SPA3102 PSTN to voip

Similar Questions

• Set the SPA3102 PSTN

  
  
  

• Transfer to PSTN calls on the SPA3102 VOIP does not work with voip providers who do not proxy outbond

  I use the SPA3102 to transmit calls from PSTN to Voip network. Work very well with a local Voip provider calls Vono (www.vono.net.br), I think that this provider is the one who has the outgoing proxy. However, I would like to use the SPA3102 forwardin PSTN calls with other providers such as Freevoipdeal, VoipWise, VoipRaider etc... These suppliers have no outbound proxy. How can I get around this? According to the link below, you need to plan an outbound proxy. What should do?

  http://www.Cisco.com/en/us/products/ps10024/products_qanda_item09186a0080a359ae.shtml#q

  For example, freevoipdeal voip provider guides that leave the field blank outgoing proxy as below.

  https://www.freevoipdeal.com/SIP

  You can leave the outgoing Proxy field blank.  You do not have to have an entry in this field for the SPA3012 works correctly.

• The SPA3102 to PAP2T PSTN line

  Hello, I need to use the RTC of another building connected to the first by a link radio. I have an SPA3102 connected to the RTC and PAP2T line in the other building. the configuration I did is the following:

  On the line PSTN SPA3102 tab:

  PSTN-to-VoIP Gateway Enable: Yes

  PSTN Caller Auth method: no

  PSTN through line 1 ring: no

  PSTN to VoIP CID CID: Yes

  PSTN Caller default DP: 2

  Off hook while VoIP call: no

  Dial Plan 2: (S0<>[email protected]/ * / _address:port >)

  where userid is any user name you configured in line PAP2T configuration, ip_address is the PAP2T ip address, port is the Sip Port configured on line PAP2T configuration.

  Call without Reg: Yes

  PSTN response time: 3

  Under the tab line SPA3102:

  Turn on numbering IP: Yes

  Configuring the tab line PAP2T :

  Call for years without Reg: Yes

  Problem is I get the PAP2T call but I'm not able to do. Can U help me understand my mistake?

  Thank you

  One more thing... I have connected the spa3102 on my local network through an "ethernet" port and there is nothing plugged into the port of 'internet '. Is this a problem?

  Yes, I'd say it's generally a problem.  My opinion is that the SPA3102 was not designed/tested to make calls using the SPA3102 network premises and which could be the reason why you have audio problems.  You must be using the 'Internet' decision-making (WAN) and for the ip addresses using DHCP from the router or use a static ip address in the subdomain of your main router.

  If you change the IP addresses, you need to adjust the ip addresses used in your direct ip call configuration.   You must use ip addresses that will not change over time, if the router is restarted.  The use of router is to route the packets between the ip address used for the SPA3102 and the PAP2T.  You are generally not required to configure a path between the two addresses.

  In addition, you are running an earlier version of the SPA3102 (3.3.6) firmware. The current version of the firmware is 5.2.13 and can be download here:
  https://software.Cisco.com/download/release.html?mdfid=282414112&SOFTWAR...

  Make the above change and post a few new sip debug trace of problem calls and configurations showing the ip address changes.  The syslog posted show the title of the messages, but do not show the detail of the messages (Sip, Sip 200 OK, etc.).  Details will show the addresses by the call for audio.  Maybe a program different syslog can show detail.  Cisco has a simple windows pc program you can download here.

  https://supportforums.Cisco.com/document/36921/using-slogsrvexe-utility

  On the SPA3102 and the PAP2T System tab, you have an ip address for a syslog and a debug server.  You only need an ip address for the server to debug.

• Connection of remote offices SPA3102

  My boss just bought two SPA3102 VoIP routers in an attempt to set up a VoIP connection to a remote office, thus avoiding expensive international calls.  I'm looking for advice on setting up.

  This is the ideal configuration, we're after.  Box, it is in the seat, and it is connected via the RTC port on a phone line (in fact the PBX).  His IP address is 192.168.32.8 (we are without using the routing of each box function, then the two boxes are connected to the network via the WAN port, with the side LAN not used at all).  Any incoming call on the telephone line must be connected directly on the phone connected to the box of two.  There is no phone connected to the box one.

  Box 2 is in remote desktop. For testing purposes, its IP address is currently 192.168.32.9.  He got a telephone connected to the telephone port, and the PSTN port is connected to a standard telephone line.  We want the phone to use the RTC line by default, but if the user dials 9 then, we wish the appeal directed to the PSTN line on box a (i.e. PBX.), where it can dial into a PBX it seeks.

  The offices are connected by a virtual private network, so there is no NAT anywhere to worry.

  I have a problem in this place, what with the instruction light manual and confusing options, so any help much appreciated!

  Here are the settings I've worked. I have omitted some.

  Head office

  WAN configuration:

  Connection type: static IP address

  Static IP: 192.168.32.8

  Subnet mask: 255.255.255.0

  Enable WAN Web server: Yes

  Line 1:

  Activate the line: no

  PSTN line:

  Activate the line: Yes

  Use outbound Proxy: no

  Register: no

  Call without Reg: Yes

  Call for years without Reg: Yes

  Numbering plan 1: (P1[email protected] >)

  (causes the call to transfer to the other box after a delay of one second)

  Enable VoIP to PSTN Gateway: Yes

  PSTN to activate the VoIP gateway: Yes

  PSTN through line 1 ring: no

  PSTN response time: 0

  PSTN Ring through delay: 0

  Remote Desktop box:

  WAN configuration:

  Connection type: static IP address

  Static IP: 192.168.32.9

  Subnet mask: 255.255.255.0

  Enable WAN Web server: Yes

  Line 1:

  The line activation: Yes

  Register: no

  Call without Reg: Yes

  Appeal of ABS without Reg: Yes

  Auto RTC backup: Yes

  Dial plan: (0 x<:@gw0>|) <>xx. <:@gw0>| <7:192.168.32.8:5061>)

  (This operation transfers the call to the seat on numbering 7 9 box gets an outside line, any number beginning with 0 is also formed through the PSTN line)

  PSTN line:

  Activate the line: Yes

  Enable VoIP to PSTN Gateway: Yes

  PSTN-to-VoIP gateway enable: no

  I have a problem on the left: calls to the phone remotely (both by the PBX and telephone line remote) are very quiet.  I played with the settings of Gain PSTN SPA to compensate, but I set the remote zone to its maximum of 12 and it is still too quiet, so I think that there is a problem elsewhere.

• spa3102 from point to point

  Environment: PBX in the basement with charger of fiber on the second floor.  PBX does not support an IP conference room phone but all office phones are WHAT VOIP connected to a card in the PBX.  I need to get an analog extension to the second floor.

  How can I configure the SPA 3102, connected to a line on the PBX and ethernet to pass and the SPA 3102 on the second floor of the connected switch and telephone analog?

  If your PBX is digital and supported sip you may be able to simply use an analog terminal adapter (ATA) and install a new extension on the PBX.

  If this is not possible, it is possible to use a SPA3102 to an analog telephone line from the PBX interface and extend this line to a distant analog terminal as an another SPA3102 adapter or other terminal adapter analog voip which takes supported unregistered calls.  Using a SPA3102, is not as elegant or professional than the commercial market TC1910 that was referenced and is sold specifically for this purpose of analog telephone extension.

  "Back to back" to the SPA3102 2 configuration has been around for a while.  Search on 'SPA3102' back to back and you will find the help as the detail in this thread
  http://community.Linksys.com/T5/VoIP-adapters/connect-back-to-back-2-Spa...

  Basically, you are configuring the SPA3102 PSTN line tab to attach to the analog phone line and communicate automatically with the ata remote using numbering direct ip unregistered.  For a call from a PBX, when receiving sounds of the PBX the SPA3102 will automatically call the ATA distant.  For an outgoing call from the ATA distance, when the caller takes the phone off the hook ATA automatically dials the SPA3102, which will return a tone to the caller.

  The ata remote is configured on the tab as a voip phone line.

  Settings key on both units is registry: no, do call without Reg: Yes, and call without Reg: Yes

  Other parameters are noted in the thread referenced above.

  Call is made using the ip address: port of the map.  You configure the adapters with static ip addresses.

• SPA 3102 - call "RTC for VOIP" setting is not on CLI/CID phone line active.

  Hi guys

  Recently, I discovered if the PSTN line is having with CLI/CID then helped the PSTN to VOIP Call not established with the standard settings. But others see (VOIP to PSTN) works well OK. Basically, it works well with the PSTN without CLI/CID lines. I guess when with CLI, there different voltage levels online. ?  You would let me know what are the parameters that I need to SPA in order to work with the PSTN line that allowed the CLI/CID please? This situation is covered by Sri Lanka Telecom PSTN lines. At the moment I have IT of standard parameters such as it comes with the SPA3102.

  Your feedback and help here...

  Syslog & debugging can save so much time...

  Line with no CLI:

   INVITE sip:[email protected]/* */ SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bK-a16a89a0 From: PradeephSL [email protected]/* */>;tag=4cc44dbb58f1a944o1 To: [email protected]/* */> Remote-Party-ID: PradeephSL [email protected]/* */>;screen=yes;party=calling Call-ID: [email protected]/* */ CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="17772695735" realm="callcentric.com" nonce="684082550c6f8ce50da482371a591df7" uri="sip:[email protected]/* */" algorithm=MD5 response="01c15ae601d9d9f5af10023f908a1a4c" opaque="" Contact: PradeephSL [email protected]/* */:5061> Expires: 240 User-Agent: Linksys/SPA3102-5.2.13(GW002) Content-Length: 441 Allow: ACK BYE CANCEL INFO INVITE NOTIFY OPTIONS REFER Supported: x-sipura replaces Content-Type: application/sdp ... 

  This INVITATION is accepted by proxy.

  Line with CLI:

   INVITE sip:[email protected]/* */ SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5061;branch=z9hG4bK-ea75ea0 From: PradeephSL [email protected]/* */>;tag=2d4fafa5d94faa2co1 To: [email protected]/* */> Remote-Party-ID: PradeephSL [email protected]/* */>;screen=yes;party=calling Call-ID: [email protected]/* */ CSeq: 102 INVITE Max-Forwards: 70 Proxy-Authorization: Digest username="17772695735" realm="callcentric.com" nonce="fd5a5e637b6e56fea56886ee25aababa" uri="sip:[email protected]/* */" algorithm=MD5 response="b6b1aa7104d1be764de5c74f369ef5be" opaque="" Contact: PradeephSL [email protected]/* */:5061> Expires: 240 User-Agent: Linksys/SPA3102-5.2.13(GW002) Content-Length: 441 Allow: ACK BYE CANCEL INFO INVITE NOTIFY OPTIONS REFER Supported: x-sipura replaces Content-Type: application/sdp ...

  This INVITATION is rejected by proxy with "403 incorrect authentication."

  But Howard has already struck and explained...

• dial the SPA3102 how an RTC to my cell?

  Hi all

  I found another thread on this topic but the link was broken to a former page sepura.

  In any case, as the title says I need to transfer my PSTN calls to my cell phone after a few rings

  im guessing im using the PSTN-to-VoIP gateway Setup? is anyone know what settings should I?

  Thank you

  Tim

  I don't see any voip provider configured on the tab line PSTN, and I do not know 1234567890 is not your cell phone number.  The call is sent by the configuration on your PSTN line tab.

• SPA3102 CID has not managed to asterisk

  Hello

  I have several 3102 to FXO line with asterisk integration.

  The only thing that I was unable to make it work, even with the option:

  PSTN to VoIP CID CID: Yes

  Was to the CID go to the asterisk, if I check the info on 3102 tab, in last PSTN Caller:, I can see the last number of the caller.

  I have what would be missing?

  Thanks in advance,
  Jorge Bastos,

  It seems that "yours" RTC gives CID information between the first and second ring. Currently, PROMPTS for Asterisk is sent immediately, for example before the CID is known. It must be delayed until the CID is available.

  Go to the section "FXO Timer values".

  The "PSTN Answer Delay" value number (in seconds) which is a little more than sound period on your PSTN line. Period of ring is about 5 second in most countries. so, I recommend that you use 7 seconds here in this case.

• help with SPA 3102 (question graphcal)

  HI guys here is my situation (I draw so it would be easier for future reference):

  

  I want to pick up A phone and dial the Ext 101 101, 102 for Ext 102 and so on.
  also, I want to put routed my call for location 2 location 1 for example transferring calls to 104.

  any help would be really appreciated even a starting point as for example the configuration of spa to route my call not NAT, so it can connect to the other.

  castro69 wrote:
  also, I want to put routed my call for location 2 location 1 for example transferring calls to 104.
  any help would be really appreciated even a starting point as for example the configuration of spa to route my call not NAT, so it can connect to the other.

  I guess that's an analog PBX, otherwise you wouldn't need the SPA3102 through the internet.

  For communications between the SPA3102, I would use direct ip call, using the external ip address and the sip port numbers.  Think that the SPA3102 is two separate cards inside the box where treat you everyone with its sip port number added to the external ip address common.

  I would setup port sip distinctive numbers on each of the baths to keep things straight.  You have a number of separate port for tabs of line 1 and line PSTN.  You will need to send these to the SPA3102 adapter port numbers in their respective routers or firewall of the router will reject incoming packets on the internet.  I would also convey the port range rtp for voice, flow packs.
  On each tab of the line 1 and RTC, you define NAT Mapping Enable: YES, not record, make call without Reg Yes, years call without Reg No. I put your external ip address on the Sip tab under EXT IP.  This will tell the SPA3102 to use this address in the sip signaling. I assume you are using static external ip addresses.   On each tab of the line 1 you would activate IP Dial Yes.

  The analog PBX is connected to the FXO port on one of the Spa.  You should check the voltage level hung up and won and then set the line parameter usage on the RTC of the SPA line tab to halfway between the two readings.  You can read the levels of tension on the PSTN line tab.  Calls to the PBX of the PSTN line tab will go through the voip to PSTN gateway.  I set up the catwalk with http authentication and configure a user name and password.

  Details are starting to become quite complicated.  I'd get running through steps.  Get a job step before moving on to the next step.

  The 1st step would be to get A phone call/receive calls to a PBX.  You can configure the line 1 for FXS phone attached A to use port location PSTN 2 as the proxy using http authentication, and you can then dial the extensions you want to call.  Location 1 SPA3102 will send a guest of the sip Protocol to the tab location 2 SPA3102 from pstn line and the SPA3102 will dial the number on the FXO port to the PBX.

  For calls coming from the other direction of a PBX to slot 2 SPA3102 the only place where you can connect a voip call is in the SPA3102 numbering plan.  If you want to call only phone that is easy, install you just dialers-messengers automatic telephone in the pstn-to-voip dial plan.

  I'm not clear about what you want to do with phone B I take is Extension 104.

  I like your designs.  Can save a lot of words.

• PAP2T vs SPA3000

  I tried to find my solution on this forum but I have not found a clear solution I'll ask the question in a new thread. (if reapeted, I'm sorry)

  I want to connect two different VoIP providers to one analog phone. The reason is that I have been offered pay less for calls but no DIDs and others - free DIDs in places, I'm interested. Obviously, the second provider will be used for incoming calls only.

  Now, I know PAP2T supports two VoIP providers, but the thing I couldn't understand is: can route incoming calls from these two suppliers in an analog phone (connected to one of the two RJ11s) and route outgoing calls via a provider selected (in my case supplier no 1) OR is it a 1 to 1 relationship provider of Voip 1 is related to the analog phone not 1 for incoming and outgoing calls and provider of VoIP 2 is related to the analog phone no 2 for ONLY incoming and outgoing calls.

  I found some information that the gateway SPA3102 supports two VoIP on an analog phone and it also supports call from a regular landline so it would probably cover my needs, but I don't want to get a more advanced camera if that more simple it covers too much more.

  Please notify.

  In regards, Andy

  The PAP2T provides a 1:1 relationship as you describe.  A line goes to one phone, the other line goes to another phone (unless you have an analog handset 2 lines).

  With the SPA3102, you can configure the provider that gives you the incoming calls to register for line 1.  You can configure the only provider coming out as a "gateway".  The only incumbent provider must allow you to make calls without registration and must not require a proxy out.  Configure you your dial plan to send outgoing calls to the provider of the gateways.  In this way, you can use one attached handset to access all calls.  Some providers require registration before they allow you to make a call and so will not work in this senario.

• SPA232D problems integrating the POTS via FXO

  I just bought a SPA232D and I have problems with the initial configuration with the POTS via the FXO line.  I'm back to the original factory configuration for partition the problem.  If I connect a phone to the fixed telephone line work and turn the SP232D off, all phone funtion works, inbound and outbound more voicemail from the central server by the PSTN.

  If I power up the SPA232D with the line FXO connected to the POTS RJ11 and FXS line phone, I can make outgoing calls through the PSTN.  However the incoming calls to the PSTN not sound and the service of voicemail never folds on the Central POTS but disconnects.  Sometimes I see the hook out in the status page.

  In the State of the engine, if I register a DECT handset and set it to the PSTN and do to the PSTN by default, I make a call through the PSTN, it sounds and I get a successful connection.  I can also make outgoing calls through the PSTN.  However, if I do not answer the call voicemail service never folds on the Central POTS but disconnects.

  Clearly, I have good location of the SPA232D with the POTS (US, Verizon (formerly Frontier), considered an old switch of POTS, not a great line, Upstate New York).  But since it works when it is off, the basics seem to be there.

  Does anyone have advice.  I can't find any close correspondence on the resources of the web.

  Thank you

  Dan Davis

  The notice just found elsewhere - it may or may not help:

  1. You can set this to 'no' (No. When disabled, PSTN calls will not be auto-répondu by the SPA.
  2. "The SPA-3000 is two have independent in a box." For 'line 1 to @gw0' and ' line PSTN Ring Thru for line 1 "feature, the SPA-3000 places in fact call VoIP on one side of the box to the other."

  In addition, read page 210 of the Cisco SPA232D enhanced mobility phone adapter Administration Guide

  It describes the process of appeal RTC / 1 a little curve.

  From the pages 209-210, I wish that:

  1. ring of appeal incomming thru online FXS FXO
  2. If not caught in time, it is autoanswered by ATA. He began to play the tone. The caller must enter target number via DTMF
  3. target number is handled by the default dial plan

  If no target number not entered until volume dial timeouted, the call is completed.

  It seems to comply with your observation.

  Then try to PSTN-to-VoIP Gateway allow no value and try again.

• SPA3102 problem: State PSTN: still rings in the State

  We got a new router Linksys SPA3102 VOIP gateway, and we have some problems to operate.

  The SPA3102 is properly connected to the local network, but it seems to have problems with the incoming PSTN line.

  However, it does not sound when I call the number RTC. On the other hand, State PSTN always says 'Ringing '...

  When I restart the SPA3102, then the extension on the pbx sipx she must call rings (without anyone calling the PSTN number).

  The status page says something like:

  Voltage: 14 (V) loop current: 0.0 registration of (my) State: not recorded the last record to: next record in: last call VoIP number: [email protected] last call number RTC: last calling VoIP: last PSTN Caller:, last RTPC Disconnect reason: CPC Signal RTC activity Timer: 30000 (ms) mapped Port SIP: 5061 Type of call: VoIP Gateway call VoIP State: Idle PSTN State : ringer VoIP: tone RTC:

  (Note: the bridge didn't need to register for the sipx-pbx).

  While it fires after a reboot of the SPA3102, nothing happens when I dial my mobile PSTN - I hear only ringing on mobile (it says busy or more).

  Does anyone have advice?

  Andreas

  It did not break. The problem was with the cable to connect to the UK British Telecom. I haven't used the gray wire that came with it, but used a telephone cord UK I had to hang out - bad move.

  When we talk about Linuxemporium, there where I bought the SPA3102, I was made aware that the cable might be the problem. So I went on the road in a small phone shop to get a US - UK adapter (packaged with the SPA3102 is still an American cable), that didn't fix it. I had another well-known adapter from a retailer of Electronics UK, and that one has solved the problem. The SPA3102 fortunately now works as my PSTN gateway.

  I hope this can help others with similar problems.

• SPA3102 - disable incoming PSTN

  I have an SPA3102 I am putting in place so that

  -ATA outgoing calls through voip (work)

  -incoming calls from the PSTN are ignored

  -outgoing emergency calls 911, etc. are out by RTC (work)

  -outgoing calls if VOIP connection downwards through the PSTN. (work)

  I can't seem to find a way to have it ignore incoming PSTN calls.  I have this configuration in a suite of grandmother, as she makes calls sparingly, but we wanted to the PSTN where the voip connection is broken.  Now that I have everything set up, every time there is an incoming call on RTC, it sounds the phone on the ata...

  Is there a way to disable this?

  Thank you

  John

  JRoesner wrote:

  -incoming calls from the PSTN are ignored

  I can't seem to find a way to have it ignore incoming PSTN calls.

  Is there a way to disable this?

  On the tab game ring Thru line 1 PSTN PSTN line: No.

• SPA3000 dial of Voip to PSTN problem

  
  

  Do it with the dial plan.  Set the PSTN Caller default DP a number available and configure the dial plan:

  (S0<:[email protected]>) where [email protected] is the sip uri when you want to transfer the call.

  If it's a PSTN number you want to transfer the call to you must configure tab PSTN line with the credentials of voip to voip account, you want to use to transfer the call.

  Of course, the line 1 tab you set enable component IP: Yes, if you send to a sip uri.