Voice-port rings with no voice

[encarter]

Hi all again....

I have a 4-Port FXO card on a router and we have 4 pots lines associated to it.

When i place a call from it for example to my cell phone. My cell will ring, my IP Phone will not hear ring back. My cell will ring like 2 times and my ip phone will give a fast busy.

The same happens when im placing an inbound call.

Here is the "dubug vpm signal"

Sep  3 16:09:13.371: htsp_timer_stop3 htsp_setup_req
Sep  3 16:09:13.375: htsp_process_event: [0/2/0, FXOLS_ONHOOK, E_HTSP_SETUP_REQ]fxols_onhook_setup
Sep  3 16:09:13.375: [0/2/0] set signal state = 0xC timestamp = 0
Sep  3 16:09:13.375: dsp_set_sig_state: [0/2/0] packet_len=12 channel_id=128 packet_id=39 state=0xC timestamp=0x0
Sep  3 16:09:13.375: TGRM: reg_invoke_tgrm_call_update(0, 2, 0, 65535, 1, TGRM_CALL_BUSY, TGRM_CALL_VOICE, TGRM_DIRECTION_OUT)
Sep  3 16:09:13.375: htsp_timer - 1300 msec
Sep  3 16:09:14.675: htsp_process_event: [0/2/0, FXOLS_WAIT_DIAL_TONE, E_HTSP_EVENT_TIMER]fxols_wait_dial_timer  htsp_dial
Sep  3 16:09:16.915: htsp_process_event: [0/2/0, FXOLS_WAIT_DIAL_DONE, E_DSP_DIALING_DONE]fxols_wait_dial_done htsp_progress
Sep  3 16:09:16.915: htsp_timer - 350 msec
Sep  3 16:09:16.919: htsp_call_bridged invoked
Sep  3 16:09:17.267: htsp_process_event: [0/2/0, FXOLS_WAIT_CUT_THRU, E_HTSP_EVENT_TIMER]fxols_handle_cut_thru
Sep  3 16:09:17.267: htsp_timer_stop    [b] **** IT WILL HANG HERE WHILE MY CELL RINGS THEN I HEAR FAST BUSY ******[/b]
Sep  3 16:09:29.007: htsp_timer_stop3 htsp_release_req: cause 47, no_onhook 0
Sep  3 16:09:29.011: htsp_process_event: [0/2/0, FXOLS_OFFHOOK, E_HTSP_RELEASE_REQ]fxols_offhook_release
Sep  3 16:09:29.011: htsp_timer_stop 
Sep  3 16:09:29.011: htsp_timer_stop2 
Sep  3 16:09:29.011: htsp_timer_stop3 
Sep  3 16:09:29.011: [0/2/0] set signal state = 0x4 timestamp = 0
Sep  3 16:09:29.011: dsp_set_sig_state: [0/2/0] packet_len=12 channel_id=128 packet_id=39 state=0x4 timestamp=0x0
Sep  3 16:09:29.011: TGRM: reg_invoke_tgrm_call_update(0, 2, 0, 65535, 1, TGRM_CALL_IDLE, TGRM_CALL_VOICE, TGRM_DIRECTION_OUT)
Sep  3 16:09:29.011: htsp_timer - 2000 msec
Sep  3 16:09:29.011: TGRM: reg_invoke_tgrm_call_update(0, 2, 0, 65535, 1, TGRM_CALL_BUSY, TGRM_CALL_VOICE, TGRM_DIRECTION_OUT)
Sep  3 16:09:29.027: flex_dsprm_close_cleanup
Sep  3 16:09:31.011: htsp_process_event: [0/2/0, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
Sep  3 16:09:31.011: TGRM: reg_invoke_tgrm_call_update(0, 2, 0, 65535, 1, TGRM_CALL_IDLE, TGRM_CALL_VOICE, TGRM_DIRECTION_OUT)
Sep  3 16:09:31.011: dsp_req_sig_state: [0/2/0] packet_len=8 channel_id=128 packet_id=40 
Sep  3 16:09:31.011: htsp_process_event: [0/2/0, FXOLS_ONHOOK, E_DSP_SIG_0100]

 

[encarter]

ok figured it out.

It was two things...

1) I did a reload this fixed the ringback on my IP Phone, but still no audio voice.
I noticed that when i ran sh call active voice brief that the originating IP (in this case my IP Phone) was 0.0.0.0 after the reload my IP address of my phone showed up.

Thinking about this more, I did make quite a few changes going from CME to H323.

2) The second thing was h323-gateway voip bind srcaddr under my lan interface. I really didnt think this was needed for a h323 gateway unless you were sourcing a different IP than what was configured on the interface.

Adding that line gave me the 2-way voice call.

Yay !!!

Issue Resolved.

 

[BuckWeet]

Its been a while since I've done dial-peers.. I believe you can set some of these settings under the dial-peer as well.. FYI for ring back, if/when you upgrade to ISDN, you have to configure call signaling under the dial-peers.


BuckWeet