[cisco-voip] Resource bug - CSCsq66414 - 6.1(2.1000.13)
Robert Knapp
robert.knapp at spanlink.com
Wed Oct 8 19:07:16 EDT 2008
Check for routing loops with CUPS (if present), had similar issue last week (code yellow)
Thanks,
Robert Knapp
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Today's Topics:
1. Resource bug - CSCsq66414 - 6.1(2.1000.13) (Brad Ellis)
2. Re: Cannot get more than 4 calls to the PSTN (FrogOnDSCP46EF)
3. Re: Ringing Cut Through (STEVEN CASPER)
----------------------------------------------------------------------
Message: 1
Date: Wed, 8 Oct 2008 15:13:22 -0700
From: "Brad Ellis" <brad at ccbootcamp.com>
Subject: [cisco-voip] Resource bug - CSCsq66414 - 6.1(2.1000.13)
To: "Cisco VoIPoE List" <cisco-voip at puck.nether.net>
Message-ID:
<E76ACA8C13EF1847BC69538B522C36630143EDB5 at nlimail.nlidomain>
Content-Type: text/plain; charset="us-ascii"
Hi! I am running into this bug at a customer site. It's taking down
their entire UC system. The UC runs out of resources and cant handle
calls.
http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method
=fetchBugDetails&bugId=CSCsq66414
Has anyone else run into this? It's an 800 phone system w/ a sub and
pub. It's causing major customer sat issues.
thanks,
Brad Ellis
CCIE#5796 (R&S / Security)
CCSI# 30482
CEO / President
CCBOOTCAMP - A Cisco Sponsored Organization (SO)
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Message: 2
Date: Thu, 9 Oct 2008 09:46:28 +1100
From: FrogOnDSCP46EF <ciscoboy2006 at gmail.com>
Subject: Re: [cisco-voip] Cannot get more than 4 calls to the PSTN
To: cisco-voip at puck.nether.net
Message-ID:
<92ed69a80810081546s20ab9c7fla1ed191d19446151 at mail.gmail.com>
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It's definitely DSP issue. Increase your DSP's mate.
-frog
>
> Louis Koekemoer
>
> Cisco IPT Systems Engineer
>
> Dimension Data
>
> Tel. +27 (11) 575 6560
>
> Mob. +27 (76) 608 7221
>
> Fax. +27 (11) 576 6560
>
> Louis.Koekemoer at za.didata.com<http://Louis.Koekemoer@za.didata.com>
>
>
>
> P Before printing I think about the environment
>
>
>
> For more information about Dimension Data, please go to
> www.dimensiondata.com<http://www.dimensiondata.com/>
>
>
>
> From: smithsonianwa at gmail.com<mailto:smithsonianwa at gmail.com> [mailto:
> smithsonianwa at gmail.com<mailto:smithsonianwa at gmail.com>] On Behalf Of Tim
> Smith
> Sent: 08 October 2008 05:07 PM
> To: Louis Koekemoer (ZA)
> Cc: cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
> Subject: Re: [cisco-voip] Cannot get more than 4 calls to the PSTN
>
>
>
> Hi,
>
>
>
> Have you also configured your DSP's for other things like transcoding /
> conferencing / mtp?
>
>
>
> Are all the outbound calls definitely G711?
>
>
>
> What gateway is it?
>
>
>
> Cheers,
>
>
>
> Tim
>
>
>
>
>
>
> On Wed, Oct 8, 2008 at 4:04 PM, Louis Koekemoer (ZA) <
> Louis.Koekemoer at za.didata.com<mailto:Louis.Koekemoer at za.didata.com>>
> wrote:
>
>
>
> Guys,
>
>
>
> I need some advice. I have a customer with Multiple(about 5 and growing)
> SRST sites hanging of CallManager 6.x. They have WAN connectivity via ADSL,
> and a BRI port for redundancy. They then have 4 x 2 port BRI cards for
> Voice. I have a PVDM2-32 in each gateway. I cannot get more than 4 calls to
> the PSTN on any one site. 5th call gives me the following error: "Cause i =
> 0x80AF - Resource unavailable, unspecified"
>
>
>
> Any ideas?
>
>
>
>
>
> Louis Koekemoer
>
> Cisco IPT Systems Engineer
>
> Dimension Data
>
> Tel. +27 (11) 575 6560
>
> Mob. +27 (76) 608 7221
>
> Fax. +27 (11) 576 6560
>
> Louis.Koekemoer at za.didata.com<http://Louis.Koekemoer@za.didata.com/>
>
>
>
> P Before printing I think about the environment
>
>
>
> For more information about Dimension Data, please go to
> www.dimensiondata.com<http://www.dimensiondata.com/>
>
>
>
> This email and all contents are subject to the following disclaimer:
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--
Smile, you'll save someone else's day!
Frog
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Message: 3
Date: Wed, 8 Oct 2008 19:01:33 -0400
From: STEVEN CASPER <SCASPER at mtb.com>
Subject: Re: [cisco-voip] Ringing Cut Through
To: "Voice Noob" <voicenoob at gmail.com>
Cc: cisco-voip at puck.nether.net
Message-ID: <48ED038E.9874.0039.0 at mtb.com>
Content-Type: text/plain; charset="us-ascii"
Thanks! That got the ring on the phones starting with the second caller ringback.
Steve
>>> "Voice Noob" <voicenoob at gmail.com> 10/8/2008 12:03 PM >>>
IF they are on FXO port remove the caller-id enable command. The Caller-id is sent during the ringing setup on FXO lines and it is timing out waiting for the information before it sends the call to your IP phone.
On Wed, Oct 8, 2008 at 10:53 AM, STEVEN CASPER <SCASPER at mtb.com> wrote:
We have noticed that when we call a location from an outside line, before the phone in the site rings, the customer has heard 2.5 rings on their phone. Is there any way to reduce the time for the cut through of the ringing? This a site with analog lines on FXOs in a 2811 using H323 talking to CUCM 6.1.1
Thanks!
Steve
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