Send cisco-voip mailing list submissions to
[email protected]
To subscribe or unsubscribe via the World Wide Web, visit
https://puck.nether.net/mailman/listinfo/cisco-voip
or, via email, send a message with subject or body 'help' to
[email protected]
You can reach the person managing the list at
[email protected]
When replying, please edit your Subject line so it is more specific
than "Re: Contents of cisco-voip digest..."
Today's Topics:
1. Re: H.323 DTMF to SIP 2833 (Divin John (dijohn))
2. Time of Day routing and dates/times in same time schedule
(Erick B.)
3. Agent Desk Setting/WorkFlow Reason Codes Not Associating to
Agent in CAD | IPCC/UCCE Version: 7.5 (Ryan Burtch)
4. Re: H.323 DTMF to SIP 2833 (Mark Holloway)
5. Re: H.323 DTMF to SIP 2833 (Chris Ward (chrward))
6. Re: H.323 DTMF to SIP 2833 (Mark Holloway)
7. Re: H.323 DTMF to SIP 2833 (Chris Ward (chrward))
8. prefix digits for outbound calls (Erick Wellnitz)
----------------------------------------------------------------------
Message: 1
Date: Tue, 16 Jul 2013 17:29:43 +0000
From: "Divin John (dijohn)" <[email protected]>
To: Nate VanMaren <[email protected]>,
"[email protected]" <[email protected]>
Subject: Re: [cisco-voip] H.323 DTMF to SIP 2833
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
CUCM on the H323 leg doesn't support rfc2833 (rtp-nte).
So, to stop CUCM from invoking an MTP, the SIP side should support Out of band
(notify maybe)
Regards,
Divin
From: Nate VanMaren <[email protected]<mailto:[email protected]>>
Date: Tuesday, 11 June 2013 1:26 AM
To: "[email protected]<mailto:[email protected]>"
<[email protected]<mailto:[email protected]>>
Subject: [cisco-voip] H.323 DTMF to SIP 2833
What H.323 DTMF method can I use on a H.323 dial-peer to CUCM that will not
invoke an MTP when the outbound side of CUCM is a SIP Trunk that is configured
to use 2833.
I am currently using h245-alphanumeric, which seems to be invoking an MTP.
Thanks,
-Nate
NOTICE: This email message is for the sole use of the intended recipient(s) and
may contain confidential and privileged information. Any unauthorized review,
use, disclosure or distribution is prohibited. If you are not the intended
recipient, please contact the sender by reply email and destroy all copies of
the original message.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<https://puck.nether.net/pipermail/cisco-voip/attachments/20130716/342fb804/attachment-0001.html>
------------------------------
Message: 2
Date: Tue, 16 Jul 2013 10:27:12 -0500
From: "Erick B." <[email protected]>
To: voip puck <[email protected]>
Subject: [cisco-voip] Time of Day routing and dates/times in same time
schedule
Message-ID:
<cahsnbqxorysgzu1temhtrmpv1ndaq0gfhkk_ue1rerne2ic...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
On CUCM 8.62a SU2 and having issue with time of day routing not functioning
as expected.
We basically are using this for open and closed hour partitions to send
calls to a ARC during open hours and to an afterhours number during closed
hours. There are multiple sites using ARC so the open/close hours in ARC
don't work out and some have a different afterhours number.
The hours work fine and the calls route fine.
Once we add a date time period to the time schedule for holidays it does
not send the call to the afterhours number on the date. For example we
added time period called July4 to the time schedule and the July4 time
period is a repeating date from July 4th to July 4th with hours 00:00 to
24:00, I also tried 00:15 to 23:45.
The holiday date time period is added to the Closed hours and the closed
partition is listed first in the CSS.
I have moved this to unity to handle for now but would like the call to
stay on the Call Manager if possible.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<https://puck.nether.net/pipermail/cisco-voip/attachments/20130716/0c1b2f8b/attachment-0001.html>
------------------------------
Message: 3
Date: Tue, 16 Jul 2013 22:12:09 -0400
From: Ryan Burtch <[email protected]>
To: voip puck <[email protected]>
Subject: [cisco-voip] Agent Desk Setting/WorkFlow Reason Codes Not
Associating to Agent in CAD | IPCC/UCCE Version: 7.5
Message-ID:
<CAK+Shf6WgLrRKepJhoT9jjt_0nP0hZHbt8xYX1F=ufx1fug...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
All:
I have created a new Agent Desk Setting and a Workflow in Cisco Desktop
Administrator, both with the same name. The agent that I'm testing with
has been assigned to the respective Desk setting. In Cisco Desktop
Administrator, I added several additional Not Ready and Logout Reason
Codes, but they aren't associating to the agent when I log into CAD.
Anyone have any ideas?
Sincerely,
Ryan Burtch
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<https://puck.nether.net/pipermail/cisco-voip/attachments/20130716/ae81010a/attachment-0001.html>
------------------------------
Message: 4
Date: Wed, 17 Jul 2013 08:13:26 -0400
From: Mark Holloway <[email protected]>
To: Divin John (dijohn) <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] H.323 DTMF to SIP 2833
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"
John is correct, if you're using Alphanumeric on H323 you should be able to use
SIP INFO without the need of an MTP. The question is what's on the other end of
the SIP Trunk, and will they support SIP INFO?
DTMF interworking between SIP and H.323 in the signaling plane, using the
Alphanumeric method
of UserInputIndication
On Jul 16, 2013, at 1:29 PM, Divin John (dijohn) <[email protected]> wrote:
> CUCM on the H323 leg doesn't support rfc2833 (rtp-nte).
> So, to stop CUCM from invoking an MTP, the SIP side should support Out of
> band (notify maybe)
>
> Regards,
> Divin
>
> From: Nate VanMaren <[email protected]>
> Date: Tuesday, 11 June 2013 1:26 AM
> To: "[email protected]" <[email protected]>
> Subject: [cisco-voip] H.323 DTMF to SIP 2833
>
>
>
> What H.323 DTMF method can I use on a H.323 dial-peer to CUCM that will not
> invoke an MTP when the outbound side of CUCM is a SIP Trunk that is
> configured to use 2833.
>
> I am currently using h245-alphanumeric, which seems to be invoking an MTP.
>
> Thanks,
> -Nate
>
>
> NOTICE: This email message is for the sole use of the intended recipient(s)
> and may contain confidential and privileged information. Any unauthorized
> review, use, disclosure or distribution is prohibited. If you are not the
> intended recipient, please contact the sender by reply email and destroy all
> copies of the original message.
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<https://puck.nether.net/pipermail/cisco-voip/attachments/20130717/b81c5dd0/attachment-0001.html>
------------------------------
Message: 5
Date: Wed, 17 Jul 2013 13:17:08 +0000
From: "Chris Ward (chrward)" <[email protected]>
To: Mark Holloway <[email protected]>, "Divin John (dijohn)"
<[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] H.323 DTMF to SIP 2833
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
CUCM support RFC2833 so long as one of the legs is SIP. An H323 to SIP call
through CUCM should allow RFC2833.
+Chris
Unity Connection TME
From: cisco-voip [mailto:[email protected]] On Behalf Of Mark
Holloway
Sent: Wednesday, July 17, 2013 8:13 AM
To: Divin John (dijohn)
Cc: [email protected]
Subject: Re: [cisco-voip] H.323 DTMF to SIP 2833
John is correct, if you're using Alphanumeric on H323 you should be able to use
SIP INFO without the need of an MTP. The question is what's on the other end of
the SIP Trunk, and will they support SIP INFO?
DTMF interworking between SIP and H.323 in the signaling plane, using the
Alphanumeric method
of UserInputIndication
On Jul 16, 2013, at 1:29 PM, Divin John (dijohn)
<[email protected]<mailto:[email protected]>> wrote:
CUCM on the H323 leg doesn't support rfc2833 (rtp-nte).
So, to stop CUCM from invoking an MTP, the SIP side should support Out of band
(notify maybe)
Regards,
Divin
From: Nate VanMaren <[email protected]<mailto:[email protected]>>
Date: Tuesday, 11 June 2013 1:26 AM
To: "[email protected]<mailto:[email protected]>"
<[email protected]<mailto:[email protected]>>
Subject: [cisco-voip] H.323 DTMF to SIP 2833
What H.323 DTMF method can I use on a H.323 dial-peer to CUCM that will not
invoke an MTP when the outbound side of CUCM is a SIP Trunk that is configured
to use 2833.
I am currently using h245-alphanumeric, which seems to be invoking an MTP.
Thanks,
-Nate
NOTICE: This email message is for the sole use of the intended recipient(s) and
may contain confidential and privileged information. Any unauthorized review,
use, disclosure or distribution is prohibited. If you are not the intended
recipient, please contact the sender by reply email and destroy all copies of
the original message.
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<https://puck.nether.net/pipermail/cisco-voip/attachments/20130717/b842f34c/attachment-0001.html>
------------------------------
Message: 6
Date: Wed, 17 Jul 2013 11:08:15 -0400
From: Mark Holloway <[email protected]>
To: Chris Ward (chrward) <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] H.323 DTMF to SIP 2833
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"
Wouldn't this still require MTP? It looks like he is trying to avoid using MTP
for the conversion.
On Jul 17, 2013, at 9:17 AM, Chris Ward (chrward) <[email protected]> wrote:
> CUCM support RFC2833 so long as one of the legs is SIP. An H323 to SIP call
> through CUCM should allow RFC2833.
>
> +Chris
> Unity Connection TME
>
> From: cisco-voip [mailto:[email protected]] On Behalf Of
> Mark Holloway
> Sent: Wednesday, July 17, 2013 8:13 AM
> To: Divin John (dijohn)
> Cc: [email protected]
> Subject: Re: [cisco-voip] H.323 DTMF to SIP 2833
>
> John is correct, if you're using Alphanumeric on H323 you should be able to
> use SIP INFO without the need of an MTP. The question is what's on the other
> end of the SIP Trunk, and will they support SIP INFO?
>
> DTMF interworking between SIP and H.323 in the signaling plane, using the
> Alphanumeric method
> of UserInputIndication
>
>
> On Jul 16, 2013, at 1:29 PM, Divin John (dijohn) <[email protected]> wrote:
>
>
> CUCM on the H323 leg doesn't support rfc2833 (rtp-nte).
> So, to stop CUCM from invoking an MTP, the SIP side should support Out of
> band (notify maybe)
>
> Regards,
> Divin
>
> From: Nate VanMaren <[email protected]>
> Date: Tuesday, 11 June 2013 1:26 AM
> To: "[email protected]" <[email protected]>
> Subject: [cisco-voip] H.323 DTMF to SIP 2833
>
>
>
> What H.323 DTMF method can I use on a H.323 dial-peer to CUCM that will not
> invoke an MTP when the outbound side of CUCM is a SIP Trunk that is
> configured to use 2833.
>
> I am currently using h245-alphanumeric, which seems to be invoking an MTP.
>
> Thanks,
> -Nate
>
>
> NOTICE: This email message is for the sole use of the intended recipient(s)
> and may contain confidential and privileged information. Any unauthorized
> review, use, disclosure or distribution is prohibited. If you are not the
> intended recipient, please contact the sender by reply email and destroy all
> copies of the original message.
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<https://puck.nether.net/pipermail/cisco-voip/attachments/20130717/7cee6058/attachment-0001.html>
------------------------------
Message: 7
Date: Wed, 17 Jul 2013 15:18:53 +0000
From: "Chris Ward (chrward)" <[email protected]>
To: Mark Holloway <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] H.323 DTMF to SIP 2833
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Since its in-band, it wouldn't require an MTP... The 2833 would flow as part of
the RTP stream.
Adding rtp-nte to the dial peer should remove the need for an MTP if the other
leg (from CUCM's perspective at least) is SIP.
+Chris
Unity Connection TME
From: Mark Holloway [mailto:[email protected]]
Sent: Wednesday, July 17, 2013 11:08 AM
To: Chris Ward (chrward)
Cc: Divin John (dijohn); [email protected]
Subject: Re: [cisco-voip] H.323 DTMF to SIP 2833
Wouldn't this still require MTP? It looks like he is trying to avoid using MTP
for the conversion.
On Jul 17, 2013, at 9:17 AM, Chris Ward (chrward)
<[email protected]<mailto:[email protected]>> wrote:
CUCM support RFC2833 so long as one of the legs is SIP. An H323 to SIP call
through CUCM should allow RFC2833.
+Chris
Unity Connection TME
From: cisco-voip
[mailto:[email protected]<mailto:[email protected]>]
On Behalf Of Mark Holloway
Sent: Wednesday, July 17, 2013 8:13 AM
To: Divin John (dijohn)
Cc: [email protected]<mailto:[email protected]>
Subject: Re: [cisco-voip] H.323 DTMF to SIP 2833
John is correct, if you're using Alphanumeric on H323 you should be able to use
SIP INFO without the need of an MTP. The question is what's on the other end of
the SIP Trunk, and will they support SIP INFO?
DTMF interworking between SIP and H.323 in the signaling plane, using the
Alphanumeric method
of UserInputIndication
On Jul 16, 2013, at 1:29 PM, Divin John (dijohn)
<[email protected]<mailto:[email protected]>> wrote:
CUCM on the H323 leg doesn't support rfc2833 (rtp-nte).
So, to stop CUCM from invoking an MTP, the SIP side should support Out of band
(notify maybe)
Regards,
Divin
From: Nate VanMaren <[email protected]<mailto:[email protected]>>
Date: Tuesday, 11 June 2013 1:26 AM
To: "[email protected]<mailto:[email protected]>"
<[email protected]<mailto:[email protected]>>
Subject: [cisco-voip] H.323 DTMF to SIP 2833
What H.323 DTMF method can I use on a H.323 dial-peer to CUCM that will not
invoke an MTP when the outbound side of CUCM is a SIP Trunk that is configured
to use 2833.
I am currently using h245-alphanumeric, which seems to be invoking an MTP.
Thanks,
-Nate
NOTICE: This email message is for the sole use of the intended recipient(s) and
may contain confidential and privileged information. Any unauthorized review,
use, disclosure or distribution is prohibited. If you are not the intended
recipient, please contact the sender by reply email and destroy all copies of
the original message.
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<https://puck.nether.net/pipermail/cisco-voip/attachments/20130717/50826305/attachment-0001.html>
------------------------------
Message: 8
Date: Wed, 17 Jul 2013 10:22:12 -0500
From: Erick Wellnitz <[email protected]>
To: cisco-voip <[email protected]>
Subject: [cisco-voip] prefix digits for outbound calls
Message-ID:
<CAK0wOsDpZiO=zj94t-_vli4weu7c25nb2ebi6dwhdv76_h5...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
I'm seeing some odd behavior in my route lists for prefixing digits.
If I prefix digits at the route pattern level everything works fine. If I
try to prefix digits at the route list level (on each route group assigned
to the route list) the digits are not prefixed.
My pattern is 9011.44XXXXXXXXXX
I strip predot on the route pattern then I prefix + and the call routes
over our sip trunk as expected.
If I strip predot on the route pattern and prefix + at the RL level the +
is not prefixed and the call fails.
My educated guess is that perhaps I should not strip predot at the pattern
level nad instead do it at the RL level.
I am unable to do much testing until close of business.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<https://puck.nether.net/pipermail/cisco-voip/attachments/20130717/33aa1ee4/attachment-0001.html>
------------------------------
Subject: Digest Footer
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip
------------------------------
End of cisco-voip Digest, Vol 117, Issue 14
*******************************************
