Assistance in getting a FXO gateway working with 3 POTS lines

Discussion in '3CX Phone System - General' started by Ted Mittelstaedt, Jul 16, 2017.

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  1. Ted Mittelstaedt

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    Hi All,

    This weekend I just downloaded and installed the latest Debian ISO of 3Cx on an ESXi system of mine.

    So far everything appears to work except for one thing - incoming calls from PSTN

    I can make calls in between soft extensions, whether the extensions are behind the address translator or not.

    I can call out to the PSTN through the FXO gateway from soft extensions no problem.

    I just cannot receive inbound calls.

    The gateway is a Cisco 2600 with 4 voice FXO cards in it.

    The incoming POTS lines plug into that.

    Configs for the Cisco are from here:

    http://web.archive.org/web/20050212113924/http://www.tape.net/~gerry/asterisk/cisco26x0.html

    and here

    https://www.3cx.com/community/threads/3cx-and-cisco-router.20142/

    and here

    https://www.voip-info.org/wiki-Asterisk+cisco+FXO

    Note that I have IOS 12.2, Cisco did not introduce userID authorization into SIP until IOS 12.4

    One last thing:

    I would appreciate you not attempting to show your brilliance by telling me to get a different gateway. I am already aware of that "solution" It so happens that I had the hardware already and BEFORE I sink a lot of time and money into 3CX I'm willing to give it a try with what I have. If it does not work I can walk away from it now and write it off as just another OSS derivative product that has evolved into a turnkey system that isn't interoperable with anything anymore.

    I've already looked through the Linux command line and it seems plain that 3CX has changed all the configs so they can only be accessed by the web interface. So tricks like modifying sip.conf won't work. I suspect that if 3cx put a GENERIC entry for a regular SIP trunk in I could define 3 of them and point the Cisco to that and it would work.

    It seems plain that this configuration worked in the past with 3CX and it works with current asterisk. I'm quite willing to assume I have something wrong in the 3CX config.

    Thanks! I'll be happy to post configs/ router syslogs if anyone is interested.
     
  2. leejor

    leejor Well-Known Member

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    You have won half the battle by getting outgoing calls working, that usually proves to be more difficult for most.
    If you haven't done so, check the 3CX Activity log for an incoming call attempt. If here is one, it should show where 3Cx is trying to send the call, which would help pinpoint ant "incoming rules" settings that you have incorrect. It could also be a blacklisted IP.

    If there is no sign of a call attempt from the gateway, then you may have to use a Syslog feature on the gateway to what is happening internally and where it is sending the call, or why it isn't. In essence, the gateway "forwards" the call to the trunk number @ the 3CX IP/port.
     
  3. Ted Mittelstaedt

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    leejor,

    Thanks for responding. There is no entry at all in the activity log. (there are tons of entries from boneheads out on the Internet attempting to connect to the PBX probably to try getting free calls of course) Here is part of the debug output from the Cisco when the FXO port gets the call from the PSTN and tries opening a SIP connection to 3CX (the Cisco is 172.16.1.6 and the 3CX is 172.16.1.8)

    Jul 16 15:34:19 172.16.1.6 5649: 15:11:29: 0x8272FD74 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
    Jul 16 15:34:19 172.16.1.6 5650: 15:11:29: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
    Jul 16 15:34:19 172.16.1.6 5651: 15:11:29: CCSIP-SPI-CONTROL: act_idle_call_setup
    Jul 16 15:34:19 172.16.1.6 5652: 15:11:29: Using Voice Class Codec, tag = 1
    Jul 16 15:34:19 172.16.1.6 5653:
    Jul 16 15:34:19 172.16.1.6 5654: 15:11:29: act_idle_call_setup: preferred_codec set[0] type :g729r8 bytes: 20
    Jul 16 15:34:19 172.16.1.6 5655: 15:11:29: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION
    Jul 16 15:34:19 172.16.1.6 5656: 15:11:29: 0x8272FD74 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_CONNECTING)
    Jul 16 15:34:19 172.16.1.6 5657: 15:11:29: 0x8272FD74 : State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_IDLE, SUBSTATE_CONNECTING)
    Jul 16 15:34:19 172.16.1.6 5658: 15:11:29: CCSIP-SPI-CONTROL: act_idle_connection_created
    Jul 16 15:34:19 172.16.1.6 5659: 15:11:29: CCSIP-SPI-CONTROL: act_idle_connection_created: Connid(1) created to 172.6.1.8:5060, local_port 54147
    Jul 16 15:34:19 172.16.1.6 5660: 15:11:29: sipSPIAddLocalContact
    Jul 16 15:34:19 172.16.1.6 5661: 15:11:29: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
    Jul 16 15:34:19 172.16.1.6 5662: 15:11:29: CCSIP-SPI-CONTROL: sip_stats_method
    Jul 16 15:34:19 172.16.1.6 5663: 15:11:29: 0x8272FD74 : State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_SENT_INVITE, SUBSTATE_NONE)
    Jul 16 15:34:19 172.16.1.6 5664: 15:11:29: Sent:
    Jul 16 15:34:19 172.16.1.6 5665: INVITE sip:102@172.6.1.8;user=phone;phone-context=unknown SIP/2.0^M
    Jul 16 15:34:19 172.16.1.6 5666: Via: SIP/2.0/UDP 172.16.1.6:5060^M
    Jul 16 15:34:19 172.16.1.6 5667: From: "4452257" <sip:4452257@172.16.1.6>^M
    Jul 16 15:34:19 172.16.1.6 5668: To: <sip:102@172.6.1.8;user=phone;phone-context=unknown>^M
    Jul 16 15:34:19 172.16.1.6 5669: Date: Mon, 01 Mar 1993 15:11:33 GMT^M
    Jul 16 15:34:19 172.16.1.6 5670: Call-ID: DF8B6E8D-156D11CC-80E498B6-701E77C@172.16.1.6^M
    Jul 16 15:34:19 172.16.1.6 5671: Cisco-Guid: 3750416997-359469516-2162333878-117565308^M
    Jul 16 15:34:19 172.16.1.6 5672: User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled^M
    Jul 16 15:34:19 172.16.1.6 5673: CSeq: 101 INVITE^M
    Jul 16 15:34:19 172.16.1.6 5674: Max-Forwards: 6^M
    Jul 16 15:34:19 172.16.1.6 5675: Timestamp: 730998693^M
    Jul 16 15:34:19 172.16.1.6 5676: Contact: <sip:4452257@172.16.1.6:5060;use
    Jul 16 15:34:19 172.16.1.6 5677: r=phone>^M
    Jul 16 15:34:19 172.16.1.6 5678: Expires: 300^M
    Jul 16 15:34:19 172.16.1.6 5679: Content-Type: application/sdp^M
    Jul 16 15:34:19 172.16.1.6 5680: Content-Length: 137^M
    Jul 16 15:34:19 172.16.1.6 5681: ^M
    Jul 16 15:34:19 172.16.1.6 5682: v=0^M
    Jul 16 15:34:19 172.16.1.6 5683: o=CiscoSystemsSIP-GW-UserAgent 8000 4684 IN IP4 172.16.1.6^M
    Jul 16 15:34:19 172.16.1.6 5684: s=SIP Call^M
    Jul 16 15:34:19 172.16.1.6 5685: c=IN IP4 172.16.1.6^M
    Jul 16 15:34:19 172.16.1.6 5686: t=0 0^M
    Jul 16 15:34:19 172.16.1.6 5687: m=audio 20740 RTP/AVP 18 0 4 2^M
    Jul 16 15:34:19 172.16.1.6 5688:
    Jul 16 15:34:19 172.16.1.6 5689: 15:11:30: CCSIP-SPI-CONTROL: act_sentinvite_wait_100
    Jul 16 15:34:19 172.16.1.6 5690: 15:11:30: sipSPIAddLocalContact
    Jul 16 15:34:19 172.16.1.6 5691: 15:11:30: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
    Jul 16 15:34:19 172.16.1.6 5692: 15:11:30: CCSIP-SPI-CONTROL: sip_stats_method
    Jul 16 15:34:19 172.16.1.6 5693: 15:11:30: Sent:
    Jul 16 15:34:19 172.16.1.6 5694: INVITE sip:102@172.6.1.8;user=phone;phone-context=unknown SIP/2.0^M
    Jul 16 15:34:19 172.16.1.6 5695: Via: SIP/2.0/UDP 172.16.1.6:5060^M
    Jul 16 15:34:19 172.16.1.6 5696: From: "4452257" <sip:4452257@172.16.1.6>^M
    Jul 16 15:34:19 172.16.1.6 5697: To: <sip:102@172.6.1.8;user=phone;phone-context=unknown>^M
    Jul 16 15:34:19 172.16.1.6 5698: Date: Mon, 01 Mar 1993 15:11:33 GMT^M
    Jul 16 15:34:19 172.16.1.6 5699: Call-ID: DF8B6E8D-156D11CC-80E498B6-701E77C@172.16.1.6^M
    Jul 16 15:34:19 172.16.1.6 5700: Cisco-Guid: 3750416997-359469516-2162333878-117565308^M
    Jul 16 15:34:19 172.16.1.6 5701: User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled^M
    Jul 16 15:34:19 172.16.1.6 5702: CSeq: 101 INVITE^M
    Jul 16 15:34:19 172.16.1.6 5703: Max-Forwards: 6^M
    Jul 16 15:34:19 172.16.1.6 5704: Timestamp: 730998693^M
    Jul 16 15:34:19 172.16.1.6 5705: Contact: <sip:4452257@172.16.1.6:5060;use
    Jul 16 15:34:20 172.16.1.6 5706: r=phone>^M
    Jul 16 15:34:20 172.16.1.6 5707: Expires: 300^M
    Jul 16 15:34:20 172.16.1.6 5708: Content-Type: application/sdp^M
    Jul 16 15:34:20 172.16.1.6 5709: Content-Length: 137^M
    Jul 16 15:34:20 172.16.1.6 5710: ^M
    Jul 16 15:34:20 172.16.1.6 5711: v=0^M
    Jul 16 15:34:20 172.16.1.6 5712: o=CiscoSystemsSIP-GW-UserAgent 8000 4684 IN IP4 172.16.1.6^M
    Jul 16 15:34:20 172.16.1.6 5713: s=SIP Call^M
    Jul 16 15:34:20 172.16.1.6 5714: c=IN IP4 172.16.1.6^M
    Jul 16 15:34:20 172.16.1.6 5715: t=0 0^M
    Jul 16 15:34:20 172.16.1.6 5716: m=audio 20740 RTP/AVP 18 0 4 2^M
    Jul 16 15:34:20 172.16.1.6 5717:
    J
     
  4. Ted Mittelstaedt

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    Here is the next part

    ul 16 15:34:20 172.16.1.6 5718: 15:11:31: CCSIP-SPI-CONTROL: act_sentinvite_wait_100
    Jul 16 15:34:20 172.16.1.6 5719: 15:11:31: sipSPIAddLocalContact
    Jul 16 15:34:20 172.16.1.6 5720: 15:11:31: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
    Jul 16 15:34:20 172.16.1.6 5721: 15:11:31: CCSIP-SPI-CONTROL: sip_stats_method
    Jul 16 15:34:20 172.16.1.6 5722: 15:11:31: Sent:
    Jul 16 15:34:20 172.16.1.6 5723: INVITE sip:102@172.6.1.8;user=phone;phone-context=unknown SIP/2.0^M
    Jul 16 15:34:20 172.16.1.6 5724: Via: SIP/2.0/UDP 172.16.1.6:5060^M
    Jul 16 15:34:20 172.16.1.6 5725: From: "4452257" <sip:4452257@172.16.1.6>^M
    Jul 16 15:34:20 172.16.1.6 5726: To: <sip:102@172.6.1.8;user=phone;phone-context=unknown>^M
    Jul 16 15:34:20 172.16.1.6 5727: Date: Mon, 01 Mar 1993 15:11:34 GMT^M
    Jul 16 15:34:20 172.16.1.6 5728: Call-ID: DF8B6E8D-156D11CC-80E498B6-701E77C@172.16.1.6^M
    Jul 16 15:34:20 172.16.1.6 5729: Cisco-Guid: 3750416997-359469516-2162333878-117565308^M
    Jul 16 15:34:20 172.16.1.6 5730: User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled^M
    Jul 16 15:34:20 172.16.1.6 5731: CSeq: 101 INVITE^M
    Jul 16 15:34:20 172.16.1.6 5732: Max-Forwards: 6^M
    Jul 16 15:34:20 172.16.1.6 5733: Timestamp: 730998694^M
    Jul 16 15:34:20 172.16.1.6 5734: Contact: <sip:4452257@172.16.1.6:5060;use
    Jul 16 15:34:21 172.16.1.6 5735: r=phone>^M
    Jul 16 15:34:21 172.16.1.6 5736: Expires: 300^M
    Jul 16 15:34:21 172.16.1.6 5737: Content-Type: application/sdp^M
    Jul 16 15:34:21 172.16.1.6 5738: Content-Length: 137^M
    Jul 16 15:34:21 172.16.1.6 5739: ^M
    Jul 16 15:34:21 172.16.1.6 5740: v=0^M
    Jul 16 15:34:21 172.16.1.6 5741: o=CiscoSystemsSIP-GW-UserAgent 8000 4684 IN IP4 172.16.1.6^M
    Jul 16 15:34:21 172.16.1.6 5742: s=SIP Call^M
    Jul 16 15:34:21 172.16.1.6 5743: c=IN IP4 172.16.1.6^M
    Jul 16 15:34:21 172.16.1.6 5744: t=0 0^M
    Jul 16 15:34:21 172.16.1.6 5745: m=audio 20740 RTP/AVP 18 0 4 2^M
    Jul 16 15:34:21 172.16.1.6 5746:
    Jul 16 15:34:22 172.16.1.6 5747: 15:11:33: CCSIP-SPI-CONTROL: act_sentinvite_wait_100
    Jul 16 15:34:22 172.16.1.6 5748: 15:11:33: sipSPIAddLocalContact
    Jul 16 15:34:22 172.16.1.6 5749: 15:11:33: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
    Jul 16 15:34:22 172.16.1.6 5750: 15:11:33: CCSIP-SPI-CONTROL: sip_stats_method
    Jul 16 15:34:22 172.16.1.6 5751: 15:11:33: HandleUdpSocketWrites - Send failed errno 146
    Jul 16 15:34:22 172.16.1.6 5752: 15:11:33: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060
    Jul 16 15:34:22 172.16.1.6 5753: 15:11:33: Sent:
    Jul 16 15:34:22 172.16.1.6 5754: INVITE sip:102@172.6.1.8;user=phone;phone-context=unknown SIP/2.0^M
    Jul 16 15:34:22 172.16.1.6 5755: Via: SIP/2.0/UDP 172.16.1.6:5060^M
    Jul 16 15:34:22 172.16.1.6 5756: From: "4452257" <sip:4452257@172.16.1.6>^M
    Jul 16 15:34:22 172.16.1.6 5757: To: <sip:102@172.6.1.8;user=phone;phone-context=unknown>^M
    Jul 16 15:34:22 172.16.1.6 5758: Date: Mon, 01 Mar 1993 15:11:36 GMT^M
    Jul 16 15:34:22 172.16.1.6 5759: Call-ID: DF8B6E8D-156D11CC-80E498B6-701E77C@172.16.1.6^M
    Jul 16 15:34:22 172.16.1.6 5760: Cisco-Guid: 3750416997-359469516-2162333878-117565308^M
    Jul 16 15:34:22 172.16.1.6 5761: User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled^M
    Jul 16 15:34:22 172.16.1.6 5762: CSeq: 101 INVITE^M
    Jul 16 15:34:22 172.16.1.6 5763: Max-Forwards: 6^M
    Jul 16 15:34:22 172.16.1.6 5764: Timestamp: 730998696^M
    Jul 16 15:34:22 172.16.1.6 5765: Contact: <sip:4452257@172.16.1.6:5060;use
    Jul 16 15:34:22 172.16.1.6 5766: r=phone>^M
    Jul 16 15:34:22 172.16.1.6 5767: Expires: 300^M
    Jul 16 15:34:22 172.16.1.6 5768: Content-Type: application/sdp^M
    Jul 16 15:34:22 172.16.1.6 5769: Content-Length: 137^M
    Jul 16 15:34:22 172.16.1.6 5770: ^M
    Jul 16 15:34:22 172.16.1.6 5771: v=0^M
    Jul 16 15:34:22 172.16.1.6 5772: o=CiscoSystemsSIP-GW-UserAgent 8000 4684 IN IP4 172.16.1.6^M
    Jul 16 15:34:22 172.16.1.6 5773: s=SIP Call^M
    Jul 16 15:34:22 172.16.1.6 5774: c=IN IP4 172.16.1.6^M
    Jul 16 15:34:22 172.16.1.6 5775: t=0 0^M
    Jul 16 15:34:22 172.16.1.6 5776: m=audio 20740 RTP/AVP 18 0 4 2^M
    Jul 16 15:34:22 172.16.1.6 5777:
    Jul 16 15:34:22 172.16.1.6 5778: 15:11:33: CCSIP-SPI-CONTROL: act_socket_error
    Jul 16 15:34:22 172.16.1.6 5779: 15:11:33: CCSIP-SPI-CONTROL: sipSPIInitiateCallDisconnect : Initiate call disconnect(127) for outgoing call
    Jul 16 15:34:22 172.16.1.6 5780: 15:11:33: 0x8272FD74 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)
    Jul 16 15:34:22 172.16.1.6 5781: 15:11:33: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT
    Jul 16 15:34:22 172.16.1.6 5782: 15:11:33: CCSIP-SPI-CONTROL: act_disconnecting_disconnect
    Jul 16 15:34:22 172.16.1.6 5783: 15:11:33: CCSIP-SPI-CONTROL: act_disconnecting_disconnect: Invalid condition
    Jul 16 15:34:22 172.16.1.6 5784: 15:11:33: CCSIP-SPI-CONTROL: sipSPICallCleanup
    Jul 16 15:34:22 172.16.1.6 5785: 15:11:33: sipSPIIcpifUpdate :CallState: 2 Playout: 0 DiscTime:5469316 ConnTime 0
    Jul 16 15:34:22 172.16.1.6 5786:
    Jul 16 15:34:22 172.16.1.6 5787: 15:11:33: 0x8272FD74 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE)
    Jul 16 15:34:23 172.16.1.6 5788: 15:11:33: The Call Setup Information is :
    Jul 16 15:34:23 172.16.1.6 5789: Call Control Block (CCB) : 0x8272FD74
    Jul 16 15:34:23 172.16.1.6 5790: State of The Call : STATE_DEAD
    Jul 16 15:34:23 172.16.1.6 5791: TCP Sockets Used : NO
    Jul 16 15:34:23 172.16.1.6 5792: Calling Number : 4452257
    Jul 16 15:34:23 172.16.1.6 5793: Called Number : 102
    Jul 16 15:34:23 172.16.1.6 5794: Negotiated Codec : No Codec
    Jul 16 15:34:23 172.16.1.6 5795: Negotiated Dtmf-relay : 0
    Jul 16 15:34:23 172.16.1.6 5796: Source IP Address (Media): 172.16.1.6
    Jul 16 15:34:23 172.16.1.6 5797: Source IP Port (Media): 20740
    Jul 16 15:34:23 172.16.1.6 5798: Destn IP Address (Media): 0.0.0.0
    Jul 16 15:34:23 172.16.1.6 5799: Destn IP Port (Media): 0
    Jul 16 15:34:23 172.16.1.6 5800: Destn SIP Addr:port : 172.6.1.8:5060
    Jul 16 15:34:23 172.16.1.6 5801: Destination Name : 172.6.1.8
    Jul 16 15:34:23 172.16.1.6 5802:
    Jul 16 15:34:23 172.16.1.6 5803: 15:11:33:
    Jul 16 15:34:23 172.16.1.6 5804: Disconnect Cause (CC) : 127
    Jul 16 15:34:23 172.16.1.6 5805: Disconnect Cause (SIP) : 200
    Jul 16 15:34:23 172.16.1.6 5806:
     
  5. Ted Mittelstaedt

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    And here is the next part

    Jul 16 15:34:37 172.16.1.6 5807: 15:11:47: 0x82730658 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
    Jul 16 15:34:37 172.16.1.6 5808: 15:11:47: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP
    Jul 16 15:34:37 172.16.1.6 5809: 15:11:47: CCSIP-SPI-CONTROL: act_idle_call_setup
    Jul 16 15:34:37 172.16.1.6 5810: 15:11:47: Using Voice Class Codec, tag = 1
    Jul 16 15:34:37 172.16.1.6 5811:
    Jul 16 15:34:37 172.16.1.6 5812: 15:11:47: act_idle_call_setup: preferred_codec set[0] type :g729r8 bytes: 20
    Jul 16 15:34:37 172.16.1.6 5813: 15:11:47: Queued event from SIP SPI : SIPSPI_EV_CREATE_CONNECTION
    Jul 16 15:34:37 172.16.1.6 5814: 15:11:47: 0x82730658 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_CONNECTING)
    Jul 16 15:34:37 172.16.1.6 5815: 15:11:47: 0x82730658 : State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_IDLE, SUBSTATE_CONNECTING)
    Jul 16 15:34:37 172.16.1.6 5816: 15:11:47: CCSIP-SPI-CONTROL: act_idle_connection_created
    Jul 16 15:34:37 172.16.1.6 5817: 15:11:47: CCSIP-SPI-CONTROL: act_idle_connection_created: Connid(1) created to 172.6.1.8:5060, local_port 55469
    Jul 16 15:34:37 172.16.1.6 5818: 15:11:47: sipSPIAddLocalContact
    Jul 16 15:34:37 172.16.1.6 5819: 15:11:47: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
    Jul 16 15:34:37 172.16.1.6 5820: 15:11:47: CCSIP-SPI-CONTROL: sip_stats_method
    Jul 16 15:34:37 172.16.1.6 5821: 15:11:47: 0x82730658 : State change from (STATE_IDLE, SUBSTATE_CONNECTING) to (STATE_SENT_INVITE, SUBSTATE_NONE)
    Jul 16 15:34:37 172.16.1.6 5822: 15:11:47: Sent:
    Jul 16 15:34:37 172.16.1.6 5823: INVITE sip:102@172.6.1.8;user=phone;phone-context=unknown SIP/2.0^M
    Jul 16 15:34:37 172.16.1.6 5824: Via: SIP/2.0/UDP 172.16.1.6:5060^M
    Jul 16 15:34:37 172.16.1.6 5825: From: "4452257" <sip:4452257@172.16.1.6>^M
    Jul 16 15:34:37 172.16.1.6 5826: To: <sip:102@172.6.1.8;user=phone;phone-context=unknown>^M
    Jul 16 15:34:37 172.16.1.6 5827: Date: Mon, 01 Mar 1993 15:11:51 GMT^M
    Jul 16 15:34:37 172.16.1.6 5828: Call-ID: EA4632A3-156D11CC-80E998B6-701E77C@172.16.1.6^M
    Jul 16 15:34:37 172.16.1.6 5829: Cisco-Guid: 3930429051-359469516-2162661558-117565308^M
    Jul 16 15:34:37 172.16.1.6 5830: User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled^M
    Jul 16 15:34:37 172.16.1.6 5831: CSeq: 101 INVITE^M
    Jul 16 15:34:37 172.16.1.6 5832: Max-Forwards: 6^M
    Jul 16 15:34:37 172.16.1.6 5833: Timestamp: 730998711^M
    Jul 16 15:34:37 172.16.1.6 5834: Contact: <sip:4452257@172.16.1.6:5060;use
    Jul 16 15:34:37 172.16.1.6 5835: r=phone>^M
    Jul 16 15:34:37 172.16.1.6 5836: Expires: 300^M
    Jul 16 15:34:37 172.16.1.6 5837: Content-Type: application/sdp^M
    Jul 16 15:34:37 172.16.1.6 5838: Content-Length: 137^M
    Jul 16 15:34:37 172.16.1.6 5839: ^M
    Jul 16 15:34:37 172.16.1.6 5840: v=0^M
    Jul 16 15:34:37 172.16.1.6 5841: o=CiscoSystemsSIP-GW-UserAgent 8346 2654 IN IP4 172.16.1.6^M
    Jul 16 15:34:37 172.16.1.6 5842: s=SIP Call^M
    Jul 16 15:34:37 172.16.1.6 5843: c=IN IP4 172.16.1.6^M
    Jul 16 15:34:37 172.16.1.6 5844: t=0 0^M
    Jul 16 15:34:37 172.16.1.6 5845: m=audio 20882 RTP/AVP 18 0 4 2^M
    Jul 16 15:34:37 172.16.1.6 5846:
    Jul 16 15:34:37 172.16.1.6 5847: 15:11:48: CCSIP-SPI-CONTROL: act_sentinvite_wait_100
    Jul 16 15:34:37 172.16.1.6 5848: 15:11:48: sipSPIAddLocalContact
    Jul 16 15:34:37 172.16.1.6 5849: 15:11:48: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
    Jul 16 15:34:37 172.16.1.6 5850: 15:11:48: CCSIP-SPI-CONTROL: sip_stats_method
    Jul 16 15:34:37 172.16.1.6 5851: 15:11:48: Sent:
    Jul 16 15:34:37 172.16.1.6 5852: INVITE sip:102@172.6.1.8;user=phone;phone-context=unknown SIP/2.0^M
    Jul 16 15:34:37 172.16.1.6 5853: Via: SIP/2.0/UDP 172.16.1.6:5060^M
    Jul 16 15:34:37 172.16.1.6 5854: From: "4452257" <sip:4452257@172.16.1.6>^M
    Jul 16 15:34:37 172.16.1.6 5855: To: <sip:102@172.6.1.8;user=phone;phone-context=unknown>^M
    Jul 16 15:34:37 172.16.1.6 5856: Date: Mon, 01 Mar 1993 15:11:51 GMT^M
    Jul 16 15:34:37 172.16.1.6 5857: Call-ID: EA4632A3-156D11CC-80E998B6-701E77C@172.16.1.6^M
    Jul 16 15:34:37 172.16.1.6 5858: Cisco-Guid: 3930429051-359469516-2162661558-117565308^M
    Jul 16 15:34:37 172.16.1.6 5859: User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled^M
    Jul 16 15:34:37 172.16.1.6 5860: CSeq: 101 INVITE^M
    Jul 16 15:34:37 172.16.1.6 5861: Max-Forwards: 6^M
    Jul 16 15:34:37 172.16.1.6 5862: Timestamp: 730998711^M
    Jul 16 15:34:37 172.16.1.6 5863: Contact: <sip:4452257@172.16.1.6:5060;use
    Jul 16 15:34:38 172.16.1.6 5864: r=phone>^M
    Jul 16 15:34:38 172.16.1.6 5865: Expires: 300^M
    Jul 16 15:34:38 172.16.1.6 5866: Content-Type: application/sdp^M
    Jul 16 15:34:38 172.16.1.6 5867: Content-Length: 137^M
    Jul 16 15:34:38 172.16.1.6 5868: ^M
    Jul 16 15:34:38 172.16.1.6 5869: v=0^M
    Jul 16 15:34:38 172.16.1.6 5870: o=CiscoSystemsSIP-GW-UserAgent 8346 2654 IN IP4 172.16.1.6^M
    Jul 16 15:34:38 172.16.1.6 5871: s=SIP Call^M
    Jul 16 15:34:38 172.16.1.6 5872: c=IN IP4 172.16.1.6^M
    Jul 16 15:34:38 172.16.1.6 5873: t=0 0^M
    Jul 16 15:34:38 172.16.1.6 5874: m=audio 20882 RTP/AVP 18 0 4 2^M
    Jul 16 15:34:38 172.16.1.6 5875:
    Jul 16 15:34:38 172.16.1.6 5876: 15:11:49: CCSIP-SPI-CONTROL: act_sentinvite_wait_100
    Jul 16 15:34:38 172.16.1.6 5877: 15:11:49: sipSPIAddLocalContact
    Jul 16 15:34:38 172.16.1.6 5878: 15:11:49: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
    Jul 16 15:34:38 172.16.1.6 5879: 15:11:49: CCSIP-SPI-CONTROL: sip_stats_method
    Jul 16 15:34:38 172.16.1.6 5880: 15:11:49: Sent:
    Jul 16 15:34:38 172.16.1.6 5881: INVITE sip:102@172.6.1.8;user=phone;phone-context=unknown SIP/2.0^M
    Jul 16 15:34:38 172.16.1.6 5882: Via: SIP/2.0/UDP 172.16.1.6:5060^M
    Jul 16 15:34:38 172.16.1.6 5883: From: "4452257" <sip:4452257@172.16.1.6>^M
    Jul 16 15:34:38 172.16.1.6 5884: To: <sip:102@172.6.1.8;user=phone;phone-context=unknown>^M
    Jul 16 15:34:38 172.16.1.6 5885: Date: Mon, 01 Mar 1993 15:11:52 GMT^M
    Jul 16 15:34:38 172.16.1.6 5886: Call-ID: EA4632A3-156D11CC-80E998B6-701E77C@172.16.1.6^M
    Jul 16 15:34:38 172.16.1.6 5887: Cisco-Guid: 3930429051-359469516-2162661558-117565308^M
    Jul 16 15:34:38 172.16.1.6 5888: User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled^M
    Jul 16 15:34:38 172.16.1.6 5889: CSeq: 101 INVITE^M
    Jul 16 15:34:38 172.16.1.6 5890: Max-Forwards: 6^M
    Jul 16 15:34:38 172.16.1.6 5891: Timestamp: 730998712^M
    Jul 16 15:34:38 172.16.1.6 5892: Contact: <sip:4452257@172.16.1.6:5060;use
    Jul 16 15:34:39 172.16.1.6 5893: r=phone>^M
    Jul 16 15:34:39 172.16.1.6 5894: Expires: 300^M
    Jul 16 15:34:39 172.16.1.6 5895: Content-Type: application/sdp^M
    Jul 16 15:34:39 172.16.1.6 5896: Content-Length: 137^M
    Jul 16 15:34:39 172.16.1.6 5897: ^M
    Jul 16 15:34:39 172.16.1.6 5898: v=0^M
    Jul 16 15:34:39 172.16.1.6 5899: o=CiscoSystemsSIP-GW-UserAgent 8346 2654 IN IP4 172.16.1.6^M
    Jul 16 15:34:39 172.16.1.6 5900: s=SIP Call^M
    Jul 16 15:34:39 172.16.1.6 5901: c=IN IP4 172.16.1.6^M
    Jul 16 15:34:39 172.16.1.6 5902: t=0 0^M
    Jul 16 15:34:39 172.16.1.6 5903: m=audio 20882 RTP/AVP 18 0 4 2^M
    Jul 16 15:34:39 172.16.1.6 5904:
     
  6. Ted Mittelstaedt

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    And here is the last of it

    Jul 16 15:34:40 172.16.1.6 5905: 15:11:51: CCSIP-SPI-CONTROL: act_sentinvite_wait_100
    Jul 16 15:34:40 172.16.1.6 5906: 15:11:51: sipSPIAddLocalContact
    Jul 16 15:34:40 172.16.1.6 5907: 15:11:51: Queued event from SIP SPI : SIPSPI_EV_SEND_MESSAGE
    Jul 16 15:34:40 172.16.1.6 5908: 15:11:51: CCSIP-SPI-CONTROL: sip_stats_method
    Jul 16 15:34:40 172.16.1.6 5909: 15:11:51: HandleUdpSocketWrites - Send failed errno 146
    Jul 16 15:34:40 172.16.1.6 5910: 15:11:51: udpsock_close_connect: Socket fd: 1 closed for connid 1 with remote port: 5060
    Jul 16 15:34:40 172.16.1.6 5911: 15:11:51: Sent:
    Jul 16 15:34:40 172.16.1.6 5912: INVITE sip:102@172.6.1.8;user=phone;phone-context=unknown SIP/2.0^M
    Jul 16 15:34:40 172.16.1.6 5913: Via: SIP/2.0/UDP 172.16.1.6:5060^M
    Jul 16 15:34:40 172.16.1.6 5914: From: "4452257" <sip:4452257@172.16.1.6>^M
    Jul 16 15:34:40 172.16.1.6 5915: To: <sip:102@172.6.1.8;user=phone;phone-context=unknown>^M
    Jul 16 15:34:40 172.16.1.6 5916: Date: Mon, 01 Mar 1993 15:11:54 GMT^M
    Jul 16 15:34:40 172.16.1.6 5917: Call-ID: EA4632A3-156D11CC-80E998B6-701E77C@172.16.1.6^M
    Jul 16 15:34:40 172.16.1.6 5918: Cisco-Guid: 3930429051-359469516-2162661558-117565308^M
    Jul 16 15:34:40 172.16.1.6 5919: User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled^M
    Jul 16 15:34:40 172.16.1.6 5920: CSeq: 101 INVITE^M
    Jul 16 15:34:40 172.16.1.6 5921: Max-Forwards: 6^M
    Jul 16 15:34:40 172.16.1.6 5922: Timestamp: 730998714^M
    Jul 16 15:34:40 172.16.1.6 5923: Contact: <sip:4452257@172.16.1.6:5060;use
    Jul 16 15:34:40 172.16.1.6 5924: r=phone>^M
    Jul 16 15:34:40 172.16.1.6 5925: Expires: 300^M
    Jul 16 15:34:40 172.16.1.6 5926: Content-Type: application/sdp^M
    Jul 16 15:34:40 172.16.1.6 5927: Content-Length: 137^M
    Jul 16 15:34:40 172.16.1.6 5928: ^M
    Jul 16 15:34:40 172.16.1.6 5929: v=0^M
    Jul 16 15:34:40 172.16.1.6 5930: o=CiscoSystemsSIP-GW-UserAgent 8346 2654 IN IP4 172.16.1.6^M
    Jul 16 15:34:40 172.16.1.6 5931: s=SIP Call^M
    Jul 16 15:34:40 172.16.1.6 5932: c=IN IP4 172.16.1.6^M
    Jul 16 15:34:40 172.16.1.6 5933: t=0 0^M
    Jul 16 15:34:40 172.16.1.6 5934: m=audio 20882 RTP/AVP 18 0 4 2^M
    Jul 16 15:34:40 172.16.1.6 5935:
    Jul 16 15:34:40 172.16.1.6 5936: 15:11:51: CCSIP-SPI-CONTROL: act_socket_error
    Jul 16 15:34:40 172.16.1.6 5937: 15:11:51: CCSIP-SPI-CONTROL: sipSPIInitiateCallDisconnect : Initiate call disconnect(127) for outgoing call
    Jul 16 15:34:40 172.16.1.6 5938: 15:11:51: 0x82730658 : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)
    Jul 16 15:34:40 172.16.1.6 5939: 15:11:51: Queued event From SIP SPI to CCAPI/DNS : SIPSPI_EV_CC_CALL_DISCONNECT
    Jul 16 15:34:40 172.16.1.6 5940: 15:11:51: CCSIP-SPI-CONTROL: act_disconnecting_disconnect
    Jul 16 15:34:40 172.16.1.6 5941: 15:11:51: CCSIP-SPI-CONTROL: act_disconnecting_disconnect: Invalid condition
    Jul 16 15:34:40 172.16.1.6 5942: 15:11:51: CCSIP-SPI-CONTROL: sipSPICallCleanup
    Jul 16 15:34:40 172.16.1.6 5943: 15:11:51: sipSPIIcpifUpdate :CallState: 2 Playout: 0 DiscTime:5471115 ConnTime 0
    Jul 16 15:34:40 172.16.1.6 5944:
    Jul 16 15:34:40 172.16.1.6 5945: 15:11:51: 0x82730658 : State change from (STATE_DISCONNECTING, SUBSTATE_NONE) to (STATE_DEAD, SUBSTATE_NONE)
    Jul 16 15:34:41 172.16.1.6 5946: 15:11:51: The Call Setup Information is :
    Jul 16 15:34:41 172.16.1.6 5947: Call Control Block (CCB) : 0x82730658
    Jul 16 15:34:41 172.16.1.6 5948: State of The Call : STATE_DEAD
    Jul 16 15:34:41 172.16.1.6 5949: TCP Sockets Used : NO
    Jul 16 15:34:41 172.16.1.6 5950: Calling Number : 4452257
    Jul 16 15:34:41 172.16.1.6 5951: Called Number : 102
    Jul 16 15:34:41 172.16.1.6 5952: Negotiated Codec : No Codec
    Jul 16 15:34:41 172.16.1.6 5953: Negotiated Dtmf-relay : 0
    Jul 16 15:34:41 172.16.1.6 5954: Source IP Address (Media): 172.16.1.6
    Jul 16 15:34:41 172.16.1.6 5955: Source IP Port (Media): 20882
    Jul 16 15:34:41 172.16.1.6 5956: Destn IP Address (Media): 0.0.0.0
    Jul 16 15:34:41 172.16.1.6 5957: Destn IP Port (Media): 0
    Jul 16 15:34:41 172.16.1.6 5958: Destn SIP Addr:port : 172.6.1.8:5060
    Jul 16 15:34:41 172.16.1.6 5959: Destination Name : 172.6.1.8
    Jul 16 15:34:41 172.16.1.6 5960:
    Jul 16 15:34:41 172.16.1.6 5961: 15:11:51:
    Jul 16 15:34:41 172.16.1.6 5962: Disconnect Cause (CC) : 127
    Jul 16 15:34:41 172.16.1.6 5963: Disconnect Cause (SIP) : 200
    Jul 16 15:34:41 172.16.1.6 5964:
     
  7. Ted Mittelstaedt

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    by the way that is:

    Jul 16 15:34:41 172.16.1.6 5958: Destn SIP A d d r : P o r t : 172.6.1.8:5060

    i'll do my own obfuscation, thank you very much!!! (I mean, really?!?!)
     
  8. leejor

    leejor Well-Known Member

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    Yeah, that automatic changing "colon p" to an emoji is a bit annoying on the latest forum incarnation.
    You said that there is no entry, yet there appears to be, so which is it?

    Is this the incoming call being directed to extension 102? Normally, the gateway will send to the trunk number, such as 90001, 90002, then 3CX routes based on how you have that trunk provisioned. When you created the trunk in 3CX it is normally assigned a 5 digit trunk number, that increments as new trunks are added. It used to be 9xxxxx, I think the latest versions start with 1xxxxx.. Try re-doing the configuration file (and the 3CX trunk datafill) that you created using something other than an extension number (102).

    If you have a read through this, https://www.3cx.com/community/threads/provisioning-a-linksys-spa-3102-gateway.34293/ looking at the screenshots, you might get an idea of what 3CX wants to see. Yes, it's a SPA-3102, but other than being a single gateway, is going to communicate to 3CX in the same manner. It's a mater of translating those settings over to a config file for your device..

    As I said, 3CX simply forwards the call to the trunk number in 3CX.
     
    #8 leejor, Jul 17, 2017
    Last edited: Jul 17, 2017
  9. Ted Mittelstaedt

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    Yes, that is an attempt to direct to extension 102 I was experimenting with. I have tried sending the call to the trunk number and it doesn't work either. Basically I just get a ring-no-answer.

    I deleted the trunk config then re-entered it. Same issue. Here's what I just did:

    In SIP trunks, clicked on Add Gateway selected Generic Gateway Device. Put in 172.16.1.6 as gateway IP, changed Type of Authentication to do not require - IP based. Saved that. Then in Inbound Rules, Add DID rule, named Default, DID/DDI 90001, Extension 000 Operator

    Then in Outbound rules I add a 9 rule that selects the generic gateway device and drops the 9 digit.
    Same problem, can call out, cannot call in.

    Then here is the relevant parts of the Cisco config:


    voice service voip
    !
    voice class codec 1
    codec preference 1 g729r8
    codec preference 2 g711ulaw
    codec preference 3 g723r63 bytes 96
    codec preference 4 g726r16 bytes 80


    voice-port 1/0/0
    no battery-reversal
    timing hookflash-out 50
    connection plar opx 90001
    description trunk 445-2255
    station-id number 4452257
    !
    voice-port 1/0/1
    no battery-reversal
    timing hookflash-out 50
    connection plar opx 90001
    description trunk 445-2257
    station-id number 4452257
    !
    voice-port 1/1/0
    no battery-reversal
    timing hookflash-out 50
    connection plar opx 90001
    description trunk 445-2256
    station-id number 4452256
    !

    dial-peer voice 1 voip
    preference 1
    destination-pattern 90001
    voice-class codec 1
    session protocol sipv2
    session target sip-server
    no vad

    sip-ua
    retry invite 3
    retry response 3
    retry bye 3
    retry cancel 3
    timers expires 300000
    sip-server ipv4:172.6.1.8:5060
    !
     
  10. leejor

    leejor Well-Known Member

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    OK, what do the 3CX logs look like for an outgoing call? Does the trunk group show as registered? Not having used that CIsco device before, I'm not familiar with all of the settings involved.

    One critical setting with other gateways is the need for "one stage dialling", for calls towards 3CX. Not sure if that shows in some of the other settings. Since you had outgoing working previously, (and saw logs of incoming calls), in theory, you should have been able to leave all settings untouched EXCEPT for the trunk group number, at both ends.
     
  11. Ted Mittelstaedt

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    The 3CX log just shows a plain old outgoing call like nothing doing. There's no detail whatsoever it just shows a call was made successfully. (in other words - the 3CX logs in the current product suck)

    The trunk shows as green but the registration info is blank because as I said - Cisco IOS 12.2 does not support userID/password authentication for SIP calls. Thus, if you define any trunk (no matter what device is used) as NOT requiring authentication then it will always show green and it won't show as "registered" because it ISN'T registered - only trunks that use authentication get "registered"

    Here's an example of what I'm referring to:

    https://www.voip-info.org/wiki-Asterisk+cisco+FXO

    It may possibly come as a surprise to some VoIP users but userID/password authentication (registration) is NOT a required component of SIP

    I am beginning to think this is a bug in version 15 of 3CX. That is, the programmers defined "no authentication" in the webinterface for defining generic gateways - but when you select that, it doesn't actually turn off authentication.

    The reason the logging stinks is that it is only showing incoming SIP connections that actually successfully complete userID/password authentication but are blocked by policy. In other words when you send a SIP connection in with a userID of 1000@1.1.1.1 and a password of 1000 it gets authenticated into 3CX then 3CX looks in it's internal policy to see if that's allowed - and if it is not then it drops the call and THEN make a log entry of "Discarded message, because of blocked User-Agent"

    Which is why if you expose 3CX to the Internet you need to use good passwords on your SIP extensions.

    What is really needed here is to be able to directly access whatever passes for sip.conf now and add in the permissions for an unauthenticated SIP trunk connection from a specific IP address, since the webinterface has failed to do this (even though it says it did) or better yet have the capability of debug logging available to look at ALL incoming SIP connections even the ones that don't authenticate in.

    Unfortunately, the original Cisco FXO cards only worked in the 2600 and they didn't have a lot of IOS versions out there past 12.2 that support the hardware and add-in cards for the first gen VICs.
     
    #11 Ted Mittelstaedt, Jul 17, 2017
    Last edited: Jul 17, 2017
  12. Ted Mittelstaedt

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    Just update on this:

    I had the wrong sip server IP address in the Cisco config - 172.6.1.8 instead of what it should have been, 172.16.1.8

    However, now it is connecting to 3CX but I got an error saying "no proxy" and IOS 12.2 does not support sip proxy. IPS version 12.3 does so I booted the router off of that and tried again - this time I am getting a conection, getting a response from 3CX, the logs look great - except no call log no nothing in the 3CX. Unfortunately, IOS 12.3 is the last IOS version that is supported on the regular Cisco 2620. IOS 12.3 does not support SIP trunk registration - although the Generic Gateway config in 3CX allows you to select no registration - so I'm at a loss now.

    I think I will do a new post that details everything from the get-go with a new log from IOS 12.3
     
  13. 12494

    12494 Member

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    So what is your objective? Are you wanting to spend a lot of time to make your device work or do you want to know if 3CX works with supported devices?
    I can tell you that 3CX works well, and easily with no wasted time, with Patton FXOs. I think you will find that my experience is not unusual. Personally I would rather a quick painless solution rather than the alternative.
    Good luck.
    Allen
     
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  14. Ted Mittelstaedt

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    [edited by Vali_3CX]

    If I wanted an easy solution I could pay the 30 pieces of silver a month to 3CX for their cloud solution. Why bother with the Patton hardware?

    The Cisco VIC FXO v1 solution is likely not going to be as good as a Patton box since allegedly it doesn't support Caller ID. So there are reasons that someone who doesn't have all the gear would have little incentive to run out and buy it all instead of a Patton. Certainly I wouldn't advise doing that. But for someone who has never used 3CX before and who has the gateway gear, trying to get it to work is free. It's also immediate. To get the Patton device I have to buy it then wait for it to arrive.

    And on top of that - I'm so close I can taste it. It works for outbound calls, it should work for inbound calls.

    I have to assume that since 3CX has the Generic FXO Gateway device they have no objections to someone plugging anyone's FXO gateway into it. I would also assume that they and future customers of theirs would find value in either being told "no this hardware will never work" or "yes this works but here's the drawbacks" or "yes this works perfectly"
     
    #14 Ted Mittelstaedt, Jul 21, 2017
    Last edited by a moderator: Jul 21, 2017
  15. 12494

    12494 Member

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    Hi Ted,
    Good answer. And I get that and respect it. My background is mathematics before I got involved in technology. Both areas give me plenty of opportunities to solve problems, which clearly, is also what you also like to do. With me though, over the years, my emphasis has shifted from cost to benefit. In any event, thanks for the reply and good luck.
    Regards,
    A.
    P.S. Interesting to see that your post/reply has been edited by someone else. However the email notification I received has a leading paragraph that is missing from the online post so presumably I can see what was edited out. And it looks pretty benign to me. Oh well.
     
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  16. leejor

    leejor Well-Known Member

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    Wireshark should show what is being sent to 3CX and along with the incoming call Activity log would point to what needs to be "adjusted".
     
  17. Ted Mittelstaedt

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    Please refer to my newest post on this titled "Is trunk authorization by IP address broken" for a better discussion on this
     
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