Inbound calls on h323 gateway with FXO ports

[techguy]

Hi,

I am facing problem in inbound calls in H.323 gateway with FXO ports. Outbound calls are working perfect but inbound calls doesnt land anywhere and keeps ringing. Both connection plar to CTI RP or connection plar to one extension doesnt work.

I dubug FXO ports with "debug vpm signal" only and get the following messages.

*Dec 30 11:55:02.391: htsp_timer_stop3 htsp_setup_req
*Dec 30 11:55:02.391: htsp_process_event: [0/0/0, FXOLS_ONHOOK, E_HTSP_SETUP_REQ]fxols_onhook_setup
*Dec 30 11:55:02.391: [0/0/0] set signal state = 0xC timestamp = 0
*Dec 30 11:55:02.391: htsp_timer - 1300 msec
*Dec 30 11:55:02.663: htsp_process_event: [0/0/0, FXOLS_WAIT_DIAL_TONE, E_DSP_SIG_0110]fxols_disc_clear
*Dec 30 11:55:02.663: htsp_timer_stop2
*Dec 30 11:55:02.663: htsp_timer - 1300 msec
*Dec 30 11:55:03.963: htsp_process_event: [0/0/0, FXOLS_WAIT_DIAL_TONE, E_HTSP_EVENT_TIMER]fxols_wait_dial_timer  htsp_dial
*Dec 30 11:55:05.987: htsp_process_event: [0/0/0, FXOLS_WAIT_DIAL_DONE, E_DSP_DIALING_DONE]fxols_wait_dial_done htsp_progress
*Dec 30 11:55:05.987: htsp_timer - 350 msec
*Dec 30 11:55:05.987: htsp_call_bridged invoked
*Dec 30 11:55:06.339: htsp_process_event: [0/0/0, FXOLS_WAIT_CUT_THRU, E_HTSP_EVENT_TIMER]fxols_handle_cut_thru
*Dec 30 11:55:06.339: htsp_timer_stop
*Dec 30 11:55:07.255: htsp_process_event: [0/0/0, FXOLS_OFFHOOK, E_HTSP_VOICE_CUT_THROUGH]fxols_proc_voice
*Dec 30 11:55:53.623: htsp_timer_stop3
*Dec 30 11:55:53.639: htsp_process_event: [0/0/0, FXOLS_OFFHOOK, E_HTSP_RELEASE_REQ]fxols_offhook_release
*Dec 30 11:55:53.639: htsp_timer_stop
*Dec 30 11:55:53.639: htsp_timer_stop2
*Dec 30 11:55:53.639: htsp_timer_stop3
*Dec 30 11:55:53.639: [0/0/0] set signal state = 0x4 timestamp = 0
*Dec 30 11:55:53.639: htsp_timer - 2000 msec
*Dec 30 11:55:53.915: htsp_process_event: [0/0/0, FXOLS_GUARD_OUT, E_DSP_SIG_0110]
*Dec 30 11:55:55.639: htsp_process_event: [0/0/0, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
*Dec 30 11:55:55.639: htsp_process_event: [0/0/0, FXOLS_ONHOOK, E_DSP_SIG_0100]

[Aaron Harrison]

Can you post your config?

Does the target configured on connection plar match a dial-peer you have configured?

Aaron

[techguy]

Yes, Of course.

voice-port 0/0/0
 trunk-group LOCAL
 input gain -3
 output attenuation -1
 echo-cancel coverage 64
 no comfort-noise
 cptone SA
 timing hookflash-out 50
 connection plar 2233
 description *** LOCAL LINE ***
 music-threshold -45
 bearer-cap Speech
!
voice-port 0/0/1
 trunk-group LOCAL
 input gain -3
 output attenuation -1
 echo-cancel coverage 64
 no comfort-noise
 cptone SA
 timing hookflash-out 50
 connection plar 2233
 description *** LOCAL LINE ***
 music-threshold -45
 bearer-cap Speech

!

dial-peer voice 2233 voip
 description To CallManager Pub
 preference 1
 destination-pattern 2233
 session target ipv4:10.50.2.3
 dtmf-relay h245-alphanumeric
 req-qos guaranteed-delay audio
 acc-qos guaranteed-delay audio
 codec g711ulaw
 fax-relay ecm disable
 fax rate 14400
 fax nsf 000000
 fax protocol pass-through g711ulaw
 no vad
!
dial-peer voice 2234 voip
 description To CallManager Sub
 preference 2
 destination-pattern 2233
 session target ipv4:10.50.2.4
 dtmf-relay h245-alphanumeric
 req-qos guaranteed-delay audio
 acc-qos guaranteed-delay audio
 codec g711ulaw
 fax-relay ecm disable
 fax rate 14400
 fax nsf 000000
 fax protocol pass-through g711ulaw
 no vad

[Aaron Harrison]

OK - so have you verified those dial-peers work?

e.g. from the router try this:

csim start 2233

That should do a test call to your phone. It will be dead when you answer, but proves the call will route to the point where it rings.

Aaron

[techguy]

it happened as you said.

[Aaron Harrison]

Hmm.

Try this instead:

connection plar opx 2233

Also if that fails, do a 'debug voip dial-peer default' when calling the FXO to verify whether it triggers the peer...

Otherwise I would try stripping back your dial-peers to a simpler config and building from there... for example, removing the RSVP lines. 

Aaron

[techguy]

First we need to confirm that if FXO ports got alert when call lands on port. then we can check that dial-peer is matching or not?

not showing any output when enabling debug voip dialpeer default.

[Adrian Saavedra]

Hi,

Are you sure the call is hitting those dialpeers?. You can validate that doing a debug voip dialpeer all. You can also check doing a show voice call status.

If call is hitting the dialpeer, then you can check if that number is reachable via the inbound CSS of your gateway defined in CallManager.

- Adrian.

[techguy]

As suggested by Aaron, its happening. Inbound Calls are coming on 2233.