Outgoing/Incoming Issue - Linksys SPA 3102

Discussion in '3CX Phone System - General' started by aad999, Mar 7, 2009.

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  1. aad999

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    We are not able to make PSTN outgoing or incoming call using Linksys spa 3102. I have configured the linksys device as per the 3CX document.
    Local extensions work fine.

    Telephone line ----Connected to Line port
    Ethernet cable----Connected to Ethernet port

    Outbound rule is created to dial PSTN outgoing number.

    When i dial PSTN outgoing number the call ring one time and shows connected.
    When i dial in my PSTN number it keeps rining.

    22:43:51.015 [CM503007]: Call(18): Device joined: sip:10001@127.0.0.1:5061

    22:43:51.015 [CM503007]: Call(18): Device joined: sip:1001@192.168.1.112:5070;rinstance=14175665db07b636

    22:43:51.015 [CM505002]: Gateway:[MyLinksys] Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-3.3.6(GW)] Transport: [sip:192.168.1.101:5060]

    22:43:51.000 [CM503002]: Call(18): Alerting sip:10001@127.0.0.1:5061

    22:43:50.390 [CM503004]: Call(18): Calling: PSTNline:99857687@(Ln.10001@MyLinksys)@[Dev:sip:10001@127.0.0.1:5061]

    22:43:50.359 [CM503010]: Making route(s) to <sip:099857687@192.168.1.101:5060>

    22:43:50.359 [CM505001]: Ext.1001: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CX VoIP Client;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CX Phone 7.0.4526.0] Transport: [sip:192.168.1.101:5060]

    22:43:50.343 [CM503001]: Call(18): Incoming call from Ext.1001 to <sip:099857687@192.168.1.101:5060>

    Any help would appreciated.
     
  2. Cjay

    Cjay New Member

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    Hi,
    I can't quite get my head around the following line in your 3cx log:

    22:43:50.390 [CM503004]: Call(18): Calling: PSTNline:99857687@(Ln.10001@MyLinksys)@[Dev:sip:10001@127.0.0.1:5061]

    To me this would suggest that 3cx thinks your 3102 is on the same IP as your 3cx server as it is trying to use the internal loopback IP - which clearly can't be the actual case. You need to see the real IP of the 3102 in this line!

    Is the 3102 showing correctly registered in 'PSTN devices section?

    Chris
     
  3. aad999

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    Yes, it is registered correctly in 3cx system. Even i am surprised to see the localhost ip. Where can i go and change it?
     
  4. Cjay

    Cjay New Member

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    What do you have if you click on 'PSTN Devixce' & 'Edit Gateway' for your 3102 device? You should have the real IP of your 3102 under the general tab.
     
  5. aad999

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    Yes it is there...192.168.1.250 (SPA 3102).
     
  6. Cjay

    Cjay New Member

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    Strange! Does your 3102 show registered? If you log in to this device does the Voice/Info tab show registered in the 'PSTN Line Status' section?
     
  7. aad999

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    I did not check PSTN Line Status. I will check it out and let you know once i go back to office.
     
  8. aad999

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    The PSTN line status shows onPSTN Line Status
    Hook State: On
    Line Voltage: 50 (V)
    Loop Current: 0.0 (mA)
    Registration State: Registered
    Last Registration At: 1/1/2003 12:00:08
    Next Registration In: 59 s
    Last Called VoIP Number:
    Last Called PSTN Number:
    Last VoIP Caller: Last PSTN Caller: ,
    Last PSTN Disconnect Reason:
    PSTN Activity Timer: 300000 (ms)
    Mapped SIP Port: Call Type:
    VoIP State: Idle PSTN State: Idle
     
  9. discovery1

    discovery1 Member

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    If you check the properties of the port 10001 do you have the options ticked to "allow incoming calls" and "allow outbound calls"

    Also - the Ethernet cable should be connected to the WAN port on the SPA-3102
     
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  10. aad999

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    Thanks cjay & discovery1.
    The issue is resolved by connecting to WAN port. I wonder why do we have ethernet port and why is it called LAN IP. It is very confusing.Would appreciate if you explain.
    Thanks again.
     
  11. discovery1

    discovery1 Member

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    The LAN port is used when the 3102 is being used as an internet router and performing NAT to the LAN.

    Since we are using it as an ATA we want the SIP packets from 3CX (which would normally come from the VOIP provider on the internet/WAN) to arrive and exit on the WAN port and be passed through to the PSTN line.

    The setup guide mentions the WAN port setup in steps 6 and 7 but does not explicitly tell you to plug the network cable into the WAN port.

    http: //www.3cx.com/voip-gateways/linksys-3102/ - NO LONGER AVAILABLE
     
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  12. leejor

    leejor Well-Known Member

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    If you already have a router on your network you can set the 3102 to "Bridge" mode. You can then plug a computer or another VoIP device into the LAN port and it will get it's IP from your router as well. Using the 3102 as a router really only makes sense if it is being used directly on the internet and you have no other router to plug you other devices into.
     
  13. William400

    William400 Well-Known Member

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    Hi

    In V7.1 we have a template for the 3102 that allows configuration of the device by means of provisioning.

    Once you finish the config and save the configuration, the same configuration is saved to the 3CX provisioning folder.

    Then go to the Linksys interface;

    Click on the "Admin" > "Adavnced" > “Provisioning” page.
    In the “Provisioning” page “Profile Rule” option, input http://10.172.0.116/management/provisioning/linksys3102.xml where in this example, 10.172.0.116 is the IP of the IP PBX web interface. If using Cassini, the URL is http://10.172.0.116:5481/provisioning/linksys3102.xml.

    This should give you a headstart to getting the device running.
     
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  14. alan2010iq

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    Hi,

    I am having the same problem with SPA3102 running with 3CX V9.0. The PSTN line statues shows the following:

    The PSTN line status shows onPSTN Line Status
    Hook State: On
    Line Voltage: 50 (V)
    Loop Current: 0.0 (mA)
    Registration State: Registered
    Last Registration At: 15/5/2011 16:21:02
    Next Registration In: 340 s
    Last Called VoIP Number:
    Last Called PSTN Number:
    Last VoIP Caller: Last PSTN Caller: ,
    Last PSTN Disconnect Reason:
    PSTN Activity Timer: 300000 (ms)
    Mapped SIP Port: Call Type:
    VoIP State: Idle PSTN State: Idle

    and the VoIP peer is 199@127.0.0.1

    I think the problem is that the local extension registered on the Line 1 shows as @127.0.0.1 instead the IP address of the 3CX system. I have upgraded the SAP3102 firmwware to V5.0, the same problem. I have also used other brand new SPA3102 with default settings after configuring, the problem remains the same. The ethernet cable is connected to Internet port.

    I have also checked the SPA3102 debug syslog and the DTMF digits of the calling number on the PSTN are all correct.

    I wonder why the registration of line 1 is made as ext@127.0.0.1 instead of 127@3cxIP????

    Any help would be much appreciated as I am really puzzled with this problem for the last couple of days.

    Thanks,

    Alan
     
  15. dreamscape

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    Hi Alan

    I have the problem where I can't make outgoing calls through the 3102 onto PSTN, but can receive from PSTN -3102 -3CX- ext. (I have found that it doesn't work if you use the * for the CallerID pattern.)

    When making an outgoing call, I hear one ringtone and then the call ends.

    In order of what I have done to this point.
    1. Factory reset the 3102.
    2. Logged in, opened up WAN config, set static WAN IP and got cennection to the network via WAN port.
    3. Created new Trunk for Linksys 3102 NEW
    4. Entered correct IP
    5. Created config file although have not successfully managed to provision although I can find the xml via cassini url (i've had to input from 3CX guide manually)
    6. Used 3CX guide to make changes, turning off Line 1 etc.
    7. Added dial plan 8 for port
    8. Changed region settings inline with Andrew Oakleys website for UK
    9. Got green light in trunk status.

    (This config works for Skype Gateway. Can definately recieve calls ok although I haven't tried making calls out because I don't really know how to even go about it!)

    The 3CX logs show incoming calls all ok, but when I dial out it shows no answer.

    It doesn't look to me as if outgoing calls are being passed into the 3102 for it to dial out.

    Is dial plan 8 used to dial back out, or should line 1 be enabled? I'm confused!?

    Thanks

    Mark

    Incoming Call
    11:45:34.834 [CM503008]: Call(2): Call is terminated
    11:45:29.499 [CM503007]: Call(2): Device joined: sip:101@192.168.1.124:5060
    11:45:29.499 [CM503007]: Call(2): Device joined: sip:10003@192.168.1.121:5062
    11:45:21.340 [CM505001]: Ext.101: Device info: Device Identified: [Man: Cisco;Mod: SPA Series;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, no-recvonly] UserAgent: [Cisco/SPA504G-7.4.8] PBX contact: [sip:101@192.168.1.120:5060]
    11:45:21.340 [CM503002]: Call(2): Alerting sip:101@192.168.1.124:5060
    11:45:21.246 [CM503025]: Call(2): Calling Ext:Ext.101@[Dev:sip:101@192.168.1.124:5060]
    11:45:21.231 [CM503004]: Call(2): Route 1: Ext:Ext.101@[Dev:sip:101@192.168.1.124:5060]
    11:45:21.231 [CM503010]: Making route(s) to <sip:101@192.168.1.120:5060>
    11:45:21.215 [CM505002]: Gateway:[Linksys] Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-5.1.10(GW)] PBX contact: [sip:10003@192.168.1.120:5060]
    11:45:21.215 [CM503001]: Call(2): Incoming call from 07815??????@(Ln.10003@Linksys) to <sip:101@192.168.1.120:5060>
    11:45:21.184 [CM503012]: Inbound any hours rule (*) for 10003 forwards to DN:101

    Outgoing Call

    11:51:45.709 [CM503008]: Call(4): Call is terminated
    11:51:43.572 [CM503007]: Call(4): Device joined: sip:10003@192.168.1.121:5062
    11:51:43.572 [CM503007]: Call(4): Device joined: sip:101@192.168.1.124:5060
    11:51:43.431 Currently active calls - 1: [4]
    11:51:42.542 [CM505002]: Gateway:[Linksys] Device info: Device Identified: [Man: Linksys;Mod: SPA Series;Rev: General] Capabilities:[reinvite, no-replaces, able-no-sdp, recvonly] UserAgent: [Linksys/SPA3102-5.1.10(GW)] PBX contact: [sip:10003@192.168.1.120:5060]
    11:51:42.542 [CM503002]: Call(4): Alerting sip:10003@192.168.1.121:5062
    11:51:42.464 [CM503025]: Call(4): Calling PSTNline:00447815??????@(Ln.10003@Linksys)@[Dev:sip:10003@192.168.1.121:5062]
    11:51:42.449 [CM503004]: Call(4): Route 1: PSTNline:00447815??????@(Ln.10003@Linksys)@[Dev:sip:10003@192.168.1.121:5062]
    11:51:42.433 [CM503010]: Making route(s) to "10003"<sip:07815??????@192.168.1.120>
    11:51:42.433 [CM505001]: Ext.101: Device info: Device Identified: [Man: Cisco;Mod: SPA Series;Rev: General] Capabilities:[reinvite, replaces, unable-no-sdp, no-recvonly] UserAgent: [Cisco/SPA504G-7.4.8] PBX contact: [sip:101@192.168.1.120:5060]
    11:51:42.417 [CM503001]: Call(4): Incoming call from Ext.101 to "10003"<sip:07815??????@192.168.1.120>


    3cx .120
    3102 .121 port 10003
    Phone EXT .124 ext 101
     
  16. alan2010iq

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    Hi,

    In my case, I have made some progress in understanding the problem, however, I have no solution so far. What is certain now is that the problem of no caller ID of incoming PSTN calls is related to caller ID method or PSTN caller ID pattern. When I put * in the PSTN caller ID pattern, nothing detected and 3CX does not detected the local VoIP call even as 10007 which the PSTN virtual extension. However, when I leave the PSTN caller ID pattern blank, then 3CX detects the call as 10007 but disconnects and ring comes up again in sequence on the operator extension!!!! which is very strange.
    So I think I am having a problem that the SPA3102 does not detects the caller ID correctly. Looking at the debug log of another SPA3102 on a different PSTN line which is working ok, the log shows start digits + caller ID digits and stop digit. However, on the SPA3102 which has the caller ID detection problem, it only shows the calling number digits without start and stop digits. That is why I think the caller ID is not detected correctly by the SPA3102.

    I am really strugling to find out a solution for this.

    Alan
     
  17. dreamscape

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    I've actually got mine working now! Upped the gain for the PSTN to VOIP to 6 and stopped using the international dial codes as suggested elsewhere. Just calling standard UK national numbers with no strip digit on the outbound rule.

    I haven't had any luck with the * in the field and have left it blank.

    I have read that lengthening the delay on the answer may help with the caller id, but as mine seems to work fine with out it, i've left it alone.

    You might want to check what number is actually being passed through to the 3102 vs which number you are passing through the outbound rules, thats what made me suddenly think when I tried to dial out on a stand line using the full international number that i thought should have worked.

    Best of luck Alan
     
  18. leejor

    leejor Well-Known Member

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    This will depend how (soon)caller Id is sent in your area (country). The 3102 has to wait long enough to capture the CID info before alerting 3CX of the incoming call or it will show as anonymous.

    I have always left that field blank.

    Don't forget that there is a syslog function in the 3102 that you can use with a syslog program ( http://www.kiwisyslog.com/kiwi-syslog-server-overview/) running on a PC. This will show what the 3102 is actually doing with incoming digits.
     
  19. alan2010iq

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    Hi,

    This is the syslog for my SPA3102 which fails to detect incoming caller ID of PSTN calss:

    05-17-2011 01:17:02 Local2.Debug 192.168.1.44 FXO:Stop CNDD
    05-17-2011 01:17:02 Local2.Debug 192.168.1.44 AUD:Stop PSTN Tone
    05-17-2011 01:17:02 Local3.Debug 192.168.1.44 FXO:On Hook
    05-17-2011 01:17:02 Local2.Debug 192.168.1.44 AUD:Stop PSTN Tone
    05-17-2011 01:17:01 Local2.Debug 192.168.1.11 AUD:Stop PSTN Tone
    05-17-2011 01:17:01 Local3.Debug 192.168.1.11 [1]RegOK. NextReg in 1784 (1)
    05-17-2011 01:17:00 Local3.Debug 192.168.1.11 [0]RegOK. NextReg in 1784 (1)
    05-17-2011 01:16:56 Local2.Debug 192.168.1.44 FXO:Start CNDD
    05-17-2011 01:16:55 Local3.Debug 192.168.1.44 FXO:Digit=0
    05-17-2011 01:16:54 Local3.Debug 192.168.1.44 FXO:Digit=0
    05-17-2011 01:16:54 Local3.Debug 192.168.1.44 FXO:Digit=3
    05-17-2011 01:16:54 Local3.Debug 192.168.1.44 FXO:Digit=2
    05-17-2011 01:16:54 Local3.Debug 192.168.1.44 FXO:Digit=2
    05-17-2011 01:16:54 Local3.Debug 192.168.1.44 FXO:Digit=1
    05-17-2011 01:16:54 Local3.Debug 192.168.1.44 FXO:Digit=2
    05-17-2011 01:16:53 Local3.Debug 192.168.1.44 FXO:Digit=8
    05-17-2011 01:16:53 Local3.Debug 192.168.1.44 FXO:Digit=9
    05-17-2011 01:16:53 Local3.Debug 192.168.1.44 FXO:Digit=7
    05-17-2011 01:16:53 Local3.Debug 192.168.1.44 FXO:Digit=0


    As seen, the FXO:Start CNDD is after the DTMF digits are received, while this is not the case on my other SPA3102 which works fine with no problem with PSTN caller ID. Below is the syslog for the other SPA3102 which works and detects the caller ID correctly:

    05-17-2011 01:20:48 Local2.Debug 192.168.1.11 [1]AUD ALLOC CALL (port=16442)
    05-17-2011 01:20:48 Local2.Debug 192.168.1.11 Calling:10005@192.168.1.10:0
    05-17-2011 01:20:48 Local2.Debug 192.168.1.11 AUD:Stop PSTN Tone
    05-17-2011 01:20:48 Local2.Debug 192.168.1.11 AUD:Stop PSTN Tone
    05-17-2011 01:20:48 Local3.Debug 192.168.1.11 FXO:CNDD Name= Phone=07701521300
    05-17-2011 01:20:48 Local3.Debug 192.168.1.11 FXO:Digit=c
    05-17-2011 01:20:48 Local3.Debug 192.168.1.11 FXO:Digit=9
    05-17-2011 01:20:48 Local3.Debug 192.168.1.11 FXO:Digit=7
    05-17-2011 01:20:48 Local3.Debug 192.168.1.11 FXO:Digit=7
    05-17-2011 01:20:47 Local3.Debug 192.168.1.11 FXO:Digit=2
    05-17-2011 01:20:47 Local3.Debug 192.168.1.11 FXO:Digit=1
    05-17-2011 01:20:47 Local3.Debug 192.168.1.11 FXO:Digit=3
    05-17-2011 01:20:47 Local3.Debug 192.168.1.11 FXO:Digit=2
    05-17-2011 01:20:47 Local3.Debug 192.168.1.11 FXO:Digit=7
    05-17-2011 01:20:47 Local3.Debug 192.168.1.11 FXO:Digit=9
    05-17-2011 01:20:47 Local3.Debug 192.168.1.11 FXO:Digit=7
    05-17-2011 01:20:47 Local3.Debug 192.168.1.11 FXO:Digit=0
    05-17-2011 01:20:47 Local0.Info 192.168.1.11 FXO:DTMF CID START a
    05-17-2011 01:20:47 Local0.Info 192.168.1.11 FXO:DTMF CID START a
    05-17-2011 01:20:46 Local3.Debug 192.168.1.11 FXO:Digit=a
    05-17-2011 01:20:46 Local2.Debug 192.168.1.11 FXO:Start CNDD


    The above shows that in the first case, the FXO:Start CNDD starts after the FXO DTMF digits are received and the DTMF CID START a exists while in the second SPA3102 that does not exist? I am confused and don't know what to do.

    Appreciate any help on this.

    Alan
     
  20. leejor

    leejor Well-Known Member

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    Are the two 3102's the same firmware load? Are you testing them on the same phone line? Have you compared settings side by side (two browsers open on the same computer) to confirm that everything but the unique trunk settings/password, are the same?

    Have you tried backing up the working unit, doing a factory reset on the non-working unit and applying the working "load"?

    http://homecommunity.cisco.com/t5/VoIP-Routers/SPA-3102-Backup-Restore-settings/td-p/258658

    In the UK is there not a current reversal, and some sort of line test, shortly before the CID digits are sent? It may be that you have a defective unit if you can't get it working with exactly the same settings as the working one, after a factory reset.
     
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