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- About Jorge Armijo
10-12-2017
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Member since
05-26-2011
476
Posts
328
Helpful
56
Solutions
12-08-2015
02:59 PM
Hi,
1-ATA186 is SCCP only
2-ATA186 does not support T38
3-Passthrough with SCCP is NSE only (Cisco Proprietary)
4-I wouldn’t use an ATA186 with a SIP Trunk; the Fax Protocols are not compatible
5-I suggest using a VG20X as small and low cost resource or ATA187/190 instead.
HTH
~jorge
... View more
12-08-2015
02:40 PM
David,
The commands you mentioned will actually prevent the mid-call reINVITE to change the media status (inactive, sendonly, receiveonly, inactive). Hoewver, it does not neccesarily means it is a problem with the CUBE, CUCM, etc. Most of the time, the Carrier is not capable to renegotiate the media attributes correctly and hence we have one way or no audio at situations.
The commands can be added, however, without the proper logs we cannot identify what the actual cause of the issue is and provide a good RCA + a solution/workaround.
Mid-call reINVITEs are perfectly fine and expected on transfers or HOLD/RESUME scenarios, the problem appears when one of the devices in the flow cannot process the requests properly.
HTH
~jorge
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12-08-2015
02:34 PM
Hey,
15.1(4)MX is the latest available version for 2800/3800 series routers. An IOS upgrade and hence DSPware upgrade is no go.
You'll definitively need a PCM capture to isolate what the source of the craking noise is:
voice hpi capture buffer 1000000 voice hpi capture destination flash:pcm.dat Once you have configured the above, make the test call and look for the voice-port channel that the call comes in on. For instance, if the call comes in on the voice-port X/Y/Z then you will need to execute the following exec level command: test voice port X/Y/Z pcm-dump caplog 7 Once the capture is complete, make sure to disable the capture by executing the following:
test voice port X/Y/Z pcm-dump disable Also get into the config mode and disable the two commands that you had configured before:
no voice hpi capture buffer 1000000
no voice hpi capture destination flash:pcm.dat
Get the PCM capture so we can decode the audio.
HTH
~jorge
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Created by
Jorge Armijo
in Collaboration, Voice and Video Documents
in Collaboration, Voice and Video Documents
08-06-2014
05:55 AM
08-06-2014
05:55 AM
There's no official support on CME for Android Jabber: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmeroad.html ~jorge
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08-05-2014
09:18 AM
Hey Guys, That's a problem with 3905 firmware, please upgrade firmware to 9.4(1), which has the payload issue fixed. HTH ~jorge
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05-29-2014
01:42 PM
As Brian mentioned, ISM/SM-SRE modules, are shipped with no software preinstalled, since the module supports several different applications, and hence they decided to not preinstall any software. Basically you receive an empty box, ready to be installed/deployed. Cisco SRE Service Module Configuration and Installation Guide http://www.cisco.com/c/en/us/td/docs/routers/access/interfaces/software/feature/guide/ism-sm-sre.html HTH ~jorge
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Created by
Jorge Armijo
in IP Telephony and Phones
in IP Telephony and Phones
05-29-2014
01:37 PM
05-29-2014
01:37 PM
There's a CUBE in the middle, can you capture: debug ccsip messages debug voice ccapi inout Please also attach the sh run from the CUBE. HTH ~jorge
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05-29-2014
01:28 PM
Can you please do: voice register global no create profile create profile Then, reset the ip phone and capture the following debugs: debug tftp events debug ccsip messages HTH ~jorge
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05-24-2014
10:03 AM
Kudos to Manish, Indeed the SIP Profiles work for outgoing direction only, in additon to permit the Diversion Header will force the CUBE to pass all the unsupported headers as stated on the command reference, you may consider tha fact that the device behind the CUBE also allows inbound DIVERSION. voice service voip sip pass-thru { content { sdp | unsupp } | headers { unsupp | list tag }} ~jorge
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05-24-2014
09:57 AM
Please make sure the subnet range tha ip phone belongs to, it allowed over the tunnel, in addition to port 2000 (SCCP), 5060 (SIP), the VPN tunnel has to permit the RTP ports range " 16384-32767 " http://www.voip-info.org/wiki/view/RTP+Ports ~jorge
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05-24-2014
09:55 AM
Can you try: voice-card x <- Where x is usually 0 no local-bypass ~jorge
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03-14-2014
05:16 PM
Hi, There are many possible scenarios that could explain what is going on, before going further, can you please post the output from: -show diag -show inventory -show version -show voice dsp group all Remember that you can setup the codec complexity under "voice-card", and if you use "Medium or High" the calculation will be different, in addition, is there any other Voice Interface Card installed on the Router? Let's recall few things: 1 PVDM3-64 has 480 Credits per core, since the PVDM has two cores, we have a total of 960 credits. http://www.cisco.com/c/en/us/products/collateral/ios-nx-os-software/ios-ip-service-level-agreements-slas/whitepaper_C11-718333.html One single DSO on a T1-E1 consumes 15 credits if Codec Complexity Flex is being used (Default), now, some math: E1 = 30 Bchannels 4 E1 = 120 Bchannels 120 * 15 = 1800 Credits You'll need at least another PVDM3-64 to support the 4 E1 interfaces. I think you may want to verify the DSP Calculator again: http://www.cisco.com/web/applicat/dsprecal/dsp_calc.html Router: 2951 IOS: 15.1(4)M DSP Modules Required: PVDM3-128: 1 DSP Module Allocation: Router Slot 0: 120 Voice (94%) 8 Available (6%) PVDM Slot 0/0: PVDM3-128 Call Volume Allocation: Router Slot 0 Low Med High Slot 0/0: VWIC3-4MFT-T1/E1(E1) 120 BTW, I created the Calculator Session with Low Complexity, you have to make sure no g729 calls will go through the Router, otherwise, additional resources will be needed. HTH ~jorge
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Created by
Jorge Armijo
in IP Telephony and Phones
in IP Telephony and Phones
01-07-2014
11:44 AM
01-07-2014
11:44 AM
Please try to upgrade to the latest available firmware, 9.4(1), and try again. HTH -- Jorge Armijo Please remember to rate helpful responses and identify helpful or correct answers.
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Created by
Jorge Armijo
in IP Telephony and Phones
in IP Telephony and Phones
12-22-2013
03:01 PM
12-22-2013
03:01 PM
This should give you an idea where to start http://www.icciev.com/1/post/2011/09/adding-vg224-to-cucm-80-as-sccp-or-mgcp-gateway-differences-and-configurations-part-2.html -- Jorge Armijo Please remember to rate helpful responses and identify helpful or correct answers.
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12-08-2015
Hi, 1-ATA186 is SCCP only 2-ATA186 does not support T38 3-Passthrough
with SCCP is NSE only (Cisco P...
12-08-2015
David, The commands you mentioned will actually prevent the mid-call
reINVITE to change the media st...
12-08-2015
Hey, 15.1(4)MX is the latest available version for 2800/3800 series
routers. An IOS upgrade and henc...
Created by
Jorge Armijo
in Collaboration, Voice and Video Documents
08-06-2014
08-06-2014
There's no official support on CME for Android
Jabber:http://www.cisco.com/c/en/us/td/docs/voice_ip_...
08-05-2014
Hey Guys, That's a problem with 3905 firmware, please upgrade firmware
to 9.4(1), which has the payl...
05-29-2014
As Brian mentioned, ISM/SM-SRE modules, are shipped with no software
preinstalled, since the module ...
05-29-2014
There's a CUBE in the middle, can you capture: debug ccsip messagesdebug
voice ccapi inout Please al...
05-29-2014
Can you please do: voice register global no create profile create
profile Then, reset the ip phone a...
05-24-2014
Kudos to Manish, Indeed the SIP Profiles work for outgoing direction
only, in additon to permit the ...
05-24-2014
Please make sure the subnet range tha ip phone belongs to, it allowed
over the tunnel, in addition t...
03-14-2014
Hi, There are many possible scenarios that could explain what is going
on, before going further, can...
Public Statistics
| Date Registered | 05-26-2011 03:53 PM |
| Date Last Visited | 10-12-2017 10:35 PM |
| Total Messages Posted | 476 |
| Total Helpful Votes Received | 328 |
Helpful Votes Given To
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|---|---|
| 30 | |
| 5 | |
| 85 | |
| 5 | |
| 5 |
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Helpful Votes From
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| 5 | |
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| 5 | |
| 10 |
