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Unable To Dial Local Numbers With Patton 4960 PRI

Discussion in '3CX Phone System - General' started by nplummer, Dec 11, 2010.

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  1. nplummer

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    We are setting up a new system (4 digit extensions) to connect to a PRI (NI2 US) using a Patton 4960. We are able to receive calls to our DID's and place calls to numbers outside of our area code. However, when we attempt to dial a local number within our area code (dialing with and without he area code) we receive a 3CX audio error of "the call could not be established". We have a blanket oubound rule setup which is directing the calls to the Patton.

    We have already deleted the PSTN device from the 3CX system and re-added it, re-imported the config file to the Patton and rebooted that device. We have also restarted all of the 3CX services and restarted the server operating system.

    Additional System Information: V9 SP5 of 3CX. The Patton is at the recommended firmware of 5.3

    We have the logs set to verbose to see as much detail as possible. Below are 2 example calls. The bottom one is a 3 digit outbount to 611 repair (with the phone company) which was successfull. The top call is a 7 digit dial which fails with a Cause: 500 Internal Server Error

    13:12:10.000 [CM503008]: Call(50): Call is terminated
    13:12:09.015 Currently active calls - 1: [50]
    13:12:07.500 Session 1237 of leg C:50.1 is confirmed
    13:12:07.265 [CM503007]: Call(50): Device joined: sip:EndCall@127.0.0.1:40600;rinstance=fd6ff2c9fb7c1a9d
    13:12:07.265 [CM503007]: Call(50): Device joined: sip:2207@10.10.1.100:5062
    13:12:07.250 [MS210005] C:50.1:Answer provided. Connection(proxy mode):10.10.1.1:7224(7225)
    13:12:07.250 [MS210001] C:50.3:Answer received. RTP connection[unsecure]: 127.0.0.1:40654(40655)
    13:12:07.250 Remote SDP is set for legC:50.3
    13:12:07.250 [CM503002]: Call(50): Alerting sip:EndCall@127.0.0.1:40600;rinstance=fd6ff2c9fb7c1a9d
    13:12:07.046 [CM503025]: Call(50): Calling Unknown:Ext.EndCall@[Dev:sip:EndCall@127.0.0.1:40600;rinstance=fd6ff2c9fb7c1a9d]
    13:12:07.046 [MS210004] C:50.3:Offer provided. Connection(proxy mode): 127.0.0.1:7228(7229)
    13:12:07.046 [CM503016]: Call(50): Attempt to reach <sip:5496778@10.10.1.1> failed. Reason: Server Failure
    13:12:07.046 [CM503003]: Call(50): Call to sip:5496778@10.10.1.2:5060 has failed; Cause: 500 Internal Server Error; from IP:10.10.1.2:5060

    13:12:06.859 [CM503025]: Call(50): Calling Unknown:5496778@(Ln.10000@PRI)@[Dev:sip:10000@10.10.1.2:5060]
    13:12:06.859 [MS210006] C:50.2:Offer provided. Connection(by pass mode): 10.10.1.100:11788(11789)
    13:12:06.812 [CM503004]: Call(50): Route 1: Unknown:5496778@(Ln.10000@PRI)@[Dev:sip:10000@10.10.1.2:5060]
    13:12:06.812 [CM503010]: Making route(s) to <sip:5496778@10.10.1.1>
    13:12:06.812 [MS210000] C:50.1:Offer received. RTP connection: 10.10.1.100:11788(11789)
    13:12:06.812 Remote SDP is set for legC:50.1
    13:12:06.812 [CM505001]: Ext.2207: Device info: Device Identified: [Man: Yealink;Mod: T22;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, no-recvonly] UserAgent: [Yealink SIP-T22P 7.50.0.50] PBX contact: [sip:2207@10.10.1.1:5060]
    13:12:06.796 [CM503001]: Call(50): Incoming call from Ext.2207 to <sip:5496778@10.10.1.1>
    13:12:06.796 [CM500002]: Info on incoming INVITE:
    INVITE sip:5496778@10.10.1.1 SIP/2.0
    Via: SIP/2.0/UDP 10.10.1.100:5062;branch=z9hG4bK1232600085
    Max-Forwards: 70
    Contact: <sip:2207@10.10.1.100:5062>
    To: <sip:5496778@10.10.1.1>
    From: "Alan Forrest"<sip:2207@10.10.1.1>;tag=185545369
    Call-ID: 952765114@10.10.1.100
    CSeq: 2 INVITE
    Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
    Proxy-Authorization: Digest username="2207",realm="3CXPhoneSystem",nonce="414d535c03147a2628:25dfc04f1d5bb2698d9c84bb939ea8c2",uri="sip:5496778@10.10.1.1",response="5d1c83b302a7e884af0cd083149c2b41",algorithm=MD5
    Supported: replaces
    User-Agent: Yealink SIP-T22P 7.50.0.50
    Allow-Events: talk, hold, conference, refer, check-sync
    Content-Length: 0

    13:12:00.390 [CM503008]: Call(49): Call is terminated
    13:11:56.671 Session 1187 of leg C:49.1 is confirmed
    13:11:56.484 [CM503007]: Call(49): Device joined: sip:10000@10.10.1.2:5060
    13:11:56.484 [CM503007]: Call(49): Device joined: sip:2207@10.10.1.100:5062
    13:11:56.468 [MS210007] C:49.1:Answer provided. Connection(by pass mode): 10.10.1.2:4908(4909)
    13:11:56.468 [MS210001] C:49.2:Answer received. RTP connection[unsecure]: 10.10.1.2:4908(4909)
    13:11:56.468 Remote SDP is set for legC:49.2
    13:11:53.406 [CM503002]: Call(49): Alerting sip:10000@10.10.1.2:5060
    13:11:53.265 [CM503025]: Call(49): Calling Unknown:611@(Ln.10000@PRI)@[Dev:sip:10000@10.10.1.2:5060]
    13:11:53.250 [MS210006] C:49.2:Offer provided. Connection(by pass mode): 10.10.1.100:11786(11787)
    13:11:53.218 [CM503004]: Call(49): Route 1: Unknown:611@(Ln.10000@PRI)@[Dev:sip:10000@10.10.1.2:5060]
    13:11:53.218 [CM503010]: Making route(s) to <sip:611@10.10.1.1>
    13:11:53.218 [MS210000] C:49.1:Offer received. RTP connection: 10.10.1.100:11786(11787)
    13:11:53.203 Remote SDP is set for legC:49.1
    13:11:53.203 [CM505001]: Ext.2207: Device info: Device Identified: [Man: Yealink;Mod: T22;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, no-recvonly] UserAgent: [Yealink SIP-T22P 7.50.0.50] PBX contact: [sip:2207@10.10.1.1:5060]
    13:11:53.203 [CM503001]: Call(49): Incoming call from Ext.2207 to <sip:611@10.10.1.1>
    13:11:53.203 [CM500002]: Info on incoming INVITE:
    INVITE sip:611@10.10.1.1 SIP/2.0
    Via: SIP/2.0/UDP 10.10.1.100:5062;branch=z9hG4bK238670062
    Max-Forwards: 70
    Contact: <sip:2207@10.10.1.100:5062>
    To: <sip:611@10.10.1.1>
    From: "Alan Forrest"<sip:2207@10.10.1.1>;tag=1472766217
    Call-ID: 872565176@10.10.1.100
    CSeq: 2 INVITE
    Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
    Proxy-Authorization: Digest username="2207",realm="3CXPhoneSystem",nonce="414d535c03147a1872:839115b632003b24f119ca326ab3d928",uri="sip:611@10.10.1.1",response="e5e4e29268e6cb6364d6f7dfc0377bff",algorithm=MD5
    Supported: replaces
    User-Agent: Yealink SIP-T22P 7.50.0.50
    Allow-Events: talk, hold, conference, refer, check-sync
    Content-Length: 0

    Any suggestions or help would be greatly appreciated.

    Nepal Plummer
    Fahillion Technologies
    805-703-4550
     
  2. alen

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    I had a simmilar issue this week with a Patton 4960. I would be able to dial local numbers and toll free, but long distance would ring and go to busy.

    A firmware update on the patton fixed my issue (was running a firmware from 2007...).

    Hope that helps.

    Cheers.

    Alen
     
  3. nplummer

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    I contacted Patton and they also suggested a firmware update. I updated to the most current release however the problem still exists. We have plugged in a Sangoma PRI card into the same system and we are able to dial out successfully so it seems to point to a configuration problem rather than a carrier issue.

    Nepal
     
  4. leejor

    leejor Well-Known Member

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    I haven't used the Patton but it certainly sounds like there is an internal dialplan/digit length configuration that need to be set up for your area.
     
  5. broadbandcti

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    I am having the exact same problem. same isdn switch type as the OP even.

    running the latest and greatest firmware from patton. I just replaced a cisco call manager express with patton/3cx for a fairly large customer. on the cisco box it was just sending whatever number of digits the customer was dialing. nothing magical about it, number expansion and routing is handled by the carrier.

    right now I'm routing outbound traffic through a bridge connection to another 3cx that has sip trunks as a workaround, but If I don't get this sorted fairly early in the week it's gonna hit the fan. two things are gonna tip them off. 1. outbound faxing will be broken, and 2, there's some kind of odd problem with pstn calls through a bridge link where sometimes the call doesn't drop on the 3cx when the other party hangs up, and sometimes vice versa

    Kinda stressing about it right now.

    since we have the same problem, here's some preliminary debug info. first is the 3cx log entries, followed by the debug output from the smartnode. last bit is using the internal call generator in the smartnode using the sanme calling and called party number. notice that call connects successfully!

    There's something going on the mapping tables between the 3cx and the smartnode and the Carrier's doesn't seem to like something its getting.

    3cx log:

    20:19:18.748 [CM503020]: Normal call termination. Reason: Server Failure
    20:19:18.748 [CM503016]: Call(13): Attempt to reach <sip:86453808@10.209.5.202:5060> failed. Reason: Server Failure
    20:19:18.748 [CM503003]: Call(13): Call to sip:6453808@10.209.5.1:5060 has failed; Cause: 500 Internal Server Error; from IP:10.209.5.1:5060
    20:19:18.529 [CM503025]: Call(13): Calling Unknown:6453808@(Ln.10002@patton-SN)@[Dev:sip:10002@10.209.5.1:5060]
    20:19:18.529 [MS210006] C:13.2:Offer provided. Connection(by pass mode): 10.210.3.54:40044(40045)
    20:19:18.482 [CM503004]: Call(13): Route 1: Unknown:6453808@(Ln.10002@patton-SN)@[Dev:sip:10002@10.209.5.1:5060]
    20:19:18.482 [CM503010]: Making route(s) to <sip:86453808@10.209.5.202:5060>
    20:19:18.482 [MS210000] C:13.1:Offer received. RTP connection: 10.210.3.54:40044(40045)
    20:19:18.467 Remote SDP is set for legC:13.1
    20:19:18.467 [CM505001]: Ext.499: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone 4.0.13527.0] PBX contact: [sip:499@10.209.5.202:5060]
    20:19:18.467 [CM503001]: Call(13): Incoming call from Ext.499 to <sip:86453808@10.209.5.202:5060>
    20:19:18.467 [CM500002]: Info on incoming INVITE:
    INVITE sip:86453808@10.209.5.202:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.210.3.54:53053;branch=z9hG4bK-d8754z-6f0e665d290e8414-1---d8754z-;rport=53053
    Max-Forwards: 70
    Contact: <sip:499@10.210.3.54:53053;rinstance=09cc41c4a5b1fd68>
    To: <sip:86453808@10.209.5.202:5060>
    From: "Steven McNutt"<sip:499@10.209.5.202:5060>;tag=331f2d2b
    Call-ID: ZTZmNzI4ZmZkNzc2OTVmNDgxNWVkMmJhNTFlY2MyYTg.
    CSeq: 2 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
    Proxy-Authorization: Digest username="499",realm="3CXPhoneSystem",nonce="414d535c031dee9681:2e53608bb1cd343aa5894800ef00c263",uri="sip:86453808@10.209.5.202:5060",response="5ead506df61a3491a438295efa823074",algorithm=MD5
    Supported: replaces
    User-Agent: 3CXPhone 4.0.13527.0
    Content-Length: 0

    smartnode debug output from 3cx initiated call:

    10.208.5.129#01:19:23 ICC > [IF_ISDN_0] Added endpoint IF_ISDN_0-00b5c780
    01:19:23 ICC > [IF_ISDN_0] NEW CALL. Allocated Endpoint IF_ISDN_0-00b5c780
    01:19:23 ICC > [EP IF_ISDN_0-00b5c780] do not reroute because: no endpoint wi
    th call key: ZmI3NmQ5ZWRmMWExMTcyNzc3MWU1NDc1YTJmMGYzOGQ.
    01:19:23 ICC > [EP IF_ISDN_0-00b5c780] AOC-S User << Call Setup
    01:19:23 ICC > [EP IF_ISDN_0-00b5c780] AOC-D User << Call Setup
    01:19:23 ICC > [EP IF_ISDN_0-00b5c780] AOC-E User << Call Setup
    01:19:23 ICC > [EP IF_ISDN_0-00b5c780] CNIP << Name Request
    01:19:23 ICC > [EP IF_ISDN_0-00b5c780] >> [080005]
    SETUP (NI2 User)
    [04039090A2]
    Bearer capability : 3.1kHz Audio - CCITT
    circuit mode - 64kBit/s - G.711 U-law
    [1C179FA114020101020100800C7374657665206D636E757474]
    Facility : Invoke : invokeid: 00000001
    global operation
    invoke {
    present = 1
    local = 0
    argumentnamePresentationAllowedSimple = 'steve mcnutt'
    }
    [1E028583]
    Progress indicator : origination address is non-ISDN
    private network serving remote user - CCITT
    [6C0700803130303032]
    Calling party number : 10002
    unknown number - unknown numbering plan
    presentation allowed - user provided not screened
    [70088036343533383038]
    Called party number : 6453808
    unknown number - unknown numbering plan

    01:19:23 ICC > [EP IF_ISDN_0-00b5c780] Set state to CALL INITIATED
    01:19:23 ICC > [EP IF_ISDN_0-00b5c780] State: CALL INITIATED, Event: PEER CON
    NECTED
    01:19:23 ICC > [EP IF_ISDN_0-00b5c780] Hold State: IDLE, Hold Event: PEER CON
    NECTED
    01:19:24 ICC > [IF_ISDN_0] << Message: primitive=68
    01:19:24 ICC > [EP IF_ISDN_0-00b5c780] << [08007D]
    STATUS (NI2 User)
    [0802829E]
    Cause : response to status enquiry
    public network serving local user - CCITT - Q.931
    [140100]
    Call state...

    01:19:24 ICC > [EP IF_ISDN_0-00b5c780] State: CALL INITIATED, Event: TERMINAL
    STATUS IND
    01:19:24 ICC > [EP IF_ISDN_0-00b5c780] Set state to NULL
    01:19:24 ICC > [IF_ISDN_0] CLEARING CALL IF_ISDN_0-00b5c780
    01:19:24 ICC > [IF_ISDN_0] Removed endpoint IF_ISDN_0-00b5c780
    01:19:24 ICC > [IF_ISDN_0] Destroying finished calls.
    01:19:24 ICC > [IF_ISDN_0] Destroyed endpoint IF_ISDN_0-00b5c780
    01:19:24 ICC > [IF_ISDN_0] << Message: primitive=48
    01:19:24 IERR > [IF_ISDN_0] MESSAGE FOR UNKNOWN CALL KEY 152 (primitive=48)

    Smartnode output using internal call generator

    10.208.5.129#call 10002 dial 6453808 dest-interface IF_ISDN_0
    10.208.5.129#01:33:02 ICC > [IF_ISDN_0] Added endpoint IF_ISDN_0-00b4d0d8
    01:33:02 ICC > [IF_ISDN_0] NEW CALL. Allocated Endpoint IF_ISDN_0-00b4d0d8
    01:33:02 ICC > [EP IF_ISDN_0-00b4d0d8] do not reroute because: call key prope
    rty missing
    01:33:02 ICC > [EP IF_ISDN_0-00b4d0d8] AOC-S User << Call Setup
    01:33:02 ICC > [EP IF_ISDN_0-00b4d0d8] AOC-D User << Call Setup
    01:33:02 ICC > [EP IF_ISDN_0-00b4d0d8] AOC-E User << Call Setup
    01:33:02 ICC > [EP IF_ISDN_0-00b4d0d8] >> [080005]
    SETUP (NI2 User)
    [04038090A2]
    Bearer capability : speech - CCITT
    circuit mode - 64kBit/s - G.711 U-law
    [1E028583]
    Progress indicator : origination address is non-ISDN
    private network serving remote user - CCITT
    [6C0700803130303032]
    Calling party number : 10002
    unknown number - unknown numbering plan
    presentation allowed - user provided not screened
    [70088036343533383038]
    Called party number : 6453808
    unknown number - unknown numbering plan
    [7D029181]
    High layer compatibility : telephony
    CCITT

    01:33:02 ICC > [EP IF_ISDN_0-00b4d0d8] Set state to CALL INITIATED
    01:33:02 ICC > [EP IF_ISDN_0-00b4d0d8] State: CALL INITIATED, Event: PEER CON
    NECTED
    01:33:02 ICC > [EP IF_ISDN_0-00b4d0d8] Hold State: IDLE, Hold Event: PEER CON
    NECTED
    01:33:02 ICC > [EP IF_ISDN_0-00b4d0d8] State: CALL INITIATED, Event: PEER INB
    AND INFO
    01:33:03 ICC > [IF_ISDN_0] << Message: primitive=43
    01:33:03 ICC > [EP IF_ISDN_0-00b4d0d8] << [080002]
    CALL PROCEEDING (NI2 User)
    [1803A98382]
    Channel id : 2 - exclusive
    other interface - is not d-channel - CCITT - b-channel units

    01:33:03 ICC > [EP IF_ISDN_0-00b4d0d8] State: CALL INITIATED, Event: TERMINAL
    PROCEEDING IND
    01:33:03 ICC > [EP IF_ISDN_0-00b4d0d8] Set state to OUTGOING PROCEEDING
    01:33:04 ICC > [IF_ISDN_0] << Message: primitive=31
    01:33:04 ICC > [EP IF_ISDN_0-00b4d0d8] << [080001]
    ALERTING (NI2 User)
    [1E028488]
    Progress indicator : inband information available
    public network serving remote user - CCITT

    01:33:04 ICC > [EP IF_ISDN_0-00b4d0d8] State: OUTGOING PROCEEDING, Event: TER
    MINAL ALERTING IND
    01:33:04 ICC > [EP IF_ISDN_0-00b4d0d8] Set state to CALL DELIVERED
    01:33:29 ICC > [IF_ISDN_0] << Message: primitive=62
    01:33:29 ICC > [EP IF_ISDN_0-00b4d0d8] << [080007]
    CONNECT (NI2 User)
    [1E028A81]
    Progress indicator : call is not end-to-end isdn, inband info may be present
    network beyond interworking point - CCITT

    01:33:29 ICC > [EP IF_ISDN_0-00b4d0d8] State: CALL DELIVERED, Event: TERMINAL
    SETUP CONF
    01:33:29 ICC > [EP IF_ISDN_0-00b4d0d8] Set state to ACTIVE
     
  6. nplummer

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    I have some great news. I was able to work with the carrier (Telepacific) and a Patton engineer directly on the phone on Friday and we worked through the problem.

    You may need to add some items to the configuration and make one small change in the Web Gui. Basically, the carrier doesn't want to see any facility messages in the setup.

    First , try this. In the Web Gui navigate to:
    Telephony ---- > Call-Router ---- > ISDN Interface IF_ISDN_0
    Uncheck the item near the bottom under the section "Caller Name". It is labeled Accept/Emit. Uncheck that item and save.


    If that doesn't solve the problem you can try adding this to the configuration (it is what the Patton Engineer sent me).

    mapping-table called-type-of-number to called-type-of-number MAP_TON
    map unknown to national

    mapping-table called-numbering-plan to called-numbering-plan MAP_NUM_PLAN
    map unknown to isdn-telephony

    mapping-table itc to itc MAP_ITC
    map 3k1-audio to speech

    complex-function USE_MAPPING_TABLES
    execute 1 MAP_ITC
    execute 2 MAP_NUM_PLAN
    execute 3 MAP_TON


    You may not need to add that information to your config. I would try the Web Gui fix first. Fortunately I have to say that Patton's support was fantastic. I was abel to escalate to a higher level engineer fairly quickly and he was super responsive. He was willing to speak directly to the carrier and try to get things worked out. Don't give up, it will work.

    Nepal Plummer
    Fahillion Technologies
    http://www.fahillion.com
     
  7. broadbandcti

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    looks like it was the same story. The smartnode was sending extra information during call setup the carrier's switch didn't care about so it was kicking back a status code that basically amounts to huh?!?
     
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