State of the Call - State Dead

[Yort Mantup]

Having an issue with an existing setup.  4 analog lines into an FXO with a SIP connection back to Call Manager.   PSTN --> FXO --> SIP --CUCM

The four lines are setup by the carrier as a hunt group.  The main line can no longer ring in.  When dialed, it will ring the phone one time and quit.  On the caller end they just hear the line continuously ringing.

I dialed the other 3 lines and no issue with those.  Running sip debugs I get the following output for the main line when dialed.  The two things that stand out is STATE_DEAD and No Codec for negotiated codec.  I did submit a ticket with the carrier but no issue found.  No changes occurred prior to when this issue started.

The Call Setup Information is:
Call Control Block (CCB) : 0x0x3F7CF7A0
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 6169291350
Called Number : 7092401
Source IP Address (Sig ): 10.92.4.1
Destn SIP Req Addr:Port : 172.20.1.101:5060
Destn SIP Resp Addr:Port : 172.20.1.101:5060
Destination Name : 172.20.1.101

Jul 12 12:16:11.151: //11664/324DA4E3B0CE/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 10.92.4.1
Source IP Port (Media): 17206
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

Jul 12 12:16:11.151: //11664/324DA4E3B0CE/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 487

[Nashaatmusa]

Can you share the following for the main line :
debug vpm signal
debug ccsip messages

to look through on happens to the call .

[Yort Mantup]

Nashaatmusa,

Please see attached output for debug vpm signal and
debug ccsip messages

Thank you

[Nashaatmusa]

First the Call got answered from the FXO :
Jul 12 13:15:10.853: fxols_callerid_done: call being answered

I can see that the SIP part is working fine ,, but gateway sends Cancel to the CUCM when the call manager sends a message that the phone is ringing.

Then I can see this before the gateway send cancel :
Jul 12 13:15:10.969: htsp_process_event: [0/2/0, FXOLS_PROCEEDING,E_HTSP_ALERT]fxols_offhook_alert
Jul 12 13:15:11.109: htsp_dsp_message: SEND_SIG_STATUS: state=0xCtimestamp=1004 systime=17215535
Jul 12 13:15:11.109: htsp_process_event: [0/2/0, FXOLS_PROCEEDING,E_DSP_SIG_1100]fxols_offhook_disc
Jul 12 13:15:11.461: htsp_process_event: [0/2/0, FXOLS_PROCEEDING, E_HTSP_RELEASE_REQ]fxols_offhook_release

but we can't see how is dropping is the gateway or the provider !

I think you made the test of connecting an analouge phone to the same port that facing the issue and it worked fine correct ?

anyhow we need to look more into what is happening that that time :
debug vpm all -- this debugs Voice Port Module SPI information
debug hpi all -- HPI (54x) DSP message
debug vtsp all -- Voice Telephony Call Control information

This might indicate what the problem is but first make the test of the analogue phone