SPA3102

I have some questions as below listed:
1. Is SPA3102 can handled more than one call session at moment?
2. How can i share the VoIP account in the LAN for other users?
3. How many IP phones can connect to one SPA3102 for concurrent call sessions? (we have only one VoIP carrier account from lowratevoip.com)
4. please let me have a solution for multiple users and concurrent calls in the LAN with one VoIP carrier account?

1. Is SPA3102 can handled more than one call session at moment?
the SPA3102 can handle 2 lines for line 1 , 1 for main call and 1 for call waiting.
from the analog phone, you can also setup a dial plan for dialing out to 4 other gateways. the PSTN line has a 1 gateway call.
2. How can i share the VoIP account in the LAN for other users?
if you have for that one ATA, you can set those others to make call without regsiter to yes and answer call to yes, and regsiter to no.
3. How many IP phones can connect to one SPA3102 for concurrent call sessions? (we have only one VoIP carrier account from lowratevoip.com)
this one is not possible. the SPA3102 cannot be used as a voip gateway or PBX
4. please let me have a solution for multiple users and concurrent calls in the LAN
with one VoIP carrier account?
consider the SPA9000 PBX system that can can have mutiple sessions depending on the trunk line provided by the VOIP provider for you.

Similar Messages

  • SPA3102 all the time in flashing mode

    Hi, SPA3102 doesn't receive IP from DHCP and still flashing signal LED. I can access it via http and rebooted it several times via phone and 2 times flashed the newest firmware but all this for nothing - LED is blinking all the time.
    Please help me.

    I think it is bootloader going on and something in soft is bad. How can I do my gate in state SOS ?

  • How to set a *long* SIP password in SPA3102?

    Hello,
    I'm trying to configure the PSTN Line of a SPA3102 with a service provider that requires a long password. It is 64 characters long, and I suspect that is the reason the registration is not working.
    I'm pretty sure that everything else is OK. I have enabled the syslog debugging feature, so all the SIP packets exchanged during the registration are dumped. The conversation is roughly the same that happens when registering from Asterisk (except for the Call-ID and the User Agent, obviously), but the authentication values look wrong. In fact, the service provider's proxy keeps answering "407 Proxy Authentication Required".
    Has anyone else tried this? Who should I contact to know if it is a firmware bug and if there is a chance that it is fixed in the future?
    In case it is possible, it would be perfectly fine with me to send the password, or the precomputed HA1 value, over XML provisioning -- I'd just like the adapter to work with this service provider.
    Thanks!

    You have mentioned that you are registering the PSTN line of the SPA-3102
    to VOIP provider which provides long password.
    I need to know who is the service provider and description of network
    setup. If the SPA-3102 is connected directly to the modem and internet is
    working fine, then there could be a problem with the VOIP information from
    your service provider or VOIP ports are block by your ISP.
    Have you called your VOIP provider about this? Have you tried registering
    the SPA-3102 using free VOIP account like FWD and see if that works?

  • SPA3102 - Line 1 - Outgoing VOIP calls fail

    I recently reset my SPA3102.
    After configuring Line 1; I am able to receive VOIP & PSTN calls on the analog phone attached but unable to make outgoing VOIP calls, I can only make outgoing calls via PSTN.
    My dial plan is as follows:
    (xx.|<#9,xx.<:@gw0>)
    I can call the outside world by dialing #9 and then the phone number but I cannot call any VOIP extensions, when I try, I do I get the busy tone after a few seconds.
    Please help, I’ve spend hours trying to figure this out to no avail.
    Solved!
    Go to Solution.

    Thanks for your response,
    I finally resolved the problem after spending many more hours on it.
    It turns out the problem had nothing to do with my dial plan, it had to do with a service provider (SPA3102) I had configured on my asterisk box; adjusting the configuration on that solved the problem.

  • SPA3102 - Some Fine Tuning Query

    Hi,
    My SPA3102 works fine answering PSTN and VoIP calls, including call waiting and 2-way switching and 3-way calls, caller ID pass thru, etc.
    But if I have the phone "off hook" for any reason and a PSTN call comes in, (it rings on another phone directly connected to PSTN line) placing the ATA attached phone on hook does not cause phone to ring through (PSTN ring thru line 1) no matter how long I wait. So I was wondering what parameter affects this aspect as I'm sure I had this working in the past.
    Oh, by way of background information, the PSTN line voltage on the status screen shows typically 49V on hook, 7V off hook and -49V when PSTN is ringing (UK BT line).
    Any comment most welcome - Cheers from Mike

    I believe this is a normal behavior according to the admin guide:
    If Line 1 is busy (fxs port is off-hook) when the PSTN line rings, the SPA will not attempt to ring through, even if Line 1 later becomes idle (putting fxs port on-hook) while the PSTN is still ringing.

  • Back to Back ATA SPA3102 solved as PSTN extension

    Hi all,
    For a first a customer of mine required a 1KM wireless link to provide PSTN from one office to another, the WiFi has professional MicroNet access points in bridge mode with very low latency < 10ms.
    With some help from this forum I 'tuned' the two SPA3102's to work as follows.
    PSTN SPA called 'office'
    Regional TAB
    Customise settings to suit Australian conditions with a default dial tone for internal calls
    FXS port impedance to 600 ohms or Uniden WDECT would not ring always
    More echo suppression: No
    Line 1 TAB
    Line Enable Yes
    Use Proxy / Register No
    Answer // Make call without Registration Yes
    Display name home // ID home
    Dial string (<##:[email protected]:5060>)
    This dial string will call the remote phone with ##
    Enable IP dialling Yes
    PSTN TAB
    Line Enable Yes
    Use Proxy / Register No
    Answer // Make call without Registration Yes
    Display name office // ID office
    Dial Plan 1: (S0<:[email protected]:5060>)
    VoIP to PSTN Gateway enable Yes
    Line 1 VoIP Caller DP: 1
    Line 1 Fallback DP: 1
    *critical* VoIP Call DP: none
    PSTN to VoIP Setup Gateway enable Yes
    PSTN ring thru Line 1 No
    PSTN Default DP: 1
    VoIP Answer Delay: 0
    PSTN Answer Delay: 2
    Line in use voltage 30
    If a 'fax duet' is in use it can be ignored with default Ring Validation Time: of 256MS
    If a 'fax duet' is in use it can ring the remote phone with Ring Validation Time: of < 150MS
    Remote SPA called 'remote'
    Regional TAB
    FXS port impedance 600 ohms
    More echo suppression: no
    Customise settings to suit Australian conditions with a default dial tone for internal calls
    Line 1 TAB
    Line Enable Yes
    Use Proxy No
    Register No
    Make call without reg // Ans call without reg: yes
    Display name remote // ID remote
    Dial Plan: (<0: [email protected]:5061>|<##: [email protected]:5060>)
    This Dial Plan provides a PSTN dial tone if the caller goes off hook and presses '0' // zero
    This Dial Plan rings the other phone at the PSTN end as an intercom if the caller goes off hook and presses '##'
    If a PSTN dial tone is required immediately the Dial Plan is: (<S0: [email protected]:5061>)
    Enable IP Dialling Yes
    The rest of the configuration on the two SPA3102's is default. If in doubt, reset them and try again.
    When an inbound PSTN call was made, the 'remote' phone took two rings to respond, that is, the caller would hear two rings or more prior to the callee hearing any ring.
    Also a missed call would see the 'remote' phone ring least two times after the caller had hung up, all part of the technology as far as I could tell, anyway everyone seems happy to have a phone where none was ever possible previously.
    Inter office messaging as an intercom was excellent, the phones responded immediately and quality was as expected.
    I found that the recommended port impedance for the Uniden WDECT-33xx phones in use would not always ring, so 600 Ohms seemed a reliable choice and with the client possibly installing a 'fax duet' service for answer only at the first office where the PSTN was located, it is possible to have the remote or extension phone ignore that 'burr burr burr' ring sequence by using a Ring Validation of 256MS or more, if it is wanted to answer change this setting to 150MS or less and try again.
    Message Edited by AU.IFIX on 05-27-2009 02:03 PM
    Message Edited by AU.IFIX on 05-27-2009 02:05 PM

    Hi all,
    For a first a customer of mine required a 1KM wireless link to provide PSTN from one office to another, the WiFi has professional MicroNet access points in bridge mode with very low latency < 10ms.
    With some help from this forum I 'tuned' the two SPA3102's to work as follows.
    PSTN SPA called 'office'
    Regional TAB
    Customise settings to suit Australian conditions with a default dial tone for internal calls
    FXS port impedance to 600 ohms or Uniden WDECT would not ring always
    More echo suppression: No
    Line 1 TAB
    Line Enable Yes
    Use Proxy / Register No
    Answer // Make call without Registration Yes
    Display name home // ID home
    Dial string (<##:[email protected]:5060>)
    This dial string will call the remote phone with ##
    Enable IP dialling Yes
    PSTN TAB
    Line Enable Yes
    Use Proxy / Register No
    Answer // Make call without Registration Yes
    Display name office // ID office
    Dial Plan 1: (S0<:[email protected]:5060>)
    VoIP to PSTN Gateway enable Yes
    Line 1 VoIP Caller DP: 1
    Line 1 Fallback DP: 1
    *critical* VoIP Call DP: none
    PSTN to VoIP Setup Gateway enable Yes
    PSTN ring thru Line 1 No
    PSTN Default DP: 1
    VoIP Answer Delay: 0
    PSTN Answer Delay: 2
    Line in use voltage 30
    If a 'fax duet' is in use it can be ignored with default Ring Validation Time: of 256MS
    If a 'fax duet' is in use it can ring the remote phone with Ring Validation Time: of < 150MS
    Remote SPA called 'remote'
    Regional TAB
    FXS port impedance 600 ohms
    More echo suppression: no
    Customise settings to suit Australian conditions with a default dial tone for internal calls
    Line 1 TAB
    Line Enable Yes
    Use Proxy No
    Register No
    Make call without reg // Ans call without reg: yes
    Display name remote // ID remote
    Dial Plan: (<0: [email protected]:5061>|<##: [email protected]:5060>)
    This Dial Plan provides a PSTN dial tone if the caller goes off hook and presses '0' // zero
    This Dial Plan rings the other phone at the PSTN end as an intercom if the caller goes off hook and presses '##'
    If a PSTN dial tone is required immediately the Dial Plan is: (<S0: [email protected]:5061>)
    Enable IP Dialling Yes
    The rest of the configuration on the two SPA3102's is default. If in doubt, reset them and try again.
    When an inbound PSTN call was made, the 'remote' phone took two rings to respond, that is, the caller would hear two rings or more prior to the callee hearing any ring.
    Also a missed call would see the 'remote' phone ring least two times after the caller had hung up, all part of the technology as far as I could tell, anyway everyone seems happy to have a phone where none was ever possible previously.
    Inter office messaging as an intercom was excellent, the phones responded immediately and quality was as expected.
    I found that the recommended port impedance for the Uniden WDECT-33xx phones in use would not always ring, so 600 Ohms seemed a reliable choice and with the client possibly installing a 'fax duet' service for answer only at the first office where the PSTN was located, it is possible to have the remote or extension phone ignore that 'burr burr burr' ring sequence by using a Ring Validation of 256MS or more, if it is wanted to answer change this setting to 150MS or less and try again.
    Message Edited by AU.IFIX on 05-27-2009 02:03 PM
    Message Edited by AU.IFIX on 05-27-2009 02:05 PM

  • Issue connecting SPA3102 FXO port to paging interface

    Hello,
    I was referred to this community after a call to small business support, we were unable to come to a solution.
    SPA3102 v5.2.13 FW
    Viking Electronics PI-1 Paging Interface
    We were told from Viking that the interface only supports an FXO port, all the connections were verified with their support and we're able to page by connecting an analog phone straight to the interface. Plugging into the FXS port on the SPA3102 also results in a dial tone playing over the paging system.
    All settings on the SPA3102 are default minus SIP registration information, we see the unit as registered within the web interface as well as within our Asterisk CLI.
    Trying to ring the extension results in constant ringing if voicemail is disabled, and straight to voicemail if enabled. Within the Asterisk CLI we see a SIP response 503 "Service Unavailable."
    I can get the SPA3102 to respond by lowering the "Line In Use Voltage" down from its default value of 30. However all I hear is a hum and I'm not able to speak over the paging system.
    Any insight into this issue would be greatly appreciated, we believe it's just settings on the SPA3102 unit itself that need to be changed.
    Thanks!

    Looking at the Viking Pl-1 web page installation instructions, I think the key setting on the SPA3102 would be to set the voip-to-pstn gateway dial plan to NONE.  The "hum" you get when you lower the Line-In-Use setting sounds like the dial tone you would get with the default dial plan setting.
    I would try the following:
    Viking Pl-1: 
    Talk Battery Switch (PT): On
    Audio In: Cabled to FXO port of SPA3102
    SPA3102:
    Interface Analog Phone attached to SPA3102:
    Line 1 Tab
    Line Enable: YES
    Dial Plan: (S0<:@gw0>)
    PSTN Line Tab
    Line Enable: Yes
    VoIP-to-PSTN Gateway Enable: Yes
    One Stage Dialing: Yes
    Line 1 VoIP Caller DP: NONE
    VoIP Answer Delay: 0
    Line In Use Voltage: xx
    The Line In Use Voltage needs to be lower than the Talk Battery voltage supplied by the Viking Pl-1.  Usually it is set about half way between the on-hook and off-hook voltage level.  Measure the FXO (PSTN Line) on-hook voltage by reading it on the SPA3102 INFO Tab.  Set the Line In Use Voltage substantially below the on-hook talk voltage.
    If you lift the phone with the above settings you should be connected to the paging system.
    Interface to an asterisk pbx system:
    PSTN Line Tab
    Setup Registration on PSTN Line Tab to Asterisk system. 
    Register: Yes
    Proxy: xxx
    UserID: xxx
    Password: xxx
    VoIP-to-PSTN Gateway Enable: Yes
    VoIP Caller Auth Method: None
    One Stage Dialing: Yes
    VoIP Caller Default DP: NONE
    VoIP Answer Delay: 0
    Line In Use Voltage: same comments as above
    If you call the extension on the asterisk PBX you should be attached to the Viking unit.

  • SPA3102 questions

    Hey all, I just read the admin guide for the SPA 3102 but I wanna make sure I didn't misunderstand anything so, could someone confirm that I can do all the stuff below, and note I'm in the UK.
    1. Register 1 voip provider (VA) for outgoing calls, 1 different voip provider (VB) for incoming calls, and connect 1 pstn line for incoming calls.
    2. Connect a single analogue phone and recieve calls on it automatically from anyone calling either my voip (VB) or pstn number, with Caller ID from either being presented correctly.
    3. Make calls on the same phone via either the voip (VA) or pstn line (with Caller ID being sent correctly).
    4. Automatically decide which of VA or PSTN I want to go out with by dialplans, down to specific numbers (eg. all 0800 go out pstn, all starting 07, 01, 02 out via VA, 999 out PSTN, 0113212121 out PSTN). And fallback to PSTN on voip failure or power loss to spa.
    5. Any form of recording how many calls have been made, and whether these went out via PSTN or VA (and possibly if they were successful or not)? Or any information that could allow a list of these calls to be made? Just saw in the user guide there seems to be only 1 field for Last Number Called for both voip and PSTN, maybe a script could monitor this for changes and update a list on every change, then i'd have a list of all numbers dialed but not if they were succesful or not...
    6. A fun one - gain telnet/ssh access and change dialplans via the command line, even create a script on the thing to check dialplans and dialed numbers and change dialplans automatically (I want to do this based on number of pstn calls made to get around bt's "6 chargeable calls per quarter or we charge you for caller id" rule, and preferably do it via script automatically cos i'm geeky like that, though I could script on my wrt54g and have it telnet to an ata but only if i can...).
    Thanks for the help, pepsi_max2k
    oh ps, i'm loving the amount of threads i'm finding in google on problems with the spa3102 in the uk, followed immidiately by "it's fine now, just did (insert very simple external fix - new cable / mf / phone) and it works great" type posts if only all hardware fixed itself as easily as this thing seems to
    Message Edited by pepsi_max2k on 06-01-2008 04:29 AM

    thanks for that so basically, yes, it does everything i asked other than pstn fallback when voip fails. i can deal with that well, there is the no ssh thing that sucks a bit. other than multi voip accounts it then means my speedtouch is just as good (actually better as it has telnet access and logging, although only for voip calls).
    i just got a cheap fritzbox 5050 so i'm gonna see what I can do with that, hopefully i can get some decent dialplans working, and if it does everything else above then i'll be chuffed. I know it does telnet, dunno if i can grab textual call stats from there but there's gotta be a way, i know it does very detailed stats though (esp. for pstn), multi voip accounts (both in and out), CLIP should be ok so long as I can get wiring right, dialplans seem a bit simple but might just allow me to do what I want (again whether I can change these via telnet I'm not sure)...
    Anyway, still might get an SPA just as it seems pretty reliable, and I know what it can / can't do
    Message Edited by pepsi_max2k on 06-03-2008 08:34 AM

  • P2P SPA3102 setup with no SIP server/service provider

    Description:
    - Peer-to-peer ATA connection with no service provider or SIP server
    Location A (my location):
    - Linksys SPA3102
    - PABX analog device (Panasonic)
    - Public and static IP
    Location B (my partner location):
    - D-Link DVG2001S
    - Public IP
    Questions:
    - Can B ATA (D-Link) call to A ATA (Linksys) being different brands?
    - Can I pick up the B ATA attached phone and directly to have PABX internal dial tone at A, for other extension or external calls?
    - From PABX other extensions, can I call to SPA3102 extension number and directly redirect the call to B ATA?
    I was read many .pdf documents about SPA setup, and try differents configuration, but I'cant make the communication with my partner location happens.
    I think my problem is understanding and writing the correct PSTN LINE and LINE 1 Dial Plan parameters in my SPA3102.
    Can I make it happens?
    Thanks in advance,
    Bitman
    Message Edited by Bitman on 07-28-2007 05:37 AM

    Good day! I just hope this is not one of those “.pdf documents” that you have already read but basically this should work if you’re setting up two SPA3000 or SPA3102  since they have exactly the same Voice configuration:
    http://www.provu.co.uk/pdf/sipura/spa_backtoback_2x_spa3000.pdf
    I’m guessing that IP dialing should work on your two units if they both support SIP,  however, in this case, I haven’t tried setting up a D-Link before ( or you may try to wait for other users who happen to have a D-link also to post suggestions) , you may just want to try looking for documents on setting up IP dialing on D-link then try to match it’s IP dialing parameters with the SPA.

  • SPA3102 Non-standard SIP port in dial plan

    Hi, a VoIP provider uses port 5068 instead of standard 5060. I tried to use it in the dial plan of SPA3102, but failed. Using another provider that uses standard 5060 port I've narrowed down the problem to the following:
    This one works:
    (7495x.<:@sipnet.ru;usr="xxx";pwd="xxx";nat="yes">)
    Adding a SIP port prevents it from working: (7495x.<:@sipnet.ru:5060;usr="xxx";pwd="xxx";nat="yes">)
    Can anyone give any clues how to specify a port in the dial plan?
    Message Edited by V_l_a_d on 06-21-2008 06:45 AM

    As far as I know you just have to change the port.
    (S0 < :userid@spa2_WAN_EXTERNAL_IP:5061 >)
    < 011852,: >xxxxxxx< :@gw.macau-tel.com:5080;usr=Joe;pwd="90f-fkd";nat=yes >

  • Spa3102 would not forward a voip call to pstn line

    Good morning.
    I've done the implementation provided here http://community.linksys.com/t5/VoIP-Adapters/SPA-3102-and-softphone-to-
    make-calls-via-pstn-line/td-p/326390 .
    It is a way to use for outgoing calls a given pstn line from anywhere I have internet (voip to pstn).
    The spa3102 is connected to a router (with an active DHCP server and ip 192.168.1.1) from where it takes the internal
    ip (192.168.1.3).On the same network is also a computer , connected to the router ( with ip 192.168.1.2). The spa3102
    is set to bridge mode and thus inactivates the function of the router (on SPA3102), and it functions as a  simple
    network device . I have  done port forwarding (from the router) to 192.168.1.3 (SPA3102) for the port 5061 (PSTN
    LINE) ( but for 5060 for the LINE 1 also). I want to make calls from a voip softphone (x-lite 4) to the SPA 3102 and
    this to forward the voip calls to PSTN line to which it is connected. In x-lite the SPA3102 is set as a proxy so that
    i can type the phone number I want to call without being followed by the SPA3102's ip each time ( eg on  x-lite I
    give call number 2101111111 instead of 2101111111 @ wanip: 5061 where wanip is the external ip of the router).
    When x-lite is running on the computer that is on the same network with the SPA3102 everything works as expected. A
    voip call is made from x-lite ( using as a proxy the wanip everytime, or even for test purposes the dyndns domain
    that i set up for this reason), this call is passese PSTN line and the phone of the called party rings . At x-lite
    COMES indication "call established ".
    The problem occurs when I do the same procedure from x-lite installed on a computer belonging to another network (
    e.g. in another building with its own internet connection , own router, own computer , etc. ) . Always using the
    wanip the x-lite makes the voip call to the SPA3102, writes "call established" ( meaning it connected to SPA3102) but
    never routed the call to the called party ( the SPA3102 did not forward voip calls it receives to the PSTN line ) .
    Trying to find what 's wrong I've tried to disable all firewalls (soft and hard from all involved machines ) . The
    behavior is the same either the computer that makes the successful calls is connected to the network directly to the
    router  or through the port "ethernet" on the SPA3102.
    What is the difference in these two voip calls to the SPA3102 and the one  " triggers "  it to forward the call to
    PSTN line and the other does not ?
    Thanks now for any ideas you give .

    The audio sound problem is more than likely also associated with the overall addressing problem initially encountered.  As you may know, using the sip protocol the sip signalling exchanges ip addresses to be used for both the sip signalling and the exchange of rtp sound packets.  In addition there is an exchange of port numbers to be used for the exchange of rtp sound packets.  The sound is exchanged by two separate streams of packets, one stream in each direction.  The result is an ip address and port number for the rtp packets flowing from the SPA3102 to the softphone and a different ip address and port number for the rtp packets flowing from the softphone to the SPA3102.
    In your previous posting you mentioned that you "set the minimum  EXTernal rtp port at the sip tab".  Changing the "EXT RTP Port Min:" is an unusual change to make and in my opinion would only be made in special circumstances. Actually, I ran some tests and I'm not sure exactly what that setting does.  In my tests it didn't appear to affect the rtp port number used in a predictable manner.
    The common changes to make for audio problems typically would be to setup a STUN server.  A STUN server is an external server that echos back to the initial sender the external ip address and port number that the STUN server received with the message received by the server.  This allows the sender (SPA3102 or softphone) to determine its external ip address and external port numbers for both the sip signalling and rtp packets.
    A STUN server is commonly recommended to be setup with the following settings in the SPA3102:
    PSTN Line Tab:
    NAT Mapping Enable: Yes
    Sip Tab:
    Handle VIA received: yes
    Handle VIA rport: yes
    Insert VIA received: yes
    Insert VIA rport: yes
    Substitute VIA Addr: yes
    Send Resp To Src Port: yes
    STUN Enable: yes
    STUN Server:
    The following web page has a list of "Public STUN Servers"
    http://www.voip-info.org/wiki/view/STUN
    You are using CounterPath's XLite softphone.  stun.counterpath.net  is a STUN server on the list.
    I see XLite also has a setting to use a STUN server on the "Topology" tab.

  • SPA3102 new firmware 5.1.10 breaks SPA3102!

    OK, so there is a new firmware for the SPA3102 - it's available off the Cisco.com website! However it seems if you have h/w version 1.1.5 it breaks the ability to adhere to the chosen "preferred codec". For some reason, if i select G729a as my preferred codec, the unit will for some reason select G711u at the active codec to place the VoIP call. This behaviour was not evident with firmware 5.1.7 - and it's not a VSP issue!
    I've tried to fatory default the unit to resolve the problem but no luck! Tried to factory default and roll back to 5.1.7 and then to 5.1.5(a) and then 3.2.10 and stil no luck. (I had a back up of my config from a few months ago whilst using t 5.1.7 and tried to restore to that, (in case the upgrade to 5.1.10 screwed some settings) after factory defaulting whilst running 5.1.7 and still no luck)
    Anyway here is a therad with all the issues we've had http://forums.whirlpool.net.au/forum-replies.cfm?t=1130617
    So anyone from Linksys Cisco want to explain what is going on here?

    briggity wrote:
    I just updated to Safari 5.1.4 via software update. My bank is FirstBank - efirstbank.com - now when I attempt to login, it goes into an endless loop; it appears to be a javascript issue. The site opens fine in FireFox. I am running OS 10.7.3, BTW.
    Anyone else having issue with the new Safari? Is there an easy way to downgrade it? I prefer having all my business stuff running in tabs on one browser...
    Brian
    You can always Reset Safari...
    Go to the Apple Menu bar click on "Safari" title and look for the option that says "Reset Safari". You'll see a list of checkboxes with items that will be affected by the operation, uncheck those you don't want to be affected. Once you finish your selection click "OK" and confirm the operation. Safari will close and reopen in an instant.
    This operation clears, caches, cookies, and other miscellaneous items that could be causing your issue.
    Good Luck and keep us posted!

  • Need help rush for a spa3102 special setup welcom gurru

    I have to transport phone line on my network . I just bought two spa3102 that is supposed to do the job. I know how to setup the adresses but don't know anything on how to setup both to communicate together . I just need to take an analog line in spa3102 number one and over the local lan , let the other spa3102 answer like a standard phone . And make call from the second one to the fist one using it's analog line . Could someone help me i'm really in a pain with this ...
    Please your help will be appreciate
    thanks !

    Hello MF DOMO,
    try to read this article :
    http://www.provu.co.uk/pdf/sipura/spa_backtoback_2x_spa3000.pdf
    or other articles under the Linksys heading on the page :
    http://www.provu.co.uk/support.html

  • Standard SRTP (SDES) Support in SPA3102

    Hi,
    I would like to know the progress of Standard SRTP (SDES)  support in SPA3102. With users growing concern over pircary and VoIP being used as a replacement for the traditional PSTN networks, it would be great if SPA3102 supports the standard Secure RTP protocol.
    Also, are there any gateway or software implementations (in linux) which can talk the present propriety sipura SRTP protocol?
    Thanks,
    Rahul.

    Hi Alberto,
    Thanks for your prompt response. It is really sad to hear that SDES will not be supported in Sipura SPA3102. I was under the impression that it was on the roadmap but not an immediate priority.
    Also I am not a VoIP Service Provider but an individual. But I use VoIP extensively and sometimes convey sensitive information such as Credit Card details. I don't understand why the sample code should only be available for VoIP service providers. Isn't it possible that the sample code be given to an individual who has purchased SPA3102?
    I fail to see the need for allowing only VoIP providers to access this code. Any help in this regards would be greatly appreciated.
    Regards,
    Rahul.

  • SPA3102 FXS port

    Hello,
    I have a problem with my SPA3102's FXS port.
    In last April I tested my friend's two PAP2T adapters with four phones. The adapters were connected to same switch with different fix IP addresses and phones were able to ring each other. Then I got one of adapters from him as a gift and on 2nd of January I bought an SPA3102 - so I have two adapters.
    I connected my adapters to same switch. Their parameters are:
    PAP2T IP: 192.168.0.221
      FXS #1:
        SIP port: 5060
        display name: 1
        user ID: 1
      FXS #2:
        SIP port: 5061
        display name: 2
        user ID: 2
    SPA3102 IP: 192.168.0.222
      FXS:
        SIP port: 5062
        display name: 3
        user ID: 3
    Dialplans:
    PAP2T Line1:        (<2:>S0 <:[email protected]:5061>|<3:>S0 <:[email protected]:5062>|<[x*]:>S0 <:[email protected]:5060>)
    PAP2T Line2:        (<1:>S0 <:[email protected]:5060>|<3:>S0 <:[email protected]:5062>|<[x*]:>S0 <:[email protected]:5061>)
    SPA3102 FXS:      (<1:>S0 <:[email protected]:5060>|<2:>S0 <:[email protected]:5061>|<[x*]:>S0 <:[email protected]:5062>)
    Well there is problem with third phone (SPA3102 FXS port). Phones with PAP2T work perfectly, they can call each other. The 3rd phone has dialtone sometimes only but it is unable to call the other two phones. "1" and "2" can call third phone sometimes but another time I get reorder tone when I dial "3".
    I have made backups from adapters' settings. Does somebody have any idea about this problem?
    Thanks in advance,,,

    Hi,
    Problem is solved... 
    SPA3102 has two RJ45 outlets: LAN and WAN (internet). I have used LAN only because of switch and the INTERNET terminal was empty. It appears to me this situation drives SPA3102 crazy...    I had to make a "loopback" plug (I linked contact 1 to contact 3 and contact 2 to contact 6) and insert it into INTERNET outlet. So all of phones work well...
    If you want to use a PAP2T lonely (without internet access) as a "micro-PBX" with two extensions, you will have to use same plug otherwise phones won't work.

  • SPA3102 - Security Flaw that enables peope to bypass Tolls

    Hi All,
    Possible security issue with 3102 when used as an FXO Device on a PBX - is there a way to prevent this?
    I'm using several SPA3102s as FXO devices on an IP PBX. Most extensions on the PBX are toll barred so that staff cannot make toll calls. However, there is a way that staff could bypass this security.
    When an external call comes in the SPA3102 detects ringing and passes this onto the IPPBX. The IPPBX then rings a number of extensions which can then be answered in order to speak to the remote party. This is normal and fine.
    However, if the caller hangs up before the call is answered the SPA3102 believes that the line is still ringing for at least 1-2 seconds after the line has stopped ringing.
    In this case, if someone then answers the phone the SPA will go off hook as if it were an incoming call and give the internal extension external dial tone. The PBX thinks it is an incoming call so is not concerned about restricting digits dialled etc.
    In other words, a smart person with access to an extension could use their mobile to call in and hang up just before the phone is answered so that they get direct access to an external line.
    This is an age old problem, but most modern PBX's have very good ring detection and can detect within a very short time period when the ringing signal has stopped.
    The SPA3102 seems to take between 1 and 3 seconds to detect that the ringing has stopped.
    I could possibly create some rules in the IPPBX to sort out this issue, however I would rather fix the root cause of the problem, which is that the SPA passes on ringing to the PBX after the line has stopped ringing.
    I am not exactly sure of the specifics but I believe that the SPA should be able to detect voltage (or polarity?) changes on the line that the telco provides when the line is ringing that stop precisely when the line stops ringing.
    Would changing ring cadence settings in the SPA help fix this problem? Does anyone have any ideas?
    Cheers,
    Jonathan.

    They already fixed it in 2.1 but they havent released it. I read the same thing on iPhoneHacks.com which may be where you read it? hah

Maybe you are looking for