cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
426
Views
0
Helpful
6
Replies

Call between two routers does not complete

Colin Higgins
Level 2
Level 2

We have an Avaya PBX that needs to talk to an analog phone of a FXS port on a remote 2851 voice router.

The Avaya is hanging off a FXO port on a 2851 router with a four port FXO/FXS card. The routers talk to each other via IP through the corporate network, and no firewalls are involved.

When I try to call the remote phone through the PBX, it rings, the person picks up, but they don't hear anything. On my side it keeps ringing.

When I do a show dial-peer voice summary, I see the OPER PREFIX as up, the proper dest-pattern, port etc. When someone picks up the remote phone, I see voice port 2/0 go off-hook. However, if I try to look at the RTP stream, it is inactive (obviously).

Here is the config for the router connected to the Avaya PBX

voice-card 0

no dspfarm

interface GigabitEthernet0/0

description Uplink to Sublevel Voice_SW_3550

ip address 172.25.199.5 255.255.255.0

duplex auto

speed auto

!

interface GigabitEthernet0/1

no ip address

shutdown

duplex auto

speed auto

!

ip classless

ip route 0.0.0.0 0.0.0.0 172.25.199.1

!

!

no ip http server

no ip http secure-server

!

!

!

!

control-plane

!

!

!

voice-port 2/0/0

!

voice-port 2/0/1

!

voice-port 2/0/2

!

voice-port 2/0/3

!

voice-port 2/0/4

!

voice-port 2/0/5

!

voice-port 2/0/6

!

voice-port 2/0/7

!

voice-port 2/0/8

!

voice-port 2/0/9

!

voice-port 2/0/10

!

voice-port 2/0/11

!

voice-port 2/0/12

!

voice-port 2/0/13

!

voice-port 2/0/14

!

voice-port 2/0/15

!

voice-port 2/0/16

!

voice-port 2/0/17

!

voice-port 2/0/18

!

voice-port 2/0/20

!

voice-port 2/0/21

!

voice-port 2/0/22

!

voice-port 2/0/23

connection plar 2099

!

!

!

!

!

dial-peer voice 2099 voip

destination-pattern 2099

session target ipv4:172.25.76.5

incoming called-number .

!

dial-peer voice 100 pots

destination-pattern 91[2-9]..[2-9]......

incoming called-number .

port 2/0/23

forward-digits all

!

dial-peer voice 101 pots

destination-pattern [0-8]...

incoming called-number .

port 2/0/23

----

And the remote Router with the analog phone hanging off port 2/0

interface GigabitEthernet0/0

description Uplink to 00-Central-3750-48-I

ip address 172.25.76.5 255.255.255.0

duplex auto

speed auto

!

interface GigabitEthernet0/1

no ip address

shutdown

duplex auto

speed auto

!

ip classless

ip route 0.0.0.0 0.0.0.0 172.25.76.1

!

!

no ip http server

no ip http secure-server

!

!

!

!

control-plane

!

!

!

voice-port 2/0/0

!

voice-port 2/0/1

!

voice-port 2/0/2

!

voice-port 2/0/3

!

voice-port 2/0/4

!

voice-port 2/0/5

!

voice-port 2/0/6

!

voice-port 2/0/7

!

voice-port 2/0/8

!

voice-port 2/0/9

!

voice-port 2/0/10

!

voice-port 2/0/11

!

voice-port 2/0/12

!

voice-port 2/0/13

!

voice-port 2/0/14

!

voice-port 2/0/15

!

!

!

!

!

dial-peer voice 2099 pots

destination-pattern 2099

incoming called-number .

port 2/0/0

!

dial-peer voice 100 voip

destination-pattern 91[2-9]..[2-9]......

session target ipv4:172.25.199.5

incoming called-number 2099

!

dial-peer voice 101 voip

destination-pattern [0-8]...

session target ipv4:172.25.199.5

incoming called-number 2099

--

As I am not familiar with Avaya, and voice is not my main area, any advice would be great. Thanks

1 Accepted Solution

Accepted Solutions

Hope you enabled all the debugs I requested. I don't see H245 negotiation happening. Could you try disabling h245 tunneling on BOTH routers

voice service voice

h323

h245 tunnel disable    <--- hidden command so ? will not work

Also add "voice call send-alert" in global config.

Regards,

Mohit Singh

View solution in original post

6 Replies 6

MOHIT SINGH
Level 1
Level 1

In FXO port instead of connection plar use "connection plar opx" because your phone is located in remote location. Also try configuring below command on both router

voice rtp send-recv

Regards,

Mohit Singh

I put

connection plar opx 2099

on the FXO port leading to the Avaya PBX

and put voice rtp send-recv

on both routers, but I am getting the same behavior. I can call the remot phone, it rings, but when the user answers, they hear nothing, and I hear ringing.

please provide below mentioned debugs from both routers. make sure you collect them together

debug voip ccapi inout

debug vpm signal

debug h225 asn1

debug h245 asn1

Regards,

Mohit Singh

This is fromt the remote router

Jan 22 21:15:56 GMT: //317/3784296381A1/CCAPI/cc_api_call_proceeding:

   Interface=0x45A473AC, Progress Indication=NULL(0)

Jan 22 21:15:56 GMT: H225.0 INCOMING ENCODE BUFFER::= 28002B80060008914A00042800B500001240013C050100003784296382E111E381A39F3FEC44012E05800180018010A00180120140B50000120B60011000011E041E028188

Jan 22 21:15:56 GMT:

Jan 22 21:15:56 GMT: H225.0 INCOMING PDU ::=

value H323_UserInformation ::=

    {

      h323-uu-pdu

      {

        h323-message-body progress :

        {

          protocolIdentifier { 0 0 8 2250 0 4 }

          destinationInfo

          {

            vendor

            {

              vendor

              {

                t35CountryCode 181

                t35Extension 0

                manufacturerCode 18

              }

            }

            gateway

            {

              protocol

              {

                voice :

                {

                  supportedPrefixes

                  {

                  }

                }

              }

            }

            mc FALSE

            undefinedNode FALSE

          }

          callIdentifier

          {

            guid '3784296382E111E381A39F3FEC44012E'H

          }

          multipleCalls TRUE

          maintainConnection TRUE

        }

        h245Tunneling TRUE

        nonStandardControl

        {

          {

            nonStandardIdentifier h221NonStandard :

            {

              t35CountryCode 181

              t35Extension 0

              manufacturerCode 18

            }

            data '60011000011E041E028188'H

          }

        }

      }

    }

Jan 22 21:15:56 GMT: H225 NONSTD INCOMING ENCODE BUFFER::= 60011000011E041E028188

Jan 22 21:15:56 GMT:

Jan 22 21:15:56 GMT: H225 NONSTD INCOMING PDU ::=

value H323_UU_NonStdInfo ::=

    {

      version 16

      protoParam qsigNonStdInfo :

      {

        iei 30

        rawMesg '1E028188'H

      }

    }

Jan 22 21:15:56 GMT: //317/3784296381A1/CCAPI/cc_api_call_cut_progress:

   Interface=0x45A473AC, Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1),

   Cause Value=0

Jan 22 21:15:56 GMT: //317/3784296381A1/CCAPI/cc_api_call_cut_progress:

   Call Entry(Responsed=TRUE)

Jan 22 21:15:56 GMT: //316/3784296381A1/CCAPI/ccCallCutProgress:

   Progress Indication=INBAND(8), Signal Indication=SIGNAL RINGBACK(1), Cause Value=0

   Voice Call Send Alert=FALSE, Call Entry(AlertSent=FALSE)

Jan 22 21:15:56 GMT: //316/3784296381A1/CCAPI/ccCallCutProgress:

   Call Entry(Responsed=TRUE)

Jan 22 21:15:56 GMT: //316/3784296381A1/CCAPI/ccGenerateToneInfo:

   Stop Tone On Digit=FALSE, Tone=Null,

   Tone Direction=Network, Params=0x0, Call Id=316

Jan 22 21:15:56 GMT: //316/xxxxxxxxxxxx/CCAPI/ccConferenceCreate:

   (confID=0x44E69794, callID1=0x13C, callID2=0x13D, tag=0x0)

Jan 22 21:15:56 GMT: //316/3784296381A1/CCAPI/ccConferenceCreate:

   Conference Id=0x44E69794, Call Id1=316, Call Id2=317, Tag=0x0

Jan 22 21:15:56 GMT: htsp_call_bridged invoked

Jan 22 21:15:56 GMT: //316/xxxxxxxxxxxx/CCAPI/cc_api_bridge_done:

   Conference Id=0x65, Source Interface=0x45E7B524, Source Call Id=316,

   Destination Call Id=317, Disposition=0x0, Tag=0xFFFFFFFF

Jan 22 21:15:56 GMT: //317/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:

Jan 22 21:15:56 GMT: cc_api_get_xcode_stream : 4160

Jan 22 21:15:56 GMT: //317/xxxxxxxxxxxx/CCAPI/cc_api_bridge_done:

   Conference Id=0x65, Source Interface=0x45A0A0A4, Source Call Id=317,

   Destination Call Id=316, Disposition=0x0, Tag=0x0

Jan 22 21:15:56 GMT: //316/3784296381A1/CCAPI/cc_generic_bridge_done:

   Conference Id=0x65, Source Interface=0x45A0A0A4, Source Call Id=317,

   Destination Call Id=316, Disposition=0x0, Tag=0x0

Jan 22 21:15:56 GMT: //316/3784296381A1/CCAPI/ccConferenceCreate:

   Call Entry(Conference Id=0x65, Destination Call Id=317)

Jan 22 21:15:56 GMT: //317/3784296381A1/CCAPI/ccConferenceCreate:

   Call Entry(Conference Id=0x65, Destination Call Id=316)htsp_progress_notify

Jan 22 21:15:56 GMT: //316/3784296381A1/CCAPI/cc_api_caps_ind:

   Destination Interface=0x45A0A0A4, Destination Call Id=317, Source Call Id=316,

   Caps(Codec=0x1, Fax Rate=0x1, Vad=0x1,

   Modem=0x2, Codec Bytes=20, Signal Type=3)

Jan 22 21:15:56 GMT: //316/3784296381A1/CCAPI/cc_api_caps_ind:

   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),

   Playout Max=250(ms), Fax Nom=300(ms))

Jan 22 21:15:56 GMT: //316/3784296381A1/CCAPI/cc_process_notify_bridge_done:

   Conference Id=0x65, Call Id1=316, Call Id2=317

Jan 22 21:15:56 GMT: //317/3784296381A1/CCAPI/cc_api_caps_ind:

   Destination Interface=0x45E7B524, Destination Call Id=316, Source Call Id=317,

   Caps(Codec=0x4, Fax Rate=0x2, Vad=0x2,

   Modem=0x0, Codec Bytes=20, Signal Type=2)

Jan 22 21:15:56 GMT: //317/3784296381A1/CCAPI/cc_api_caps_ind:

   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),

   Playout Max=250(ms), Fax Nom=300(ms))

Jan 22 21:15:56 GMT: //317/3784296381A1/CCAPI/cc_api_caps_ack:

   Destination Interface=0x45E7B524, Destination Call Id=316, Source Call Id=317,

   Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),

   Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=9694)

Jan 22 21:15:56 GMT: //316/3784296381A1/CCAPI/cc_api_caps_ack:

   Destination Interface=0x45A0A0A4, Destination Call Id=317, Source Call Id=316,

   Caps(Codec=g729r8(0x4), Fax Rate=FAX_RATE_VOICE(0x2), Vad=ON(0x2),

   Modem=OFF(0x0), Codec Bytes=20, Signal Type=2, Seq Num Start=9694)

Jan 22 21:15:56 GMT: //316/3784296381A1/CCAPI/cc_api_voice_mode_event:

   Call Id=316

Jan 22 21:15:56 GMT: //316/3784296381A1/CCAPI/cc_api_voice_mode_event:

   Call Entry(Context=0x4639BD18)

Jan 22 21:15:56 GMT: htsp_process_event: [2/0/23, FXOLS_PROCEEDING, E_HTSP_VOICE_CUT_THROUGH]fxols_proc_voice

Jan 22 21:16:02 GMT: htsp_process_event: [2/0/23, FXOLS_PROCEEDING, E_HTSP_EVENT_TIMER]fxols_disc_confirm

Jan 22 21:16:02 GMT: htsp_timer_stop

Jan 22 21:16:02 GMT: htsp_timer_stop2

Jan 22 21:16:02 GMT: htsp_timer_stop3

Jan 22 21:16:02 GMT: //316/3784296381A1/CCAPI/cc_api_call_disconnected:

   Cause Value=16, Interface=0x45E7B524, Call Id=316

Jan 22 21:16:02 GMT: //316/3784296381A1/CCAPI/cc_api_call_disconnected:

   Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)

Jan 22 21:16:02 GMT: //316/3784296381A1/CCAPI/ccConferenceDestroy:

   Conference Id=0x65, Tag=0x0

Jan 22 21:16:02 GMT: //316/xxxxxxxxxxxx/CCAPI/cc_api_bridge_drop_done:

   Conference Id=0x65, Source Interface=0x45E7B524, Source Call Id=316,

   Destination Call Id=317, Disposition=0x0, Tag=0x0

Jan 22 21:16:02 GMT: //317/xxxxxxxxxxxx/CCAPI/cc_api_bridge_drop_done:

   Conference Id=0x65, Source Interface=0x45A0A0A4, Source Call Id=317,

   Destination Call Id=316, Disposition=0x0, Tag=0x0

Jan 22 21:16:02 GMT: //316/3784296381A1/CCAPI/cc_generic_bridge_done:

   Conference Id=0x65, Source Interface=0x45A0A0A4, Source Call Id=317,

   Destination Call Id=316, Disposition=0x0, Tag=0x0

Jan 22 21:16:02 GMT: //317/3784296381A1/CCAPI/ccCallDisconnect:

   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)

Jan 22 21:16:02 GMT: //317/3784296381A1/CCAPI/ccCallDisconnect:

   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)

Jan 22 21:16:02 GMT: //317/3784296381A1/CCAPI/cc_api_get_transfer_info:

   Transfer Number Is Null

Jan 22 21:16:02 GMT: //316/3784296381A1/CCAPI/ccCallDisconnect:

   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=16)

Jan 22 21:16:02 GMT: //316/3784296381A1/CCAPI/ccCallDisconnect:

  Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)

Jan 22 21:16:02 GMT: //316/3784296381A1/CCAPI/cc_api_get_transfer_info:

   Transfer Number Is Null

Jan 22 21:16:02 GMT: H225.0 OUTGOING PDU ::=

value H323_UserInformation ::=

    {

      h323-uu-pdu

      {

        h323-message-body releaseComplete :

        {

          protocolIdentifier { 0 0 8 2250 0 4 }

          callIdentifier

          {

            guid '3784296382E111E381A39F3FEC44012E'H

          }

        }

        h245Tunneling TRUE

      }

    }

Jan 22 21:16:02 GMT: H225.0 OUTGOING ENCODE BUFFER::= 2580060008914A0004110011003784296382E111E381A39F3FEC44012E10800180

Jan 22 21:16:02 GMT:

Jan 22 21:16:02 GMT: H225.0 INCOMING ENCODE BUFFER::= 2580060008914A0004110011003784296382E111E381A39F3FEC44012E10800180

Jan 22 21:16:02 GMT:

Jan 22 21:16:02 GMT: H225.0 INCOMING PDU ::=

value H323_UserInformation ::=

    {

      h323-uu-pdu

      {

        h323-message-body releaseComplete :

        {

          protocolIdentifier { 0 0 8 2250 0 4 }

          callIdentifier

          {

            guid '3784296382E111E381A39F3FEC44012E'H

          }

        }

        h245Tunneling TRUE

      }

    }

Jan 22 21:16:02 GMT: //317/3784296381A1/CCAPI/cc_api_call_disconnect_done:

   Disposition=0, Interface=0x45A473AC, Tag=0x0, Call Id=317,

   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)

Jan 22 21:16:02 GMT: //317/3784296381A1/CCAPI/cc_api_call_disconnect_done:

   Call Disconnect Event Sent

Jan 22 21:16:02 GMT: htsp_process_event: [2/0/23, FXOLS_PROCEEDING, E_HTSP_RELEASE_REQ]fxols_offhook_release

Jan 22 21:16:02 GMT: htsp_timer_stop

Jan 22 21:16:02 GMT: htsp_timer_stop2

Jan 22 21:16:02 GMT: htsp_timer_stop3

Jan 22 21:16:02 GMT: [2/0/23] set signal state = 0x4 timestamp = 0

Jan 22 21:16:02 GMT: htsp_timer - 2000 msec

Jan 22 21:16:02 GMT: //316/3784296381A1/CCAPI/cc_api_call_disconnect_done:

   Disposition=0, Interface=0x45E7B524, Tag=0x0, Call Id=316,

   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)

Jan 22 21:16:02 GMT: //316/3784296381A1/CCAPI/cc_api_call_disconnect_done:

   Call Disconnect Event Sent

Jan 22 21:16:04 GMT: htsp_process_event: [2/0/23, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout

Jan 22 21:16:04 GMT: htsp_process_event: [2/0/23, FXOLS_ONHOOK, E_DSP_SIG_0100]

Hope you enabled all the debugs I requested. I don't see H245 negotiation happening. Could you try disabling h245 tunneling on BOTH routers

voice service voice

h323

h245 tunnel disable    <--- hidden command so ? will not work

Also add "voice call send-alert" in global config.

Regards,

Mohit Singh

Thanks for your help

Turns out that it was a problem on the punch-down block between the router and PBX. Nevertheless, the debug commands you gave me helped me to figure it out.