TRUNK VOIP CREDITS SPA 3102

Similar Questions

• SPA-3102: unable to connect to the SIP server

  Hello

  We have a SPA-3102 installed and work properly in our US Office. We are trying to install another one in our office in India, but we cannot get this device to register with the SIP server.

  This device works perfectly when try us two different residences here in India with two different providers. But when we try to our office with a third party service provider, it is impossible to save.

  The internet connection at the office from the East through an ADSL router which is mode bridged with two IP static coming out of it. We are able to have access to the internet for our PC through this SPA-3102, but the SPA itself is unable to register with our SIP server. We use a Gizmo5 SIP server.

  The same parameters were working in our offices in the United States and two different residences in India but does not work in our office. I don't think that the ISP here has nothing to do with this problem. Would be - this works the ADSL router in bridge mode? Can I make changes in the settings of the SPA to work around this problem?

  Any help is appreciated.

  The problem could be the firewall or the router, but it is also possible that your Indian ISP is blocking voip connections.  It was reported that is not unknown in India.  Sometimes this type of blocking is done by throwing packages for voip standard sip signalling port 5060. You can easily change the SIP port in the SPA3102 to traffic returning to the SPA packages and in fact routers often changes the port without your knowledge, but packets destined for voip provider must use the port specified by the provider number.  Since you can access the internet with your pc, I would try to run tests to see if this is the case.

  I ran a few ping tests and tests to see if I could save with other providers or services that use replacing or other sip signaling ports.

• SPA-3102-backup/restore settings?

  Hello

  Is there a way to backup/restore the settings to a file in the SPA-3102 router?

  ASE

  There are several ways to save the settings of the SPA3102 according to some forums - check links ff

  http://forums.whirlpool.NET.au/Forum-replies-archive.cfm/756454.html

  http://Forum.Voxilla.com/Linksys-Sipura-VoIP-support-forum/SPA3102-compiler-configuration-backup-Mon...

• Failure to register SPA-3102 on "DSL-2750E" D-Link router

  Hi team

  I guess I'm able to clarify this here since its associated VOIP Cisco device. This SPA-3102 works fine with my old router. I wanted to have a WiFi router better and I installed D-Link router to "DSL - 2750th". Internet, everything works normally through this router but SPA3102 is does NOT record. I have a debug trace attached SIP & don't know why it's a failure of registration. Not any other filtering or firewall configured in the router and it works in accordance with all the default settings. Would you be able to give advice on this please?

  Debugging is nice, but it discloses the contents of INTERNATIONAL packages only. No responses. Intercept (incoming and outgoing) SIP packets.

  I have the average time - do not use the names in the proxy configuration. Use the IP address.

• SPA 3102 Admin Guide

  Where can I download the administrator for the Linksys SPA 3102's Guide?

  I don't remember how many times I tried to find such a link and how many hours I have spent so far with it. I could find answers to this question, but the links are no longer valid.

  Thank you for your help.

  Try this link.

• 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...

• SPA-3102 PSTN idle status, will not answer but watch sounds.

  Greetings,

  I am a complete newbie in the world of voip. I have an spa3102 and a computer virtual trixbox. When I use my softphone to compose, it makes to the spa3102 (shows the id of the last and gives a status of the appellant voip 'reply') but I get an "all circuits are busy now" and does not change the status of IDLE pstn. It also will not answer the phone for incoming calls to the PSTN.

  Outgoing stuff could be a problem of numbering plan (don't know not just guessing) here it is:

  (S0:xx. <@gw0>)

  I tried the regular (or default) (xx).

  incoming calls are supposed to go to a handset on my desk, for example ext 101 is the dial plan: (S0:101@internalip)

  I can't see the forest for the trees. Can someone give me some advice on where to look. I'm completely puzzled by something non-response externally, and why I get a all circuits are busy if there are inactive on the status page.

  an interesting note: If line1 is connected to an analog phone, it will dial a number, but always shows failover for a reason any...

  Thank you very much!!!

  Greetings!

  Thanks for the reply. I do not know if I explained properly (sounds like I don't have). I have a trixbox and I use the SPA in interface with my RTC and I want to have a fax machine (but later) on the FXS port. The problem I have is that I could not field calls to the collection. They would just ring and ring. I thought it was a dial plan (and hope it was not a hardware problem), but he just wouldn't answer the phone. He would show 'Ringing' like the PSTN status, but Voip would be "IDLE". I've been pouring through the guide of the administrator, forum messages for about 2 weeks now.

  But I didn't know what the problem was. The spa3102 is waiting until she can establish a connection (with the internal phone system) until it meets the PSTN. This is why it continues to ring. The problem why it shows it is inactive and (even with enabled full logging to a syslog box) no traffic to SIP the trixbox internal... because I had the port internal plugged in... not the WAN port (or internet). Thats right... silly me...

  I gave the wan port an address on the same subnet and moved the cable to it, once I could talk to her I change address internal to something even not on my network (if no routing problem) and left the network adapter disconnected (internal nic). Then I changed the wan to the old internal address address (trix trunk was already configured in this ip address anyway...) and PRESTO! I tried calling the number RTC and started to receive log entries showing the sip signaling and he's trying to route the call to the box of trix. It works now! So he was not answering the phone in my case, because it could not connect to the pbx or the ip phone and that because I don't use the WAN/INTERNET port on my SPA. Something I don't remember seeing in the textbooks as a "hafto' or a 'must '... I might have forgotten that of course, but I don't remember seeing him there.

  So if you enable logging, and then call this number (I used the PuTTY and 'tail-f /var/log/spa3102.log"to see the entries that they light up the box of syslog) and you don't see any sip traffic, it could be that you must use the WAN/INTERNET port instead of the internal port apparently logical... BE sure THAT ENABLE YOU REMOTE ADMIN if you decide to give it a shot.

  Also you can do the spa to answer the phone, even if it has no where to send the call (or if you want incoming people to use a PIN) on the GATEWAY PSTN - VOIP, there is a setting "off hook during the VOIP call" (default = no), but change that Yes - he would answer... and then sit there... If I had a PIN with a dial plan that is associated with this would give a fast busy after the entry of the PIN. BUT now everything is cool!

  Thank you!

• 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.

• ID (CLI of the incoming caller SPA-3102) truncates the last digit when the telephone number is longer.

  Hi team

  Depending on the subject, when the number of the caller is Longer (i.e. International call with Country code etc.) as 11 digits, it truncates the last digit of the incoming phone number. See picture attached. Full number is 60126140235, but he loses the last digit is 5. Any suggestions?

  Also, there are comma (,) before the number that consumes valuable space. How elimiante that?

   FXO:Start CNDD fxo cnddwrap_feed parse ok 0060126140 status=2 -- Caller ID: -- Name = (null) -- Remote Number = 0060126140 -- Dialable Number = (null) -- No Number Reason = (null) -- No Name Reason = (null) -- Message Waiting = (null) -- Date and Time = 07/22 18:06 FXO:CNDD name=, number=0060126140 FXO:Stop CNDD FXO:CNDD Name= Phone=0060126140

  According to the newspaper that you have provided, the number of callers number claimed 0060126140.

  Length maximum support of the number in the E.164 format is 12 digits, including the country code. SPA don't know any used number format, so it has no reason to truncate 10-digit - I guess that he supported at LEAST 12 E.164 numbers.

  As the number seems to be broken on the side operators (00 superfluous as prefix, truncated to 10 digits) there's nothing you can to with him on the side SPA. We cannot guess figures sent by the operator to you...

  I can explain what is happening even on the side of the operators (although I'm only guessing) - I guess that we are talking of two digit country code country, so 10 digits is the maximum length of the national number. It seems your operator to consider the number national number and truncate them to 10 numbers on their side.

• SPA-3102: how to fill the calls from SIP to PSTN and vice versa?

  Hello, since my ISP to my office blocked SIP ports, of I'll try and use it at home where the ATA works very well.

  If I call you ATA installed in my office in the United States (where there is no problem) to my ATA at home in India, I have how to route the call selectively to theFXS port or port FXO of the anti-terrorism Act in India? I I want to answer the call directly using the phone connected to the ATA instrument or make another local call out on the PSTN connection.

  TIA

  You get two accounts of Gizmo.  You put an account on the tab line 1, you put the other account on the PSTN line tab.  If you want to ring the phone attached to the SPA3102, you dial the account tab line 1.  If you want to get the tone to fill an outgoing line PSTN call, you call the account registered on the PSTN line tab.

• SPA 3201 - problem with incoming calls

  Hello

  I installed a new SPA 3102 connected to a mini Server asterisk; catches of phone line to connect to the existing line of the Earth and the phone.
  The unit has current firmware, and the one and only ethernet cable taken in the WAN port - these two steps make a working direction. I can dial a number on the phone, the call goes through the asterisk and goes out to the PSTN.
  The other way, however, does not work: I get syslog entries that the call is detected, but the device doesn't send what anyone on the server (as checked with wireshark)
  The dial plan is
  (S0<:1234>)
  but I also tried
  (S0<>[email protected] / * />)
  and some variations more

  This is syslog entries:

  FXO:start CNDD
  Number of the caller analysis = callingnumber
  -Caller ID:
  -Name = (null)
  -Number distance = callingnumber
  -Dialable number = (null)
  -No reason number = (null)
  FXO:CNDD = name, number is callingnumber
  FXO:stop CNDD
  Phone = FXO:CNDD name = callingnumber
  Your RTC AUD:Stop
  FXO: on the hook
  Your RTC AUD:Stop
  the next sequence is repeated several times, probably as long as the line is actually ringing
  FXO:start CNDD
  Your RTC AUD:Stop
  FXO: on the hook
  Your RTC AUD:Stop
  FXO:stop CNDD

  Where would I look next?

  I found a tip in an ongoing discussion, and in fact this has solved the problem: "PSTN ring timeout" must be longer than the time to ring + break from the ring, and "Time of response to PSTN" must be short enough

  Unfortunately, if incomplete description of configuration. There are so many dial plans to set, but it has not specified that have configured it.

  Please read before RTC call the SPA3102 to VOIP. I hope this will help you.

• Original SIPURA SPA 3000 help

  Five years ago, that I bought, directly from SIPURA, a new SPA-3000. I've never used but now I want to set up, but I want to make sure it has the latest firmware. I looked at the Linksys site (because any attempt to access www.sipura.com is redirected to Linksys) and found the firmware for the SPA-3000. Can someone tell me if this is the same unit as mine or changed from the original and the firmware does not work on mine?

  I tried to contact Linksys technical support, but their automated system tells me that I need to contact the dealer, who ironically is Linksys, since they bought Sipura. When I get to a real person that they just repeat that I have to contact the dealer of origin, they won't listen to me telling them that they are now the dealer.
  Can someone help and tell me what is the latest version of the software for this device and where I can download it? The configuration GUI displays the following information:
  Software version: 2.0.13 (GWg)
  Hardware Version: 2.0.1 (7067)

  Thank you

  Hello bsdaiwa, Yes this is the same unit. SPA-3000 has never really produced by Linksys, they later produced SPA-3102 units. Anyway, regardless of the producer of the unit (SPA or Linksys), they are able to use the same firmware.

  I myself have 2 pieces of SPA-1001 units, the 'old' box of Sipura, Linksys 'new' box, each of them having different revision HW and the two hapily run the same firmware 3.1.19 (SE) Linksys pages.

  Also, the SPA-xxxx units are protected from the flash of the incompatible firmware (for example from a different camera model), unit load the firmware first to the RAM, then check the "signature" FW, and that if it is the corresponding FW "signature" then unit flashes it it's rom flash and reboot.

• SPA3102 will not dial a number, or forward calls to Elastix

  For the past 2 days, I am trying to get this SPA 3102 to work as a PSTN gateway but no luck... My network is 172.16.1.0/24. My default gateway is 172.16.1.1 and local IP address of the SPA is 172.16.1.200. Elastix server is 172.16.1.8. I don't have any other VOIP providers so I na not configure my WAN for SPA settings (Nothing connected to the WAN port)

  Here are the screenshots and my config settings:

  The SIP Trunk settings:

  Outbound caller ID: 383579 (my local PSTN number)

  Chanels maximum: 1

  Name of the Trunk: pstn

  Parameters of peers:

  do not allow = all
  allow = ulaw
  Polycom = no
  context = of-RTC
  dtmfmode = rfc2833
  Host = 172.16.1.200
  incominglimit = 1
  NAT = never
  port = 5061
  qualify = yes
  secret = 300
  type = friend
  username = pstn
  Entrants settins empy rest under vacuum

  Road of basketing:

  DID number: 383579

  Transfer the call to ext: 205

  rest is default or is empty

  Line1 SPA is disabled because I want just difficulty dailing and receive phone calls.

  Here are a few screenshots of RTC in SPA line configuration:

  

  

  

  

  As see you in picture above I have sound but spa isn't answer...

  It comes from the phew journal asterisk call out:

  -[358291@from-internal:1] Macro execution ("SIP/205-087af3d8","user-callerid |") SKIPTTL | ") in new stack
  -Execution [s@macro-user-callerid:1] Set ("SIP/205-087af3d8", "AMPUSER = 205") in new stack
  -Execution [s@macro-user-callerid:2] GotoIf ("SIP/205-087af3d8","0?") report") in new stack
  -[S@macro-user-callerid:3] ExecIf execution ("SIP/205-087af3d8","1" | ") The value | REALCALLERIDNUM = 205 ") in new stack
  -Execution [s@macro-user-callerid:4] Set ("SIP/205-087af3d8", "AMPUSER = 205") in new stack
  -Execution [s@macro-user-callerid:5] Set ("SIP/205-087af3d8", "AMPUSERCIDNAME = Damir here") in new stack
  -Execution [s@macro-user-callerid:6] GotoIf ("SIP/205-087af3d8","0?") report") in new stack
  -Execution [s@macro-user-callerid:7] Set ("SIP/205-087af3d8", "AMPUSERCID = 205") in new stack
  -Execution [s@macro-user-callerid:8] Set ("SIP/205-087af3d8", "CALLERID (all) ="Damir Reic ' <205>' ") in new stack
  -Execution [s@macro-user-callerid:9] Set ("SIP/205-087af3d8", "REALCALLERIDNUM = 205") in new stack
  -[S@macro-user-callerid:10] ExecIf execution ("SIP/205-087af3d8","0 |") The value | Channel (Language) = ") in new stack"
  -Execution [s@macro-user-callerid:11] GotoIf ("SIP/205-087af3d8","1?") continue ") in new stack"
  -Goto (macro-utilisateur-callerid, s, 20)
  -Execution [s@macro-user-callerid:20] NoOp ("SIP/205-087af3d8","Using CallerID"Damir Reic" <205>" "") in new stack
  -Execution [358291@from-internal:2] Set ("SIP/205-087af3d8", "_NODEST =") in new stack
  -[358291@from-internal:3] Macro execution ("SIP/205-087af3d8","record-enable |") 205. OFF | ") in new stack
  -Execution [s@macro-record-enable:1] GotoIf ("SIP/205-087af3d8","1?") check ") in new stack"
  -Goto (macro-record-enable, s, 4)
  -AGI [s@macro-record-enable:4] performance ("SIP/205-087af3d8","recordingcheck |") 20090827 205946 | 1251399586.64 ") in new stack"
  -Launch Script AGI/var/lib/asterisk/acted-bin/recordingcheck
  recordingcheck |-205946 20090827 | 1251399586.64: outgoing recording not enabled
  -Recordingcheck AGI completed Script, return 0
  -Execution [s@macro-record-enable:5] MacroExit ("SIP/205-087af3d8","" "") in new stack
  -[358291@from-internal:4] Macro execution ("SIP/205-087af3d8","dialout-trunk |") 2. 358291 | ") in new stack"
  -Execution [s@macro-dialout-trunk:1] Set ("SIP/205-087af3d8", "DIAL_TRUNK = 2") in new stack
  -Execution [s@macro-dialout-trunk:2] GosubIf ("SIP/205-087af3d8","0?") Sub-pincheck | s | 1 ") in new stack"
  -Execution [s@macro-dialout-trunk:3] GotoIf ("SIP/205-087af3d8","0?") disabletrunk | 1 ") in new stack"
  -Execution [s@macro-dialout-trunk:4] Set ("SIP/205-087af3d8", "DIAL_NUMBER = 358291") in new stack
  -Execution [s@macro-dialout-trunk:5] Set ("SIP/205-087af3d8", "DIAL_TRUNK_OPTIONS = tr") in new stack
  -Execution [s@macro-dialout-trunk:6] Set ("SIP/205-087af3d8", "OUTBOUND_GROUP = OUT_2") in new stack
  -Execution [s@macro-dialout-trunk:7] GotoIf ("SIP/205-087af3d8","0?") nomax ") in new stack"
  -Execution [s@macro-dialout-trunk:8] GotoIf ("SIP/205-087af3d8","0?") chanfull ") in new stack"
  -Execution [s@macro-dialout-trunk:9] GotoIf ("SIP/205-087af3d8","0?") skipoutcid ") in new stack"
  -Execution [s@macro-dialout-trunk:10] Set ("SIP/205-087af3d8", "DIAL_TRUNK_OPTIONS =") in new stack
  -[S@macro-dialout-trunk:11] Macro execution ("SIP/205-087af3d8","outbound callerid |") 2 ") in new stack"
  -[S@macro-outbound-callerid:1] ExecIf execution ("SIP/205-087af3d8","0 |") SetCallerPres | ") in new stack
  -[S@macro-outbound-callerid:2] ExecIf execution ("SIP/205-087af3d8","0 |") The value | REALCALLERIDNUM = 205 ") in new stack
  -Execution [s@macro-outbound-callerid:3] GotoIf ("SIP/205-087af3d8","1?") normcid ") in new stack"
  -Goto (macro-sortant-callerid, s, 6)
  -Execution [s@macro-outbound-callerid:6] Set ("SIP/205-087af3d8", "USEROUTCID =") in new stack
  -Execution [s@macro-outbound-callerid:7] Set ("SIP/205-087af3d8", "EMERGENCYCID =") in new stack
  -Execution [s@macro-outbound-callerid:8] Set ("SIP/205-087af3d8", "TRUNKOUTCID = 383597") in new stack
  -Execution [s@macro-outbound-callerid:9] GotoIf ("SIP/205-087af3d8","1?") trunkcid ") in new stack"
  -Goto (macro-sortant-callerid, s, 12)
  -[S@macro-outbound-callerid:12] ExecIf execution ("SIP/205-087af3d8","1" | ") The value | CALLERID (All) = 383597 ") in new stack
  -Execution [s@macro-outbound-callerid:13] GotoIf ("SIP/205-087af3d8","1?") output ") in new stack"
  -Goto (macro-sortant-callerid, s, 11)
  -Execution [s@macro-outbound-callerid:11] MacroExit ("SIP/205-087af3d8","" "") in new stack
  -[S@macro-dialout-trunk:12] ExecIf execution ("SIP/205-087af3d8","1" | ") AGI | fixlocalprefix") in new stack
  -Launch Script AGI/var/lib/asterisk/acted-bin/fixlocalprefix
  > fixlocalprefix: using chart 0 + [0] [9] X.
  > fixlocalprefix: using mires NXXXXX 0 +.
  == fixlocalprefix: Dialpattern 0 + NXXXXX matched. 358291-> 0358291
  -Fixlocalprefix AGI completed Script, return 0
  -Execution [s@macro-dialout-trunk:13] Set ("SIP/205-087af3d8", "OUTNUM = 0358291") in new stack
  -Execution [s@macro-dialout-trunk:14] Set ("SIP/205-087af3d8", "custom = SIP/pstn") in new stack
  -[S@macro-dialout-trunk:15] ExecIf execution ("SIP/205-087af3d8","0 |") The value | DIAL_TRUNK_OPTIONS = M(setmusic^) ") in new stack"
  -[S@macro-dialout-trunk:16] Macro execution ("SIP/205-087af3d8","dialout-trunk-predial-crochet |" "") in the new battery
  -Execution [s@macro-dialout-trunk-predial-hook:1] MacroExit ("SIP/205-087af3d8","" "") in new stack
  -Execution [s@macro-dialout-trunk:17] GotoIf ("SIP/205-087af3d8","0?") bypass | 1 ") in new stack"
  -Execution [s@macro-dialout-trunk:18] GotoIf ("SIP/205-087af3d8","0?") customtrunk ") in new stack"
  -Execution [s@macro-dialout-trunk:19] Dial ("SIP/205-087af3d8","SIP/pstn/0358291 |") 300. ") in new stack"
  -Called pstn/0358291
  -SIP/pstn - 0885 has 138 sounds
  -SIP/pstn - 0885 has 138 replied SIP/205-087af3d8
  -Packet2Packet bypass 087af3d8/205-SIP and SIP/pstn - 0885 has 138
  -Remote UNIX connection
  -Connection to UNIX distance disconnected
  -[H@macro-dialout-trunk:1] Macro execution ("SIP/205-087af3d8","hangupcall:" "") in the new battery
  -Execution [s@macro-hangupcall:1] ResetCDR ("SIP/205-087af3d8", "w") in new stack
  -Execution [s@macro-hangupcall:2] NoCDR ("SIP/205-087af3d8","" "") in new stack
  -Execution [s@macro-hangupcall:3] GotoIf ("SIP/205-087af3d8","1?") skiprg ") in new stack"
  -Goto (macro-hangupcall, s, 6)
  -Execution [s@macro-hangupcall:6] GotoIf ("SIP/205-087af3d8","1?") skipblkvm ") in new stack"
  -Goto (macro-hangupcall, s, 9)
  -Execution [s@macro-hangupcall:9] GotoIf ("SIP/205-087af3d8","1?") theEnd ") in new stack"
  -Goto (macro-hangupcall, s, 11)
  -Execution [s@macro-hangupcall:11] Hangup ("SIP/205-087af3d8","" "") in new stack
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on "SIP/205-087af3d8" in the macro 'hangupcall '.
  == Spawn extension h (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/205-087af3d8.
  == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/205-087af3d8"macro"dialout-trunk.
  == Spawn extension (of-internal, 358291, 4) exited non-zero on 'SIP/205-087af3d8.

  As you can see he composed, if found the dialing rule, but nothing happens on the SPA, it will not simply compose anything on.

  I guess that something is wrong in RTC spa or configuration of trunk, but I don't know what. I tried like 5-6 guides of strach and none worked for me. Any suggestions?

  Kind regards

  Damir

  HUH! I won't curse now, but I shoul!

  Given that I'm NOT using any VOIP provider and I JUST wanted to use RTC it makes SENSE I don't connect anything to the WAN port, BUT when I configured the WAN port with my LAN IP (172.16.1.200) and put a few other IP on the LAN port, everything started working.

  It's rly rly rly is NOT logical, but I got it working now.

• 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

• 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.