PAP2T vs SPA3000

Similar Questions

• my camera (PAP2T) "cannot connect to Server" Please help me!

  I got this message to my device (PAP2T) internet page "can not connect to access the server" Please help me!

  I knew that my friends who all linksys PAP2T don device won't work no three days in my country the Libya, is this general problem or my country only?

  Thank you

  Try this link.

• PAP2T Dialout disconnects after 15 seconds. Dialin fine...?

  I have the PAP2t, I use Gizmo5 service.

  I can get calls in no problem.

  Outgoing calls, I get a problem - all calls are disconnected after 15 seconds - still the same!  I tried a ton of different settings and reset to the default values, the upgrade of the firmware...  Any ideas?

  Thank you, but I've just fixed.  Had a problem with my (dare I say?) Netgear router - WRV2000.  Latest firmware for the netgear router (from a year or more), all are incompatible.  Had to downgrade firmware for something that worked 1.1.3.9

  Now works like a charm...

• Is unlocked my PAP2T? (difference between PAP2T & PAP2T-NA)

  So I have this PAP2T - is unlocked (i.e. can it be used with any voip service)?   Guess my question is: are all unlocked PAP2T?  or are only the PAP2T-NA is unlocked?  I never used and trying to sell it and wonder if I can sell it as "unlocked."  Thanks for the help!

  J

  If you can set it up yourself, you can unlocked consifer.

  Locked devices usually have a user name and password saved in it or if the VOIP provider is the one that controls the configuration of the remote device.

  If you want to use it to another provider, make sure that you change everything first (* 72738 #) and that it does not ask for any password and you can connect to the web interface. Also make sure that it is there no entry under the rule of profile reset after.

• Fax line down to PAP2T

  I have the VOIP PAP2T adapter to work. The line fax, connection to the adapter does not work. I tried it disconnected / reconnected without success.

  Can someone help me? I connect to the Setup screen, but I do not see the restart feature.

  OK I have done the troubleshooting more

  I re - enter the IP address in the fax parameter.

  Connect analog phone to the fax line, but I couldn't hear the tone. I have heard nothing but static noise.

  I called the phone normal fax line, but I've heard that short high ptch fax sound and he just ERA. However, when I called a different fax machine from outside of the company, I heard his long ground fax for fax.

  Can someone help me?

• Help! PAP2T is defaulting to a different IP address I give.

  I'll set up a new network at home using pfSense as a router.  My old network addresses were 192.168.1.x.  My NEW network addresses are 192.168.15.x. I have ALL other devices work properly on the new network except for my PAP2T phone adapter.  I put it dynamically.  I put it statically.  I put in using the telephone keypad.  I put in using the web interface.  Whenever I make it work, there is the work for an hour or two and then drop off the network.  When I come back and check it out, it has changed its IP address to several 192.168.1.x.  These telephone adaptors work only on this subnet?  If not, why it changes it's IP address and how do I fix it to do?

  Thank you in advance!

  It turns out that my business VoIP replace my settings as soon as the telephone adapter hits their commissioning.  So... after a call to them, they were able to set the network my new to me and it's stuck.  Now... why on earth that they would allow customers access to change their IP settings and then override?  Sheesh!   Well.  Problem solved.

• Very strange interim PAP2T

  Hi, I use pap2t to my CIA/3WEB voip in canada / ontario provider. I use their profile of voip linksys with a modification for use behind the router. The problem I have is every time I reset the adapter and program it to my VoIP provider, unit works fine for about an hour later, I am not able to access the web configuration interface, I can not access the UI interface admin even not, it does not allow me to log in. Later, after a few hours the unit stops and I can't make all the calls and line 1 turns off, so it is not connected at all and still I can't access the user / admin panel to see what is the problem. It is a common problem or is it a bad configuration that changes the passwords, and then disconnects the phone?

  Also another strange think I feel is that Im from a phone nortel Vista 350 and if the line is cut on dysplay says Message waiting and the flashesh of the red light.

  Anyone know it to fix the Pap2T so that it does not change the passwords and keeps him alive on the line?

  My cable connection is always on and by newspapers, I can see that it does not disconnect.

  Thank you.

  nitewulf wrote:

  Anyone know it to fix the Pap2T so that it does not change the passwords and keeps him alive on the line?

  My cable connection is always on and by newspapers, I can see that it does not disconnect.

  So, how do recover you?

  I would try to disable tab commissioning commissioning.  Provision enable Yes allows you to remotely change your configuration.  Some voip providers are configured for this.

  "Keeps alive on the line" means generally that the PAP2 continues to record periodically.  If something is change your settings that might cause a problem.

• PAP2T - direct connection

  I was asked to set up our end of free VoIP (new to this), basically our Australia Office will connect our handsets directly to an office in India - I have no VoIP etc, just this box and Internet plan

  Our facility is PAP2T with fixed internal IP address 10.X.X.X/24, I don't have a public IP address dedicated to this so I'll try to figure out if I can port before the traffic from the router through the device, can someone please help me with the sub questions?

  1. should I wear to the front through the traffic? I have been informed by the other end the ports used are:

  UDP 5060-5061, 16684-16782 UDP, TCP 80 - I just transfer all the ports, or can I get away with transfer only some?

  2. do I need to enter their public IP somewhere? They told me to test by picking up the handset and dial XXXX #, but I do not see how my phone will know where it happens without knowing its IP address.

  3. can you tell me what are the basic settings that I have to configure in the Web interface of the PAP2T to make sure that it works?

  Any help is appreciated more.

  See you soon

  
  

• Guide the administrator Pap2t-NA?

  Where can I get information on how to make a dial plan suitable for this device?

  Also my voicemail and other features are accessible via one * command.  However, these are all saved to the device, how do I fix?

  Is it possible that a user might choose to use the second record sip on the first line?  It would be very convenient.

  For direct IP numbering, you can also use a url?  One of these can be used with a phone shortcut?   He put the phone number of someone in and it will dial the ip address stored in the address book.

  Thank you!

  The ATA Administration Guide is here.

  Also my voicemail and other features are accessible via one * command.  However, these are all saved to the device, how do I fix?

  Disable the * xx order on the PAP2T and activate the * xx command in the numbering plan?

  Is it possible that a user might choose to use the second record sip on the first line?

  NO.

  For direct IP numbering, you can also use a url?  One of these can be used with a phone shortcut?   He put the phone number of someone in and it will dial the ip address stored in the address book.

  The PAP2T has speed dial a number on the user tab.  Is that what you mean?  If you have something in speed dial 4 you dial with 4 #.

• Fixed IP address assignment to a PAP2T

  PAP2T hung in the WRT54G. When I put the config PAP2T DHCP = Yes, it works fine, but the router assigns a dynamic IP address. When I put the PAP2T for DHCP = No and provide a fixed IP, subnet mask and gateway, it does not connect to the host of VOiP. What should I do to make this work?

  Reason: My VOiP provider said I should use the port forwarding on the router for some audio problems. The other day, the router got turned off and now VOiP is assigned to a different IP address. I need to correct the address to avoid such problems in the future.

  When you assign a fixed ip address, you must also define the addresses of the DNS servers.

  Some routers will also allow you to specify a specific (fixed) local network address mac - id-based.  This is another way to solve the problem.

• No voice with incoming calls PAP2T

  I bought a LinkSys PAP2T unlocked last year that worked beautifully until a few days ago. Outgoing calls work fine, but I now have problems with incoming calls only.

  When I receive an incoming call, the phone rings, but when I pick it up, I can't hear anything and cannot the caller on the other end. After a few seconds, my phone disconnects and begins to beep as if the line is busy. This occurs regardless of who calls me and whther they use a VoIP device or a normal PSTN phone. I have not changed the settings on the router (a Belkin) or the PAP2T, he simply stopped working a few days ago! I have a connection 10MB Virgin Media broadband service for my VoIP provider is Draytel that have verified the account and settings and says it works very well on their test kit.

  Since then, I tried just using line 2 and the deactivation of line 1 to do a hard reset of the PAP2T and updated the firmware to the latest version. Even tried to change the analog phone connected, but I still have the problem.

  Does anyone have any ideas as to why it suddenly stopped working with anything has been changed and what I might be able to do to fix? The PAP2T is still the title of the 12 month warranty period then maybe she developed a fault?

  Thank you very much in advance for all those who can help you.

  Cheers, Vic

  For me it looks like the problem of RTP streams - probably, it does not pass the firewall / nat on the path between SIP provider and you.

  This may have is that your Belkin router/NAT, or on internet ISP router/NAT enabled SIP ALG (Application Layer Gateway).

  If this is the case, you will need to disable SIP ALG on your router or ask your ISP internet to disable on the firewall router.

  Another question may have been if you (accidentally) changed the port forwarding on your Belkin router, or the PAP2T local LAN IP address changed and port forwarding is no longer works.

  So my proposals to check:

  -check and DISABLE SIP ALG on your Belkin (if the router has this feature)

  -check the port forwarding on your router Belkin... IF you set up port forwarding for 5060/61 you MUST set the same for RTP ports (16384-16483)

  -If you don't have port forwarding on your Belkin, try before installation of 5060-5061 AND 16384-16483 *.

  * (If this is too many ports, set the RTP on PAP2T portrange to 16384-16394 for example and then forward only go)

  -If neither of above of aid, change your SIP Line1 of 5060 for port we will tell 6070 and RTP ports to the beach we'll tell 17300-17310

  -Finally, you can do it on your side is to remove all the port forwarding and put the PAP2T to your router Belkin DMZ settings.

• Voice, outgoing calls PAP2T problem

  Hi all

  I have a pap2t voip adapter and a lot of problems.

  Here's the situation.

  In our office, we have a linux box that acts as a router, iptables configured by shorewall. Connected to the linux box is the lan internet.

  After a difficult time, I finally got to do the adapter to connect and register with our sip (sipgate.de) provider.

  But, it's almost everything.

  What works, is that I can call from extern and get a dial tone on the phone (connected to the adapter). But that's all.

  When I try to make a call, I get the following error in the system log:

  [0:5060]-> 217.10.68.147:5060
  Sep 24 12:30:24 192.168.15.9 [0:5060]-> 217.10.68.147:5060
  Sep 24 12:30:24 192.168.15.9 sip:[email protected] GUEST SIP / 2.0 ^ M Via: SIP/2.0/UDP 77.20.237.113:5060; branch = fc32b89a z9hG4bK ^ M from: Richter ; tag = 677844a993140e19o0 ^ M at: ^ M Call-ID: [email protected] ^ M CSeq: 101 INVITE ^ M Max-Forwards: 70 ^ M Contact: Richter ^ M expires: 240 ^ M User-Agent: Linksys/PAP2T-5.1.1 (LS) ^ M Content-Length: 444 ^ M allow: ACK, BYE, CANCEL, INFO, GUEST, NOTIFY, OPTIONS, SEE ^ supported : x sipura, replace ^ M Content-Type: application/sdp ^ M ^ M v = 0 ^ o =-196655 196655 IN IP4 77.20.237.113 M ^ s M =-^ M c = IN IP4 77.20.237.113 ^ M t = 0 0 ^ M m = audio 16474 RTP/AVP 0 2 4 8 18 96 97 98 100 101 ^ a = rtpmap:0 M PCMU/8000 ^ M a rtpmap:2 G726-32/8000 = ^ M a = rtpmap:4 G723/8000 ^ M a rtpmap:8 PCMA/8000 = ^ M a rtpmap:18 G729a/8000 = ^ M a rtpmap:96 G726-40/8000 = ^ M a = rtpmap:97 G726-24/8000 ^ M a rtpmap:98 G726-16/8000 = ^ M a = rtpmap:100 NSE/8000 ^ M a = fmtp:100 192-193 ^ M a rtpmap:101 telephone-event/8000 = ^ M a = fmtp:101 0-15 ^ M a ptime:20 = ^ M a = sendrecv ^ M

  [0:5060]-> 217.10.68.147:5060
  Sep 24 12:30:44 192.168.15.9 [0:5060]-> 217.10.68.147:5060
  Sep 24 12:30:44 192.168.15.9 NOTIFY sip: 217.10.68.147 SIP / 2.0 ^ M Via: SIP/2.0/UDP 77.20.237.113:5060; branch = a39224c8 z9hG4bK ^ M from: Richter ; tag = 123c259d53ab82c7o0 ^ M at: ^ M Call-ID: [email protected] ^ M CSeq: 132 NOTIFY ^ M Max-Forwards: 70 ^ M event: KeepAlive ^ M User-Agent: Linksys/PAP2T-5.1.1 (LS) ^ M Content-Length: 0 ^ M ^ M

  Sep 24 12:30:52 192.168.15.9 [0:0] call of Rel AUD
  Sep 24 12:30:52 CC 192.168.15.9: failure w / call
  Sep 24 12:30:52 192.168.15.9 Sess terminated

  When I look into the adapter information page, I see the message that the call is not valid.

  What I find suspicious, it's that I don't send or receive packets rtp or bytes depending on the info page while I get and send a lot of sip packets and bytes.

  I tried transfer ports 16000-17000 la carte (I gave him a fixed ip address), but this custom work too.

  I googled a lot and I tried many things, which lead me to this situation, and here I stand without knowing where to go.

  Greetings and thanks in advance

  Sven

  I solved the problem.

  What I had to do was to unload the modules nf_nat_sip, nf_conntrack_sip and add them to the DONT_LOAD line in shorewall.conf.

  She's.

• Can I disable the ability to make calls in PAP2T?

  Hi all

  Could not fdind and answer online here so go.

  I have connected my VoIP (Messagenet using the PAP2T) line both my that traditional Verizon on the same Panasonic Dect phone. It works flawlessy to receive calls, but when I call dials of two devices (the PAP2T and Verizon's network fixed of course).

  I need my only VoIP number to receive calls, it is there a way to disable the PAP2T to calls (while still their reception)? This way I could use Verizon to post fixed calls US and still receive calls from the two numbers without any problem when the component and the purchase of a new device.

  Thank you

  As far as I know, the PAP2 does not provide an option to disable Inbound calls since both lines 1 and 2, but you can set up to block calls and activate DND.

  You will need to have spend you your RJ11 cord at the back of the wall and the PAP2 to make outgoing calls and receive incoming calls.

• Delayed PAP2T numbering

  My configuration:

  SIP/VOIP software: 3CX V7

  Server: Windows 2008 server + IIS7

  Phones: Linksys SPA941, Gigaset 301D

  Gateway: Linksys SPA-3102 & Fritzbox 7170

  ATA for Bell: Linksys PAP2T

  Gigabit network

  I installed a PAP2T on my system of Porter Street, which is in fact an analogue phone, dialing a number when you press the button. Porter Street itself is kind of slow before he began to compose, but since I was the PAP2T to connect it to my 3CX box it takes way too long. The factor is 3 blocks away when it starts ringing in the House

  When the doorbell rings the number of SIP ring group, it takes about 10 seconds or more before it starts to sound in the House (all SIP). I tried with a normal pone on the anti-terrorism Act, and I get the same deadline. Calling the ring of a system group SIp takes a fraction of a second.

  The delay is obviously in the PAP2T.

  PAP2T has the latest firmware. Y at - it a setting or a way to do this, go faster?

  Secondly, when I ended the call and hang up, the intercom sounds for a while, which seems a normal sound when the other side closes the connection, but after about 10 seconds, it turns into one his beeper much more strong and alarming.

  Who continues to go on for 15 seconds before coming back on 'hung up' State.

  What is the second "alarm" and how to adjust the ATA to disconnect quicke/better?

  Suggestions welcome!

  Bastiaan

  PS just see a strange element in the PAP2T information screen: "call 1 State: invalid ' (see below) what happens after the disconnection of the SIP system.

  Full name: Porter Street

  Hook status: Off

  Last entry to the: 02/02/2003 17:14:37

  Waiting message: No.

  Last number called: 802

  SIP port is mapped:

  Call 1 State: invalid

  2 State call: Dial idle 1

  Tone: SIT 1

  Hello

  I think I can help you with the delay of the original composition.

  Instead of dialing 802, programm "your doorbell" to dial # 802 (Yes, eight-zero-two-hash) sequence.

  or change the PAP2T dial plan to include the sequence like 8xxS0 or 80xS0... to make sure that any

  starting with 8 or 80 number will be dialed immediately after the 3rd digit.

  I don't know what to do with the "alarming sound", but I think that it is generated by PAP2T himself.

  And I see that TONE is in State "Tone: SIT 1 '... I have never see that...

  Sounds to me like your Mermaid... perhaps the SIP protocol was not followed correctly?

  Check the settings of the PAP2T, if there is any possibility of 'program' to SIT 1 tone and reprogram.

  In my SPA-1001, I under the REGIONAL section to SIT 1 to 4 If SIT while SIT 1 reads like this:

  985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)

  -Reprogramming or deletion - my change the sound of the siren.

  The reason why the siren sounds is command probably something invalid received SIP

• How to program speed dial to call a userid (PAP2T)?

  I have a justvoip, no phone number account, only userid. I programmed successfully in my PAP2T and I connect, receive calls, etc perfectly. The other side is a person who uses a computer and a justvoip software.

  I have a friend (login "myfriend") in the same situation (no phone number, only the userid). Is there a way to program my PAP2T to call "myfriend" when I compose 3, for example? I studied call them Fwd, but without success.

  Thanks in advance.

  Great! It worked perfectly!

  Thank you!