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- About twfredericks
11-05-2018
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Member since
07-15-2008
48
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03-15-2017
09:43 AM
Hi Haris, thanks for the response!
The phones are SCCP enpoints (7942G and 7962G). Do you recommend using a specific debug condition with the "debug voip rtp" to make sure only the specific call is caught?
Also, if I had the user press the ? button twice, what should I ask her to capture. Unfortunately I'm not onsite so have to do this troubleshooting from a distance.
Thanks again,
Tom
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Created by
twfredericks
in IP Telephony and Phones
in IP Telephony and Phones
03-15-2017
08:44 AM
03-15-2017
08:44 AM
Hi all -
I inherited administration of an old CME 7.1 deployment. For the past month or so my office administrator has been experiencing an odd issue where ONLY internal calls periodically have no audio at all. Called party can't hear anything and calling party can't hear anything. Again, this only happens on internal 3-digit dialed calls and often times when she hangs-up and tries again it will work the second time. I have some experience with VoIP but often get lost in the weeds when it comes to CCAPI or VPM debugs. I was hoping that one of your guru's might be able to point me in the right direction of a debug that could help pinpoint this issue. I have some CCAPI debug output for a call that experienced the problem attached; looks to me like the call sets up and disconnects normally but she reports no audio at all.
Greatly appreciate any advice you might give!
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- Tags:
- # voice #cisco #cme
Labels:
08-08-2014
07:42 AM
Hello Guru's - I recently impletemented a medium sized CUCME 9.1 solution in which all users (about 75) are using extension mobility because they change desks often based on the job function they are performing. The site admin requested that the users be able to change their extension mobility password that is used to log into the phone. My question is, is there a way for a user to change that password through CMUSER webpage like in full blown CUCM? If so, is there any specific configruation required to enable this functionalty? I have scoured the web and support communities for some direction but haven't found anything as of yet. I greatly appreciate your time and help! Tom
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Labels:
04-09-2014
04:07 AM
Hi Gurus! I've successfully implemented the FAC (forced authorization code) feature in CUCME 8.6 but I'm unable to find an appropriate .au file that can be used for prompting the user to enter their credentials. I have scoured the support community and Internet and have seen that some folks have used the IVR prompts enter_pin.au or enter_account.au but I can't seem to find them anywhere to download. Anyone out there have any ideas or can share these files in the correct format? Thanks a TON! Tom
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Labels:
Created by
twfredericks
in IP Telephony and Phones
in IP Telephony and Phones
08-22-2013
11:40 AM
08-22-2013
11:40 AM
AMAZING! Worked like a charm. Thanks so much for your quick reply. Any idea what would cause this would happen. I'm looking at an old archived config and the command "no battery-reversal" was not under the voice-port.
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Created by
twfredericks
in IP Telephony and Phones
in IP Telephony and Phones
08-22-2013
10:28 AM
08-22-2013
10:28 AM
Hi All - My company has a small CME deployment in Australia and yesterday the local network resource made some changes to the network related to other work but following their changes, a very weird problem is occuring on the system. The site has 4 POTS lines coming in on a VIC-4FXO card and when calls come in, the IP phones ring but as soon as they try to answer the call disconnects. I was suspecting a codec negotiation issue but since the call is coming from the PSTN, I'm not real sure because it should be G.711 which the IP phones should handle fine. my ccapi debugs show the call disconnecting with the typical 16 cause code (normal clearing). I can provide any part of the config or debug output required for some ideas..... I appreciate the help. Tom
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Labels:
Created by
twfredericks
in IP Telephony and Phones
in IP Telephony and Phones
06-19-2013
04:38 AM
06-19-2013
04:38 AM
Sheesh...I didn't realize that asking a question about a disconnect code was formulating a theory and not working technically. In fact it wasn't my theory at all. I was asking the technical experts if they knew if the SIP timers in CUCM were configurable. I'm in New York and my customer is in Paris. As you can imagine there are some challenges in coordinating test calls for traces to be collected. I'm in the process of getting that set up now.....
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Created by
twfredericks
in IP Telephony and Phones
in IP Telephony and Phones
06-19-2013
04:02 AM
06-19-2013
04:02 AM
Thanks Jorge - While I'm not a SIP expert, I was under the impression that I provided more than just a BYE message. The debug I have attached provides a disconnect cause code of 86 and I was hoping that someone on the forum might have seen this before and recognize a possible root cause. Some of my research indicates that a adjusting a SIP timer might prevent duplicate INVITE messages being sent, which might be causing the disconnect. Thanks for your suggestions. Tom
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Created by
twfredericks
in IP Telephony and Phones
in IP Telephony and Phones
06-18-2013
12:15 PM
06-18-2013
12:15 PM
Some research I've done indicates the CUCM might be sending a duplicate INVITE message to the SIP endpoint causing the active call to be dropped. If this is the case there must be a place in CUCME to configure timers or something and control how often these messages are sent....do you agree?
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Created by
twfredericks
in IP Telephony and Phones
in IP Telephony and Phones
06-18-2013
09:32 AM
06-18-2013
09:32 AM
Hello Cisco Gurus! I seek your assistance again with a weird issue I cannot seem to pinpoint myself. I have a customer that has 3 Biamp VoIP-2 Cards that are using SIP to register to my CUCME system. Thanks to one of you already, they are registering and able to make and receive calls with good quality. The issue arises when a call has been established for about 25-30 minutes it suddenly disconnects with no warning. The 3rd party device seems to stay registered to the CUCME but the call drops. I was able to collect some debug and show sip-ua commands that are hopefully helpful in diagnosing this issue. I appreciate any lead or suggestion as I've met the end of my rope and my customer is quickly losing their patience..... Some SIP config info and debugs are attached. Thanks again! Tom Fredericks
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Labels:
Created by
twfredericks
in IP Telephony and Phones
in IP Telephony and Phones
06-07-2013
07:28 AM
06-07-2013
07:28 AM
You Sir, are a GENIUS! The card registered immediately after I added "authentication realm biamp" to the global config. Thanks for the extremely quick response! Tom
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Created by
twfredericks
in IP Telephony and Phones
in IP Telephony and Phones
06-07-2013
06:49 AM
06-07-2013
06:49 AM
Hi All - I'm at a loss. I have been working on getting some Biamp VoIP-2 cards registered via SIP to my CUCM Express system for 3 days with no luck. I've looked at alot of the discussions in the forums but I'm hoping maybe someone can identify an error in my config preventing the registration. When I do a debug ccsip messages I get an 401 Unauthorized but I have verified the credentials so many times it has to be something else in my config causing the problem. Attached is a snippet of my config and some debug messages. I appreciate any help ya'll can offer. Thanks in advance!! Tom
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Labels:
Created by
twfredericks
in IP Telephony and Phones
in IP Telephony and Phones
11-19-2012
11:49 AM
11-19-2012
11:49 AM
So! I looks like you were right - and I answered prematurely.....my mistake! If the call comes from CUCM into the gateway it works now (before it was not)! I think that it still won't work with a CSIM start because the call is being generated from the gateway - although I'm not sure why. Also I set the calling plan and type to unknown on the route pattern in CUCM to be doubly sure. It will driving me nuts why it was happening though....anyone please feel free to suggest for future victims GRRRRRRR.......
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Created by
twfredericks
in IP Telephony and Phones
in IP Telephony and Phones
11-19-2012
11:35 AM
11-19-2012
11:35 AM
No dice......see below :-( voice translation-rule 30 rule 1 /^9011/ /9011/ type any unknown plan any unknown ! voice translation-profile BA-OUT translate called 30 ! ! ! dial-peer voice 10105 pots description PSTN-LD translation-profile outgoing BA-OUT destination-pattern 90[1-3]T port 0/1/0:15 prefix 0 EOGtgMARcomod01#csim start 901143629912 csim: called number = 901143629912, loop count = 1 ping count = 0 Nov 19 19:29:27.350: ISDN Se0/1/0:15 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 2 Nov 19 19:29:27.350: ISDN Se0/1/0:15 Q931: Sending SETUP callref = 0x00F9 callID = 0x807A switch = primary-4ess interface = User Nov 19 19:29:27.350: ISDN Se0/1/0:15 Q931: TX -> SETUP pd = 8 callref = 0x00F9 Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA9839F Exclusive, Channel 31 Called Party Number i = 0x91, '43629912' Plan:ISDN, Type:International Nov 19 19:29:27.374: ISDN Se0/1/0:15 Q931: RX <- SETUP_ACK pd = 8 callref = 0x80F9 Channel ID i = 0xA9839F Exclusive, Channel 31 Nov 19 19:29:31.406: ISDN Se0/1/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x80F9 csim: loop = 1, failed = 1 csim: call attempted = 1, setup failed = 1, tone failed = 0
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Created by
twfredericks
in IP Telephony and Phones
in IP Telephony and Phones
11-19-2012
11:25 AM
11-19-2012
11:25 AM
show run show version
... View more
03-15-2017
Hi Haris, thanks for the response! The phones are SCCP enpoints (7942G
and 7962G). Do you recommend ...
03-15-2017
Hi all - I inherited administration of an old CME 7.1 deployment. For
the past month or so my office...
08-08-2014
Hello Guru's -I recently impletemented a medium sized CUCME 9.1 solution
in which all users (about 7...
04-09-2014
Hi Gurus!I've successfully implemented the FAC (forced authorization
code) feature in CUCME 8.6 but ...
08-22-2013
AMAZING! Worked like a charm. Thanks so much for your quick reply.Any
idea what would cause this wou...
08-22-2013
Hi All -My company has a small CME deployment in Australia and yesterday
the local network resource ...
06-19-2013
Sheesh...I didn't realize that asking a question about a disconnect code
was formulating a theory an...
06-19-2013
Thanks Jorge -While I'm not a SIP expert, I was under the impression
that I provided more than just ...
06-18-2013
Some research I've done indicates the CUCM might be sending a duplicate
INVITE message to the SIP en...
06-18-2013
Hello Cisco Gurus! I seek your assistance again with a weird issue I
cannot seem to pinpoint myself....
06-07-2013
You Sir, are a GENIUS! The card registered immediately after I added
"authentication realm biamp" to...
06-07-2013
Hi All -I'm at a loss. I have been working on getting some Biamp VoIP-2
cards registered via SIP to ...
11-19-2012
So! I looks like you were right - and I answered prematurely.....my
mistake!If the call comes from C...
11-19-2012
No dice......see below :-(voice translation-rule 30 rule 1 /^9011/
/9011/ type any unknown plan any ...
Public Statistics
| Date Registered | 07-15-2008 07:10 AM |
| Date Last Visited | 11-05-2018 08:56 PM |
| Total Messages Posted | 48 |
Helpful Votes Given To
| User | Helpful Count |
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| 8 | |
| 4 | |
| 5 | |
| 4 | |
| 5 |
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