk and linked primary CUCM with Exchange. Users can leave and get voice mails.
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    If not:
    Does exist a way to configure Exchange UM that will work if one CUCM server is down?
    Thank you,
    Peter

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