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- About JAIME KILPATRICK
08-15-2018
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Member since
06-18-2007
9
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5
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Created by
JAIME KILPATRICK
in Cisco Bug Discussions
in Cisco Bug Discussions
09-13-2017
05:12 AM
09-13-2017
05:12 AM
All, We are running into this defect and I was wondering if anyone had any success/fallout from performing the work around by use of the new policy file from the BU? We only have one cluster impacted and it's got a lot of high profile users on it and want to try and build confidence that it won't cause any other issues. If anyone has experience with this, please share! Thank you, Jaime Kilpatrick
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Created by
JAIME KILPATRICK
in IP Telephony and Phones
in IP Telephony and Phones
11-08-2013
06:38 AM
11-08-2013
06:38 AM
All, I figured out the problem. Even though I did NOT copy an existing call handler. When it was built from scratch, it had the alternate Transfer enabled. So, consequently, even though the standard Transfer was setup to transfer etc... the alternate one was set to go to the Greeting. Once I dug into the call handler (after 4 hours of scratching my head) I disabled the alternate and it started working as I would expect. That's just a terrible default to set.
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Created by
JAIME KILPATRICK
in IP Telephony and Phones
in IP Telephony and Phones
11-07-2013
02:47 PM
11-07-2013
02:47 PM
I'm having an issue where I have a call handler setup such that it provides options. One of those options is to attempt to transfer to another call handler. That call handler is setup such that under the transfer section I've entered in an external number. I've checked the calling search spaces using the dialed number analyzer for both the voicemail port and the line on the port and both successfully route the call to the external number. I've also tested this with an IP Communicator phone setup with the same calling search space. (I tried each one individually) Both work. What happens is upon selecting the option (1) it goes directly to the second call handler's greeting. I've double and triple checked and it definitely set for attempt transfer for and the call handler that is setup with "yes, ring this extension" Yet it doesn't ring, I don't even hear a "please wait while I transfer your call" I've checked the restriction tables and everything seems correct there, nothing is set for no and the only option listed is * and yes for allow. I just can't quite pinpoint what could be causing this issue. Any help would be greatly appreciated. Thanks, Jaime Kilpatrick
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Created by
JAIME KILPATRICK
in IP Telephony and Phones
in IP Telephony and Phones
09-06-2012
01:57 PM
09-06-2012
01:57 PM
Hello, I have a 3845 gateway with two T1 circuits coming from diverse carriers. One is an ATM circuit the other a PRI. I'm experiecing slips on the PRI and I believe it to be a clocking issue where the clocks from each of the carriers is far enough off to cause them. The clocking configuration is as follows. ! ! network-clock-participate wic 0 network-clock-participate wic 1 network-clock-participate aim 0 network-clock-select 1 T1 0/0/0 network-clock-select 2 T1 0/1/0 ! What happens is T1 controller 0/1/0 shows slips. Now if I modify the network-clock-select statement such that controller 0/1/0 is the primary, it clears on that circuit but now the ATM circuit is throwing delineation errors. I am at a loss. Is there a way I can resolve this short of putting them on separate hardware or getting both circuits from the same carrier at the same CO? Any help would be greatly appreciated. Thanks, Jaime
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Created by
JAIME KILPATRICK
in IP Telephony and Phones
in IP Telephony and Phones
06-07-2012
11:24 AM
06-07-2012
11:24 AM
Alex, I attempted what you suggested by setting the delay to 1 and it worked! Thank you so much for your help!
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Created by
JAIME KILPATRICK
in IP Telephony and Phones
in IP Telephony and Phones
06-07-2012
09:42 AM
06-07-2012
09:42 AM
Chris, Thanks for the reply. I just tried to do what you suggested (aside from the gateway reboot since I have to do that under a maintenance window) and it didn't work. I removed all dial-peers related to the 0/2/0 voice port. I even removed the trunk group command from the voice port. I shut/no shut the voice port. I rebuilt a dial-peer for that port and no luck. I'm starting to think that the CO isn't sending me a ground acknowledgement. Maybe their equipment simulates GS but doesn't simulate it according to the GS standard kind of like how the old VIC-4FXO cards used to no look for the acknowledgement? If so, does anyone know of a command in IOS to have the router not wait for that? Thanks, Jaime
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Created by
JAIME KILPATRICK
in IP Telephony and Phones
in IP Telephony and Phones
06-07-2012
08:29 AM
06-07-2012
08:29 AM
Tracy, Thanks for the reply. Unfortunately, I had shut/no shut these voice ports repeatedly. The only thing I have yet to attempt is a full reboot of the gateway. That will be this evening during a maintenance window. In the mean time however I have done one thing that completely blows my mind away. Get this... I have a telnet session to the gateway and run the csim start command. In a second session I ran the command test voice port 0/2/0 detector tip-ground on during the waiting period for the ground and guess what, the call rings out and completes to my test phone. Essentially, what I believe the test command does is force the gateway into thinking that it received the ground response from the CO. At this point I haven't determined if the gateway is simply not detecting the tip-ground acknowledgement or if the CO just isn't sending it. I guess that's a test I need to conduct at this point. Anyone have any tips on how to do that? If I use a buttset on these lines it works fine since the buttset doesn't care about a tip-ground acknowledgement. Does anyone have a solid process for testing this scenario?
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Created by
JAIME KILPATRICK
in IP Telephony and Phones
in IP Telephony and Phones
06-06-2012
01:36 PM
06-06-2012
01:36 PM
To anyone with a lot of experience out there: I cannot seem to make outbound calls work through my voice gateway. Below are the details of my dilemma. I am using csim start command via telnet. I have the following gateway hardware. 2811 w/ PVDM2-16 and a VWIC2-4FXO card. IOS version is 15.0(1)M5 Voice-ports 0/2/0,0/2/1,0/2/2 have groundstart lines plugged into them. Voice port 0/2/3 has no configuration and is shut down. These lines have been verified to indeed be ground start lines. I took my buttset to them personally and could only obtain dial tone when I touched a ground wire (That I had attached to the screw that secures the VWIC2-4FXO card into the chassis to the ring side of each line. I would then get dial tone and could place a call to my cell phone (It's a local 7 digit calling area so I dialed 7 digits) I have a dial-peer setup as such dial-peer voice 8888 pots destination-pattern 9T forward-digits 7 port 0/2/0 I am accessing the gateway via telnet (as ssh does not work with the csim start command) I have debug vpm all turned on with monitor terminal Here is the output when the csim start command is issued. Jun 6 16:21:26: htsp_timer_stop3 htsp_setup_req Jun 6 16:21:26: htsp_process_event: [0/2/0, FXOGS_ONHOOK, E_HTSP_SETUP_REQ]fxogs_onhook_setup Jun 6 16:21:26: [0/2/0] set signal state = 0x0 timestamp = 0 Jun 6 16:21:26: dsp_set_sig_state: [0/2/0] packet_len=12 channel_id=128 packet_id=39 state=0x0 timestamp=0x0 Jun 6 16:21:26: TGRM: reg_invoke_tgrm_call_update(0, 2, 0, 65535, 1, TGRM_CALL_BUSY, TGRM_CALL_VOICE, TGRM_DIRECTION_OUT) Jun 6 16:21:26: htsp_timer - 10000 msec Jun 6 16:21:26: htsp_process_event: [0/2/0, FXOGS_WAIT_TIP_GROUND, E_DSP_SIG_0110] Jun 6 16:21:26: [0/2/0, FXOGS_WAIT_TIP_GROUND, E_DSP_SIG_0110] -> ERROR: INVALID INPUT Jun 6 16:21:26: htsp_process_event: [0/2/0, FXOGS_WAIT_TIP_GROUND, E_DSP_SIG_0110] Jun 6 16:21:26: [0/2/0, FXOGS_WAIT_TIP_GROUND, E_DSP_SIG_0110] -> ERROR: INVALID INPUT Jun 6 16:21:36: htsp_process_event: [0/2/0, FXOGS_WAIT_TIP_GROUND, E_HTSP_EVENT_TIMER]fxogs_offhook_disc Jun 6 16:21:36: htsp_timer_stop Jun 6 16:21:36: [0/2/0] set signal state = 0x4 timestamp = 0 Jun 6 16:21:36: dsp_set_sig_state: [0/2/0] packet_len=12 channel_id=128 packet_id=39 state=0x4 timestamp=0x0 Jun 6 16:21:36: htsp_timer - 2000 msechtsp_release_req: cause 16, no_onhook 0 Jun 6 16:21:36: htsp_process_event: [0/2/0, FXOGS_ONHOOK, E_HTSP_RELEASE_REQ]fxogs_onhook_release Jun 6 16:21:36: htsp_timer_stop2 Jun 6 16:21:36: htsp_timer_stop3 Jun 6 16:21:36: TGRM: reg_invoke_tgrm_call_update(0, 2, 0, 65535, 1, TGRM_CALL_IDLE, TGRM_CALL_VOICE, TGRM_DIRECTION_OUT) Jun 6 16:21:36: flex_dsprm_close_cleanup Jun 6 16:21:38: htsp_process_event: [0/2/0, FXOGS_ONHOOK, E_HTSP_EVENT_TIMER] Jun 6 16:21:38: [0/2/0, FXOGS_ONHOOK, E_HTSP_EVENT_TIMER] -> ERROR: INVALID INPUT I've scoured the internet looking to make heads or tails of this and the only thing I've come up with is there are some related Cisco bugs that seem to point in this direction however, I have performed a show voice dsp and I get the following: *DSP VOICE CHANNELS* CURR STATE : (busy)inuse (b-out)busy out (bpend)busyout pending LEGEND : (bad)bad (shut)shutdown (dpend)download pending DSP DSP DSPWARE CURR BOOT PAK TX/RX TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT ====== === == ========= ========== ===== ======= === == ========= == ==== ============ *DSP SIGNALING CHANNELS* DSP DSP DSPWARE CURR BOOT PAK TX/RX TYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT ====== === == ========= ========== ===== ======= === == ========= == ==== ============ C5510 001 01 {flex} 26.3.9 alloc idle 0 0 0/2/0 02 0 110/0 C5510 001 02 {flex} 26.3.9 alloc idle 0 0 0/2/1 06 0 125/0 C5510 001 03 {flex} 26.3.9 alloc idle 0 0 0/2/2 10 0 163/0 C5510 001 04 {flex} 26.3.9 alloc idle 0 0 0/2/3 14 0 55/0 ------------------------END OF FLEX VOICE CARD 0 ---------------------------- I believe the firmware should be new enough to have resolved the issue. If someone could possibly provide me with a little direction here that would be great. Below are summaries of the tests I've done. Configured as loopstart, it complains of a power denial which I think would be normal since there is no connection until the grounding has been established. Please correct me if I'm wrong btw. Jun 6 16:31:26: htsp_timer_stop3 htsp_setup_req Jun 6 16:31:26: htsp_process_event: [0/2/0, FXOLS_ONHOOK, E_HTSP_SETUP_REQ]fxols_onhook_setup Jun 6 16:31:26: [0/2/0] set signal state = 0xC timestamp = 0 Jun 6 16:31:26: dsp_set_sig_state: [0/2/0] packet_len=12 channel_id=128 packet_id=39 state=0xC timestamp=0x0 Jun 6 16:31:26: TGRM: reg_invoke_tgrm_call_update(0, 2, 0, 65535, 1, TGRM_CALL_BUSY, TGRM_CALL_VOICE, TGRM_DIRECTION_OUT) Jun 6 16:31:26: htsp_timer - 1300 msec Jun 6 16:31:26: htsp_process_event: [0/2/0, FXOLS_WAIT_DIAL_TONE, E_DSP_SIG_1100]fxols_power_denial_detected Jun 6 16:31:26: htsp_timer2 - 1000 msec Jun 6 16:31:26: htsp_timer_stop Jun 6 16:31:27: htsp_process_event: [0/2/0, FXOLS_WAIT_DIAL_TONE, E_HTSP_EVENT_TIMER2]fxols_power_den_disc Jun 6 16:31:27: htsp_timer_stop Jun 6 16:31:27: htsp_timer_stop2 Jun 6 16:31:27: [0/2/0] set signal state = 0x4 timestamp = 0 Jun 6 16:31:27: dsp_set_sig_state: [0/2/0] packet_len=12 channel_id=128 packet_id=39 state=0x4 timestamp=0x0 Jun 6 16:31:27: mars_flex_dsprm_current_codec_comp:DSP:0 FLEX Complexity Codec htsp_release_req: cause 34, no_onhook 0 Jun 6 16:31:27: htsp_process_event: [0/2/0, FXOLS_ONHOOK, E_HTSP_RELEASE_REQ]fxols_onhook_release I've reverse polarity on the circuit and the debug simply tells me that the polarity is reversed so I know that the circuit is correct as is. (It's not complaining about it.) Again, I have tested the lines and they work just fine with the buttset using a ground to the ring side from the router chassis. I'm preplexed at this point. Any help would be greatly appreciated. Thanks, Jaime
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Labels:
09-13-2017
All, We are running into this defect and I was wondering if anyone had
any success/fallout from perf...
Created by
JAIME KILPATRICK
in IP Telephony and Phones
11-08-2013
11-08-2013
All,I figured out the problem. Even though I did NOT copy an existing
call handler. When it was buil...
Created by
JAIME KILPATRICK
in IP Telephony and Phones
11-07-2013
11-07-2013
I'm having an issue where I have a call handler setup such that it
provides options. One of those op...
Created by
JAIME KILPATRICK
in IP Telephony and Phones
09-06-2012
09-06-2012
Hello,I have a 3845 gateway with two T1 circuits coming from diverse
carriers. One is an ATM circuit...
Created by
JAIME KILPATRICK
in IP Telephony and Phones
06-07-2012
06-07-2012
Alex,I attempted what you suggested by setting the delay to 1 and it
worked! Thank you so much for y...
Created by
JAIME KILPATRICK
in IP Telephony and Phones
06-07-2012
06-07-2012
Chris,Thanks for the reply. I just tried to do what you suggested (aside
from the gateway reboot sin...
Created by
JAIME KILPATRICK
in IP Telephony and Phones
06-07-2012
06-07-2012
Tracy,Thanks for the reply. Unfortunately, I had shut/no shut these
voice ports repeatedly. The only...
Created by
JAIME KILPATRICK
in IP Telephony and Phones
06-06-2012
06-06-2012
To anyone with a lot of experience out there:I cannot seem to make
outbound calls work through my vo...
Created by
JAIME KILPATRICK
in Unified Communications Infrastructure
06-18-2007
06-18-2007
All,One of my Unity admins came across an interesting alert in our
network monitoring system. It is ...
Public Statistics
| Date Registered | 06-18-2007 10:19 AM |
| Date Last Visited | 08-15-2018 02:19 AM |
| Total Messages Posted | 9 |
| Total Helpful Votes Received | 5 |
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