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Today's Topics:
1. T.38 vs modem passthrough (Scott Voll)
2. PCS 5YY-NXX numbers... comments? (Lelio Fulgenzi)
3. CUCM Log partition full after CUCM 8.6 SU1 -> SU2 upgrade
(JP Senior)
4. Re: CUCM Log partition full after CUCM 8.6 SU1 -> SU2 upgrade
(Nate VanMaren)
5. Re: DTMF Issue with one external number (Nick Matthews)
6. Re: T.38 vs modem passthrough (Nick Matthews)
7. ATA Call Pickup Group or something else? (Madziarczyk, Jonathan)
8. Re: T.38 vs modem passthrough (Eric Pedersen)
----------------------------------------------------------------------
Message: 1
Date: Mon, 22 Oct 2012 13:52:35 -0700
From: Scott Voll <[email protected]>
To: [email protected]
Subject: [cisco-voip] T.38 vs modem passthrough
Message-ID:
<CAHgd+38ZV3t-hWfOaEU5y+ZJES4pUZhfZ=g9bh_dof-wyc-...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
if everything we have is setup to use t.38 and I need Modem passthrough
also, Can they both run together?
CM 7.1
VG224
MGCP VGW 15.1T code
TIA
Scott
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Message: 2
Date: Mon, 22 Oct 2012 18:08:32 -0400 (EDT)
From: Lelio Fulgenzi <[email protected]>
To: "cisco-voip ([email protected])"
<[email protected]>
Subject: [cisco-voip] PCS 5YY-NXX numbers... comments?
Message-ID:
<1921690042.63262.1350943712627.javamail.r...@squeaky.cs.uoguelph.ca>
Content-Type: text/plain; charset="utf-8"
Anyone have any background on the PCS 5YY-NXX number assignments and how
they're used?
---
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (ANNU)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
Cooking with unix is easy. You just sed it and forget it.
- LFJ (with apologies to Mr. Popeil)
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Message: 3
Date: Mon, 22 Oct 2012 22:52:15 +0000
From: JP Senior <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] CUCM Log partition full after CUCM 8.6 SU1 ->
SU2 upgrade
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Hi, all.
Is it possible to wipe logs off an inactive partition or to manually 'zero out'
a log partition? After a CUCM upgrade we're getting flooded with
LogPartitionLowWaterMarkExceeded errors on the log partition, size is over 90%.
Previously it was approximately 45%, it appears call manager duplicated the
logs during the upgrade but did nothing to remediate the old ones still sitting
around.
Warm regards,
-JP Senior
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Message: 4
Date: Mon, 22 Oct 2012 23:30:49 +0000
From: Nate VanMaren <[email protected]>
To: JP Senior <[email protected]>, "[email protected]"
<[email protected]>
Subject: Re: [cisco-voip] CUCM Log partition full after CUCM 8.6 SU1
-> SU2 upgrade
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Stuff gets cleared once it gets past the watermarks, After upgrade I usually
do something like this (exact values depend on my mood)
Change the low and high water marks for the log partition:
. In RTMT click on 'Alert Central'
. Then select 'LogPartitionLowWaterMarkexceeded'
. Right click and select 'Set Alert/Properties' - set this to 60%
Now repeat the previous steps for LogPartitionHighWaterMarkexceedes and set
it to 70%
Restart the "Cisco Log Partition Monitoring Tool" service under
Serviceability -> Network Services
The sets the value back to the original and this should clear the logging
from space and should fix the issue .
From: [email protected]
[mailto:[email protected]] On Behalf Of JP Senior
Sent: Monday, October 22, 2012 4:52 PM
To: [email protected]
Subject: [cisco-voip] CUCM Log partition full after CUCM 8.6 SU1 -> SU2 upgrade
Hi, all.
Is it possible to wipe logs off an inactive partition or to manually 'zero out'
a log partition? After a CUCM upgrade we're getting flooded with
LogPartitionLowWaterMarkExceeded errors on the log partition, size is over 90%.
Previously it was approximately 45%, it appears call manager duplicated the
logs during the upgrade but did nothing to remediate the old ones still sitting
around.
Warm regards,
-JP Senior
The contents of this message may contain confidential and/or privileged
subject matter. If this message has been received in error, please contact
the sender and delete all copies. Like other forms of communication,
e-mail communications may be vulnerable to interception by unauthorized
parties. If you do not wish us to communicate with you by e-mail, please
notify us at your earliest convenience. In the absence of such
notification, your consent is assumed. Should you choose to allow us to
communicate by e-mail, we will not take any additional security measures
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Message: 5
Date: Mon, 22 Oct 2012 20:53:57 -0400
From: Nick Matthews <[email protected]>
To: [email protected]
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] DTMF Issue with one external number
Message-ID:
<cam-k-notufke258k1cfbrlc+d6jdh65ndeezy_odoahbd1o...@mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"
You may have to create a special dial peer for outbound for this number
with nte 100 defined.
-nick
On Mon, Oct 22, 2012 at 8:39 AM, <[email protected]> wrote:
> I checked to make sure it took the command and it also showed that nte
> 100 was preferred. However, I was not able to navigate any menu with it set
> to this. When I switched it back to the default, the other numbers that
> worked with the default setting started working again.****
>
> ** **
>
> -Bill****
>
> ** **
>
> ** **
>
> *From:* [email protected] [mailto:[email protected]] *On Behalf Of *Nick
> Matthews
> *Sent:* Sunday, October 21, 2012 10:12 PM
> *To:* Tim
> *Cc:* Hendrix, George (Bill) @ NSS - STRATIS; Derek Wyss;
> [email protected]
>
> *Subject:* Re: [cisco-voip] DTMF Issue with one external number****
>
> ** **
>
> Double check that the config took the payload type change to 100. That's
> the default for NSE's and you cannot normally change the payload type value
> unless you change the NSE value to another value first.
>
> Don't have a CLI in front of me, but it's similar to:
> rtp payload-type nse 99
> rtp payload-type nte 100
>
> -nick****
>
> On Fri, Oct 19, 2012 at 2:43 PM, Tim <[email protected]> wrote:****
>
> Yes, I have the same feeling on my side. Same problem, just one external
> number doesn?t work. The far end system is connected on the same carrier
> using SIP Trunks. Doesn?t make much sense, but from what I?ve been told
> the far end system is some sort of AdTran.****
>
> ****
>
> *From:* [email protected] [mailto:[email protected]]
> *Sent:* Friday, October 19, 2012 1:35 PM
> *To:* [email protected]; 'Derek Wyss'****
>
>
> *Cc:* [email protected]
> *Subject:* RE: [cisco-voip] DTMF Issue with one external number****
>
> ****
>
> Nope, still have this same issue. Very frustrating that the issue is only
> to one external number and I can actually see the digits being passed out
> the SIP trunk. And since it works for all other numbers, I really don?t
> see how it could be a carrier issue on our side.****
>
> ****
>
> -Bill****
>
> ****
>
> ****
>
> *From:* [email protected] [mailto:[email protected]]
> *Sent:* Friday, October 19, 2012 11:38 AM
> *To:* Hendrix, George (Bill) @ NSS - STRATIS; 'Derek Wyss'
> *Cc:* [email protected]
> *Subject:* RE: [cisco-voip] DTMF Issue with one external number****
>
> ****
>
> George,****
>
> ****
>
> Did you ever find a fix on this? I believe I?m having a similar issue.****
>
> ****
>
> Thanks****
>
> ****
>
> *From:* [email protected] [
> mailto:[email protected]<[email protected]>]
> *On Behalf Of *[email protected]
> *Sent:* Tuesday, October 16, 2012 9:18 PM
> *To:* Derek Wyss
> *Cc:* [email protected]
> *Subject:* Re: [cisco-voip] DTMF Issue with one external number****
>
> ****
>
> The RTP NTE negotiated is indeed 101. However, when I enter the commands
> to set it to 100, I get the following on the debug.****
>
> ****
>
> Preferred Codec : g711ulaw, bytes :160****
>
> Preferred DTMF relay : rtp-nte****
>
> Preferred NTE payload : 100****
>
> Early Media : No****
>
> Delayed Media : Yes****
>
> Bridge Done : No****
>
> New Media : No****
>
> DSP DNLD Reqd : No****
>
> ****
>
> Stream type : voice-only****
>
> Media line : 1****
>
> State : STREAM_ADDING (2)****
>
> Stream address type : 1****
>
> Callid : -1****
>
> Negotiated Codec : g711ulaw, bytes :160****
>
> Nego. Codec payload : 0 (tx), 0 (rx)****
>
> Negotiated DTMF relay : inband-voice****
>
> Negotiated NTE payload : 0 (tx), 0 (rx)****
>
> Negotiated CN payload : 0****
>
> ****
>
> ****
>
> Bill Hendrix | Network/VOIP Engineer****
>
> *L3 STRATIS** **POWERED BY EXCELLENCE* ****
>
> ****
>
> *From:* Derek Wyss [mailto:[email protected]]
> *Sent:* Tuesday, October 16, 2012 8:40 AM
> *To:* Hendrix, George (Bill) @ NSS - STRATIS
> *Cc:* [email protected]
> *Subject:* Re: [cisco-voip] DTMF Issue with one external number****
>
> ****
>
> What does your SDP show using debug ccsip all?
>
> I've ran into this before where the provider had a different RTP map to
> their IP customers vs their ISDN customers. What I had to do was create
> separate dialpeers for those numbers with a different RTP map. See example
> below:****
>
> v=0****
>
> o=CiscoSystemsSIP-GW-UserAgent 9601 2828 IN IP4 X.X.X.X****
>
> s=SIP Call****
>
> c=IN IP4 X.X.X.X****
>
> t=0 0****
>
> m=audio 16384 RTP/AVP 18 0 101****
>
> c=IN IP4 10.8.2.4****
>
> a=rtpmap:18 G729/8000****
>
> a=fmtp:18 annexb=yes****
>
> a=rtpmap:0 PCMU/8000****
>
> a=rtpmap:101 X-NSE/8000****
>
> a=fmtp:101 192-194****
>
> ****
>
> v=0****
>
> o=Sonus_UAC 16372 5325 IN IP4 X.X.X.X****
>
> s=SIP Media Capabilities****
>
> c=IN IP4 X.X.X.X****
>
> t=0 0****
>
> m=audio 23002 RTP/AVP 18 0 100****
>
> a=rtpmap:18 G729/8000****
>
> a=rtpmap:0 PCMU/8000****
>
> a=rtpmap:100 telephone-event/8000****
>
> a=fmtp:100 0-15****
>
> a=sendrecv****
>
> a=maxptime:20****
>
> ****
>
> As mentioned above the yellow highlighted portion of the SDP?s is where we
> can see a mismatch in our payload type. You can see the telco is sending
> 100 and we are sending 101. To resolve this issue you have to remap the
> rtp payload type for signaling and telephony events. Here is the commands
> I had to run to send 100 as our NTE to match what the provider was
> expecting?****
>
> ****
>
> Lincoln-VG01(config-dial-peer)#rtp payload-type nse 98****
>
> Lincoln-VG01(config-dial-peer)#rtp payload-type nte 100****
>
> Lincoln-VG01(config-dial-peer)#rtp payload-type nse 101****
>
> ****
>
> Because the signaling is defaulted at nte 101 and nse 100 you have to
> remove 100 from nse by assigning it a random unused value before you can
> assign 100 to nte. See this image for the default reserved values:
> https://communities.cisco.com/servlet/JiveServlet/downloadImage/2-5295-2243/450-185/defaultpayloadtype.png
> ****
>
>
> Hope this helps,
>
> Derek
>
> On Tue, Oct 16, 2012 at 7:25 AM, <[email protected]> wrote:****
>
> Hey guys,****
>
> ****
>
> We have an odd issue going on with DTMF. Below is the path to the PSTN.
> ****
>
> ****
>
> CUCM <-> h.323 Gateway <-> SIP Provider.****
>
> ****
>
> The problem is that when we dial into this one external number and press 1
> to select option 1, it doesn?t seem to accept the digit. If we dial into
> that same number from anywhere else, it works fine. Having said that, I
> can dial into other numbers from the system and have no issue at all with
> dtmf. Any ideas of what this issue is?****
>
> ****
>
> Thanks,****
>
> Bill****
>
> ****
>
> ****
>
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip****
>
> ****
>
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip****
>
> ** **
>
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Message: 6
Date: Mon, 22 Oct 2012 20:55:47 -0400
From: Nick Matthews <[email protected]>
To: Scott Voll <[email protected]>
Cc: [email protected]
Subject: Re: [cisco-voip] T.38 vs modem passthrough
Message-ID:
<CAM-K-NrgyLT6y-5GMwH4z=cy_UT=urfesls_wjpowwof0cv...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
They should be able to, yes. They initiate at different times in the fax
call so in theory the first that both support should be used. I've seen
general weirdness, so your mileage may vary. "Overconfiguring" fax is
pretty common, and it should be a bug in many cases, but if it works when
simplified it may be overlooked by development.
-nick
On Mon, Oct 22, 2012 at 4:52 PM, Scott Voll <[email protected]> wrote:
> if everything we have is setup to use t.38 and I need Modem passthrough
> also, Can they both run together?
>
> CM 7.1
> VG224
> MGCP VGW 15.1T code
>
> TIA
>
> Scott
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 7
Date: Tue, 23 Oct 2012 09:46:41 -0500
From: "Madziarczyk, Jonathan" <[email protected]>
To: <[email protected]>
Subject: [cisco-voip] ATA Call Pickup Group or something else?
Message-ID:
<4604610d6aac164aa12b9e03ab79028b01162...@exchange3.local.cityofevanston.org>
Content-Type: text/plain; charset="us-ascii"
I'm trying to figure out what technology I should be using, since there
seem to be a couple of ways to do different parts of this.
Call Manager 6.1
Department Main Number: 5555
Phone A: 7965 sccp
Phone B: Analog on ATA sccp
The Scenario is, 2 phones A & B want to be able to pick up when someone
calls 5555. They also want to be able to put the call on Hold and allow
the other phone to be able to pick up the call.
It seems as if it could be as simple as just putting the same extension
on both phones and learning the codes for an analog phone to put a call
on hold and take off of hold for the same line.
Or it seems like a call pickup group might also be needed.
Ideas? Can phone B pickup a call that phone A put on hold?
Thanks!
JM
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Message: 8
Date: Tue, 23 Oct 2012 15:41:55 +0000
From: Eric Pedersen <[email protected]>
To: Scott Voll <[email protected]>, "[email protected]"
<[email protected]>
Subject: Re: [cisco-voip] T.38 vs modem passthrough
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
If both gateways are MGCP it works great. Modem passthrough kicks in first and
then switches over to T.38 when the GW detects the fax tones.
From: [email protected]
[mailto:[email protected]] On Behalf Of Scott Voll
Sent: 22 October 2012 2:53 PM
To: [email protected]
Subject: [cisco-voip] T.38 vs modem passthrough
if everything we have is setup to use t.38 and I need Modem passthrough also,
Can they both run together?
CM 7.1
VG224
MGCP VGW 15.1T code
TIA
Scott
The contents of this message may contain confidential and/or privileged
subject matter. If this message has been received in error, please contact
the sender and delete all copies. Like other forms of communication,
e-mail communications may be vulnerable to interception by unauthorized
parties. If you do not wish us to communicate with you by e-mail, please
notify us at your earliest convenience. In the absence of such
notification, your consent is assumed. Should you choose to allow us to
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