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Today's Topics:
1. Re: Call not connecting, but it's connected (Eric Pedersen)
2. Tandberg E95 SIP reg (abbas Wali)
3. Re: Call not connecting, but it's connected (Carlo)
4. Unity Connection SMTP domain change (Mike )
5. Re: Unity Connection SMTP domain change (John Van Laecke)
6. Re: UCCX CSS ? (Pavan K)
7. Re: Unity Connection SMTP domain change (Jason Aarons (AM))
8. Call drops after taking off hold (Rynard Coetzee)
9. Re: Call drops after taking off hold (John Van Laecke)
10. Re: Call drops after taking off hold (Rynard Coetzee)
11. Re: Call drops after taking off hold (Abdul Salam .)
12. Re: Call drops after taking off hold (Jason Aarons (AM))
13. Re: Unity Connection SMTP domain change (Chris Ward (chrward))
14. Cisco 8945 Phone Power issue (Ken Rhodes)
15. UC on UCS C-Series Datastore Setup (Boon)
----------------------------------------------------------------------
Message: 1
Date: Mon, 21 Jan 2013 18:59:47 +0000
From: Eric Pedersen <[email protected]>
To: c3voip <[email protected]>, "[email protected]"
<[email protected]>
Subject: Re: [cisco-voip] Call not connecting, but it's connected
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
We call a different company's toll-free number that does the same thing. In
our case DTMF wasn't working at all, but that was with a SIP gateway. I did a
bit of digging, and the remote end played their IVR and expected input after an
ISDN PROGRESS, without ISDN CONNECT. From what I read ISDN PROGRESS is only for
the other party to play a recording, not receive any audio, so the GW didn't
consider the call set up and wouldn't send DTMF. I got the DTMF working by
changing the relay mechanism, but I just tried with Jabber and I experience the
same problem as you do.
Let me know if you get a bug ID and I'll open an SR on it so that it gets
another hit. IMO, it's the IVR that isn't configured properly but if it works
calling from a cell phone, it should work from a Cisco device also.
I was wondering why they would start their IVR that way... you're probably
right about it being toll avoidance.
Eric
From: [email protected]
[mailto:[email protected]] On Behalf Of c3voip
Sent: 16 January 2013 7:53 AM
To: [email protected]
Subject: [cisco-voip] Call not connecting, but it's connected
Hi,
So when we try to call McAfee Gold Support line at 1-800-937-2237, our call is
immediately answered by an auto-attendant, but the PRI channel does not show as
connected, so my phone does not show that the call is connected. This is
obviously some type of toll-charge avoidance. I have tried calling from my
cellphone and the behavior is the same...phone still displays "calling" even
though the AA is connected.
My problem is with trying to use Jabber either to control a hard phone or using
it as a soft phone, it cannot generate touch-tones to navigate the
auto-attendant. In the case of the soft phone, the keypad never becomes
available in the GUI and numbers on the keyboard do nothing.
Has anyone ever seen this and is there any way to get Jabber to work with these
calls?
Thanks,
-Chase
The contents of this message may contain confidential and/or privileged
subject matter. If this message has been received in error, please contact
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parties. If you do not wish us to communicate with you by e-mail, please
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Message: 2
Date: Mon, 21 Jan 2013 19:15:13 +0000
From: abbas Wali <[email protected]>
To: [email protected]
Subject: [cisco-voip] Tandberg E95 SIP reg
Message-ID:
<cafdhcp653+bapy3tvdgpyalll-cx1in4xsc155lod7cni32...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Guys,
having issues registering Tandberg Edge 95 to CUCM 8.5 via SIP as per
attachements.
.75 is one of our Subs.
I have added it on the CM side as a basic 3rd Party SIP device with no
authentication.
not sure about the Display name and the SIP address in the photo 1.
Any Help.
We have though configured it as a H323 client and that works fine. just to
add it as SIP as well for more flex.
Thanks
--
@bbas..
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Message: 3
Date: Mon, 21 Jan 2013 13:50:29 -0800
From: Carlo <[email protected]>
To: [email protected]
Subject: Re: [cisco-voip] Call not connecting, but it's connected
Message-ID: <[email protected]>
Content-Type: text/plain; charset="utf-8"
I don't think this is a bug. I ran into almost the same thing. Try this. on
CUCM int the sip profile, add the "Send PRACK for all 1xx messages". you may
also need to force early offer on the gateway.
Sent from my iPad
On Jan 21, 2013, at 10:59 AM, Eric Pedersen <[email protected]> wrote:
> We call a different company's toll-free number that does the same thing. In
> our case DTMF wasn't working at all, but that was with a SIP gateway. I did a
> bit of digging, and the remote end played their IVR and expected input after
> an ISDN PROGRESS, without ISDN CONNECT. From what I read ISDN PROGRESS is
> only for the other party to play a recording, not receive any audio, so the
> GW didn't consider the call set up and wouldn't send DTMF. I got the DTMF
> working by changing the relay mechanism, but I just tried with Jabber and I
> experience the same problem as you do.
>
> Let me know if you get a bug ID and I'll open an SR on it so that it gets
> another hit. IMO, it's the IVR that isn't configured properly but if it
> works calling from a cell phone, it should work from a Cisco device also.
>
> I was wondering why they would start their IVR that way? you're probably
> right about it being toll avoidance.
>
> Eric
>
> From: [email protected]
> [mailto:[email protected]] On Behalf Of c3voip
> Sent: 16 January 2013 7:53 AM
> To: [email protected]
> Subject: [cisco-voip] Call not connecting, but it's connected
>
> Hi,
>
> So when we try to call McAfee Gold Support line at 1-800-937-2237, our call
> is immediately answered by an auto-attendant, but the PRI channel does not
> show as connected, so my phone does not show that the call is connected.
> This is obviously some type of toll-charge avoidance. I have tried calling
> from my cellphone and the behavior is the same?phone still displays ?calling?
> even though the AA is connected.
>
> My problem is with trying to use Jabber either to control a hard phone or
> using it as a soft phone, it cannot generate touch-tones to navigate the
> auto-attendant. In the case of the soft phone, the keypad never becomes
> available in the GUI and numbers on the keyboard do nothing.
>
> Has anyone ever seen this and is there any way to get Jabber to work with
> these calls?
>
> Thanks,
> -Chase
> 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
> (such as encryption) unless specifically requested.
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
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Message: 4
Date: Mon, 21 Jan 2013 22:13:43 -0500
From: "Mike " <[email protected]>
To: "'cisco voip'" <[email protected]>
Subject: [cisco-voip] Unity Connection SMTP domain change
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"
Hi all. I have an issue where Unity connection 8.6 using single inbox sends
an email with [email protected], but the users real email is
[email protected]
Users want to be able to forward and reply to voicemails in there email
boxes to other users.
I read that changing the SMTP domain is a huge no no. Any ideas on how to
get Unity cx to send with [email protected]? I've created SMTP proxy
addresses for everyone, but it still sends it with the unitycx.domain.com.
Thanks,
Mike
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Message: 5
Date: Tue, 22 Jan 2013 04:07:26 +0000
From: John Van Laecke <[email protected]>
To: "Mike " <[email protected]>, "'cisco voip'"
<[email protected]>
Subject: Re: [cisco-voip] Unity Connection SMTP domain change
Message-ID:
<CF2E16827204E249BB155D2C1F5C1EAB1ABB5386@GLB-EXMBX-001.ghdnet.internal>
Content-Type: text/plain; charset="us-ascii"
Changing the smtp on 8.6 will make your license invalid so you will need to
talk to tac after changing.
From: [email protected]
[mailto:[email protected]] On Behalf Of Mike
Sent: Tuesday, 22 January 2013 1:14 PM
To: 'cisco voip'
Subject: [cisco-voip] Unity Connection SMTP domain change
Hi all. I have an issue where Unity connection 8.6 using single inbox sends an
email with [email protected]<mailto:[email protected]>, but
the users real email is [email protected]<mailto:[email protected]>
Users want to be able to forward and reply to voicemails in there email boxes
to other users.
I read that changing the SMTP domain is a huge no no. Any ideas on how to get
Unity cx to send with [email protected]<mailto:[email protected]>? I've
created SMTP proxy addresses for everyone, but it still sends it with the
unitycx.domain.com.
Thanks,
Mike
_____________________
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This email and all attachments are confidential. For further important
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Message: 6
Date: Mon, 21 Jan 2013 22:01:21 -0600
From: Pavan K <[email protected]>
To: "Countryman, Edward" <[email protected]>
Cc: [email protected]
Subject: Re: [cisco-voip] UCCX CSS ?
Message-ID:
<cajdpbux0ilgd-zebhz4s2svfwb-acbfx9+2ksnt1p33hg16...@mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"
Edward,
Can you check if redirect CSS is a configurable setting under the ccx
triggers. It is there in 8.5, not sure about 7
On Jan 21, 2013 10:15 AM, "Countryman, Edward" <
[email protected]> wrote:
> Does anyone know if it?s possible to have a UCCX application use a
> different CSS for redirecting a call then the one associated with the
> originating party? (We are on version 7.0)****
>
> ** **
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 7
Date: Mon, 21 Jan 2013 23:31:16 -0500
From: "Jason Aarons (AM)" <[email protected]>
To: "Mike " <[email protected]>, "'cisco voip'"
<[email protected]>
Subject: Re: [cisco-voip] Unity Connection SMTP domain change
Message-ID:
<4e38db0a1959b04c8c83edcf069b53ed0d2da02...@usispclexdb01.na.didata.local>
Content-Type: text/plain; charset="windows-1252"
If you are never going to have another CXN cluster and won't need Unity
Networking then change the SMTP domain to match.
Many small customer's will never go beyond a single CXN cluster.
Else it's working as designed by Cisco and talk to your Cisco Account Manager
about a Product Enhancement Request (PERS) :)
From: [email protected]
[mailto:[email protected]] On Behalf Of Mike
Sent: Monday, January 21, 2013 10:14 PM
To: 'cisco voip'
Subject: [cisco-voip] Unity Connection SMTP domain change
Hi all. I have an issue where Unity connection 8.6 using single inbox sends an
email with [email protected]<mailto:[email protected]>, but
the users real email is [email protected]<mailto:[email protected]>
Users want to be able to forward and reply to voicemails in there email boxes
to other users.
I read that changing the SMTP domain is a huge no no. Any ideas on how to get
Unity cx to send with [email protected]<mailto:[email protected]>? I've
created SMTP proxy addresses for everyone, but it still sends it with the
unitycx.domain.com.
Thanks,
Mike
itevomcid
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Message: 8
Date: Tue, 22 Jan 2013 04:55:42 +0000
From: Rynard Coetzee <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] Call drops after taking off hold
Message-ID:
<97D19F256B859A48A6A93E8A9210E52A84327E27@BYTESEXCH2K10N1.bytes.local>
Content-Type: text/plain; charset="us-ascii"
Hi
I have the following scenario at a client site ,call comes from PSTN via H323
GW ,I can answer the call and place the caller on hold ,when I try and resume
the call ,the Telco drops the call with a " Bearer capability not implemented"
error. Any ideas what the problem could be ?
Regards
Rynard
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Message: 9
Date: Tue, 22 Jan 2013 05:44:07 +0000
From: John Van Laecke <[email protected]>
To: Rynard Coetzee <[email protected]>,
"[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Call drops after taking off hold
Message-ID:
<CF2E16827204E249BB155D2C1F5C1EAB1ABB74B4@GLB-EXMBX-001.ghdnet.internal>
Content-Type: text/plain; charset="us-ascii"
Add bearer-cap speed to the voice port on the gateway
From: [email protected]
[mailto:[email protected]] On Behalf Of Rynard Coetzee
Sent: Tuesday, 22 January 2013 2:56 PM
To: [email protected]
Subject: [cisco-voip] Call drops after taking off hold
Hi
I have the following scenario at a client site ,call comes from PSTN via H323
GW ,I can answer the call and place the caller on hold ,when I try and resume
the call ,the Telco drops the call with a " Bearer capability not implemented"
error. Any ideas what the problem could be ?
Regards
Rynard
_____________________
This e-mail has been scanned for viruses by MessageLabs.
_____________________
This email and all attachments are confidential. For further important
information about emails sent to or from GHD or if you have received this email
in error, please refer to http://www.ghd.com/emaildisclaimer.html .
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Message: 10
Date: Tue, 22 Jan 2013 07:29:56 +0000
From: Rynard Coetzee <[email protected]>
To: John Van Laecke <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Call drops after taking off hold
Message-ID: <[email protected]>
Content-Type: text/plain; charset="windows-1252"
I already have bearer-cap speech configured on voice-port
Sent from my iPhone
On 22 Jan 2013, at 7:46 AM, "John Van Laecke"
<[email protected]<mailto:[email protected]>> wrote:
Add bearer-cap speed to the voice port on the gateway
From:
[email protected]<mailto:[email protected]>
[mailto:[email protected]] On Behalf Of Rynard Coetzee
Sent: Tuesday, 22 January 2013 2:56 PM
To: [email protected]<mailto:[email protected]>
Subject: [cisco-voip] Call drops after taking off hold
Hi
I have the following scenario at a client site ,call comes from PSTN via H323
GW ,I can answer the call and place the caller on hold ,when I try and resume
the call ,the Telco drops the call with a ? Bearer capability not implemented?
error. Any ideas what the problem could be ?
Regards
Rynard
_____________________
This e-mail has been scanned for viruses by MessageLabs.
_____________________
This email and all attachments are confidential. For further important
information about emails sent to or from GHD or if you have received this email
in error, please refer to http://www.ghd.com/emaildisclaimer.html .
_____________________
This e-mail has been scanned for viruses by MessageLabs.
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Message: 11
Date: Tue, 22 Jan 2013 13:37:40 +0530
From: "Abdul Salam ." <[email protected]>
To: Rynard Coetzee <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Call drops after taking off hold
Message-ID:
<cakav0xqawaf80_o7zbxxnsysm9vheyuzmudrpampta0hwn_...@mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"
Do u have some media inactivity related configuration ?
*---AS*
On Tue, Jan 22, 2013 at 12:59 PM, Rynard Coetzee <[email protected]
> wrote:
> I already have bearer-cap speech configured on voice-port
>
> Sent from my iPhone
>
> On 22 Jan 2013, at 7:46 AM, "John Van Laecke" <[email protected]>
> wrote:
>
> Add bearer-cap speed to the voice port on the gateway****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> *From:* [email protected] [mailto:
> [email protected]] *On Behalf Of *Rynard Coetzee
> *Sent:* Tuesday, 22 January 2013 2:56 PM
> *To:* [email protected]
> *Subject:* [cisco-voip] Call drops after taking off hold****
>
> ** **
>
> Hi****
>
> I have the following scenario at a client site ,call comes from PSTN via
> H323 GW ,I can answer the call and place the caller on hold ,when I try and
> resume the call ,the Telco drops the call with a ? Bearer capability not
> implemented? error. Any ideas what the problem could be ?****
>
> Regards****
>
> Rynard****
>
>
> _____________________
> This e-mail has been scanned for viruses by MessageLabs.****
>
> _____________________
> This email and all attachments are confidential. For further important
> information about emails sent to or from GHD or if you have received this
> email in error, please refer to http://www.ghd.com/emaildisclaimer.html .
> _____________________
> This e-mail has been scanned for viruses by MessageLabs.
>
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 12
Date: Tue, 22 Jan 2013 07:05:27 -0500
From: "Jason Aarons (AM)" <[email protected]>
To: "Abdul Salam ." <[email protected]>, Rynard Coetzee
<[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Call drops after taking off hold
Message-ID:
<4e38db0a1959b04c8c83edcf069b53ed0d2da02...@usispclexdb01.na.didata.local>
Content-Type: text/plain; charset="windows-1252"
Under the gateway command I would look for inactivity commands.
From: [email protected]
[mailto:[email protected]] On Behalf Of Abdul Salam .
Sent: Tuesday, January 22, 2013 3:08 AM
To: Rynard Coetzee
Cc: [email protected]
Subject: Re: [cisco-voip] Call drops after taking off hold
Do u have some media inactivity related configuration ?
---AS
On Tue, Jan 22, 2013 at 12:59 PM, Rynard Coetzee
<[email protected]<mailto:[email protected]>> wrote:
I already have bearer-cap speech configured on voice-port
Sent from my iPhone
On 22 Jan 2013, at 7:46 AM, "John Van Laecke"
<[email protected]<mailto:[email protected]>> wrote:
Add bearer-cap speed to the voice port on the gateway
From:
[email protected]<mailto:[email protected]>
[mailto:[email protected]<mailto:[email protected]>]
On Behalf Of Rynard Coetzee
Sent: Tuesday, 22 January 2013 2:56 PM
To: [email protected]<mailto:[email protected]>
Subject: [cisco-voip] Call drops after taking off hold
Hi
I have the following scenario at a client site ,call comes from PSTN via H323
GW ,I can answer the call and place the caller on hold ,when I try and resume
the call ,the Telco drops the call with a " Bearer capability not implemented"
error. Any ideas what the problem could be ?
Regards
Rynard
_____________________
This e-mail has been scanned for viruses by MessageLabs.
_____________________
This email and all attachments are confidential. For further important
information about emails sent to or from GHD or if you have received this email
in error, please refer to http://www.ghd.com/emaildisclaimer.html .
_____________________
This e-mail has been scanned for viruses by MessageLabs.
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
itevomcid
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Message: 13
Date: Tue, 22 Jan 2013 14:57:34 +0000
From: "Chris Ward (chrward)" <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>, "Mike "
<[email protected]>, "'cisco voip'" <[email protected]>
Subject: Re: [cisco-voip] Unity Connection SMTP domain change
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Leave your SMTP different. You need it for routing to work correctly.
The way to work around this is to setup an email address policy that associates
the connection email address with the corporate email address and sets the corp
email address as the default. Screenshots attached.
+Chris
Unity Connection TME
From: [email protected]
[mailto:[email protected]] On Behalf Of Jason Aarons (AM)
Sent: Monday, January 21, 2013 11:31 PM
To: Mike ; 'cisco voip'
Subject: Re: [cisco-voip] Unity Connection SMTP domain change
If you are never going to have another CXN cluster and won't need Unity
Networking then change the SMTP domain to match.
Many small customer's will never go beyond a single CXN cluster.
Else it's working as designed by Cisco and talk to your Cisco Account Manager
about a Product Enhancement Request (PERS) :)
From:
[email protected]<mailto:[email protected]>
[mailto:[email protected]] On Behalf Of Mike
Sent: Monday, January 21, 2013 10:14 PM
To: 'cisco voip'
Subject: [cisco-voip] Unity Connection SMTP domain change
Hi all. I have an issue where Unity connection 8.6 using single inbox sends an
email with [email protected]<mailto:[email protected]>, but
the users real email is [email protected]<mailto:[email protected]>
Users want to be able to forward and reply to voicemails in there email boxes
to other users.
I read that changing the SMTP domain is a huge no no. Any ideas on how to get
Unity cx to send with [email protected]<mailto:[email protected]>? I've
created SMTP proxy addresses for everyone, but it still sends it with the
unitycx.domain.com.
Thanks,
Mike
itevomcid
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Message: 14
Date: Tue, 22 Jan 2013 10:04:21 -0500
From: Ken Rhodes <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] Cisco 8945 Phone Power issue
Message-ID: <[email protected]>
Content-Type: text/plain; charset=us-ascii
Wondering if anyone else has run into a situation where there the screen on an
8945 IP Phone goes black (and its not the energy savings feature) and only the
MWI light on the handset is lit as well as the speakerphone, mute, headset, and
video buttons all remain lit.
------------------------------
Message: 15
Date: Tue, 22 Jan 2013 15:45:45 +0000
From: Boon <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] UC on UCS C-Series Datastore Setup
Message-ID:
<CACue4Gg=f5wqhuHyBUvczmPHFb765hm6DdaJAro=n=cchfz...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
I'm in the process of building my first C-Series UCS server from scratch.
I've configured RAID, installed ESXi and have logged into vSphere. I need
to configure a datastore which I'm ok with.
I have a question regarding the datastore setup. Is there any
recommendations from Cisco on how to configure these logically?
I've seen a couple of installs that have two datastores, one for VMs and
another for ISO images etc for quick rebuilds..
I have a 1.9TB of disk space to use and I'd like to set it up in line with
best practice if I can.
Thanks,
Rab
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End of cisco-voip Digest, Vol 111, Issue 21
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