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Today's Topics:
1. Calling Name lost inbound from PSTN over Intercluster Trunk
with QSIG (Linsemier, Matthew)
2. Re: 7965's running SCCP45.9-3-1-1S auto dial bug (Erich Novak)
3. Re: Calling Name lost inbound from PSTN over Intercluster
Trunk with QSIG (Ryan Ratliff)
4. Re: Calling Name lost inbound from PSTN over Intercluster
Trunk with QSIG (Linsemier, Matthew)
5. Re: Calling Name lost inbound from PSTN over Intercluster
Trunk with QSIG (Jason Aarons (AM))
6. Re: Calling Name lost inbound from PSTN over Intercluster
Trunk with QSIG (Linsemier, Matthew)
7. Unity 8.5.1 (Cristina Petre)
8. Re: Unity 8.5.1 (Ed Leatherman)
9. unified attendant console (Erick Wellnitz)
10. Re: unified attendant console
(Jamie Gale -X (jamgale - Arc Solutions at Cisco))
11. Re: Unity 8.5.1 (Cristina Petre)
12. Re: Unity 8.5.1 (Ed Leatherman)
13. Re: Unity 8.5.1 (Cristina Petre)
----------------------------------------------------------------------
Message: 1
Date: Sun, 28 Oct 2012 13:26:42 -0700
From: "Linsemier, Matthew" <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] Calling Name lost inbound from PSTN over
Intercluster Trunk with QSIG
Message-ID:
<11191743fbfab44995641ce6e5fa2e7f04601...@npexchmb102.tdc.internal>
Content-Type: text/plain; charset="utf-8"
All,
I asked this question back in 2008 and I am wondering if anything has changed
since then.
I have two Unified Communication clusters, one on 6.1 (existing) and a new one
on 8.6 (new greenfield cluster). They are currently connected via a
Intercluster Trunk using QSIG. Everything is working well across the trunk
from internal calling name and number as well as MWI.
The issue we are having is as we move users from the old cluster to the new
cluster, inbound calls from the PSTN on the old cluster (via MGCP gateways that
are still registered there) destined for users on the new cluster will lose
calling name. This means that during the migration (which will be going on for
3-4 weeks), users on the new cluster will lose Calling Name from outside PSTN
users.
PSTN GW ? MGCP --> UC6.1 <--- ICT ---> UC8.6 --- Phone
Is there any way to fix this?
Sincerely,
[cid:[email protected]]
Matthew M. Linsemier, CCNP / CCDP
Senior Network Engineer
The Doctors Company
? 517.324.6695
? [email protected]
Confidentiality Notice: This message and any attachments hereto may contain
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Message: 2
Date: Mon, 29 Oct 2012 10:45:18 +0000
From: Erich Novak <[email protected]>
To: Aaron Riemer <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] 7965's running SCCP45.9-3-1-1S auto dial bug
Message-ID: <ccb41c41.311e7%[email protected]>
Content-Type: text/plain; charset="us-ascii"
Hi Aaron,
have you found something out about this issue ? My TAC Engineer is not finding
anything, and to make it even worse, we're only hitting this issue when using a
non standard locale.
brgds
Erich
Von: Aaron Riemer <[email protected]<mailto:[email protected]>>
Datum: Samstag, 11. August 2012 03:27
An: 'Tim Smith' <[email protected]<mailto:[email protected]>>
Cc: VoIP List Cisco
<[email protected]<mailto:[email protected]>>
Betreff: Re: [cisco-voip] 7965's running SCCP45.9-3-1-1S auto dial bug
No the calls are always auto dialled once the speaker button is pressed or
handset lifted.
I will let you know the outcome with TAC
Cheers,
Aaron.
From: [email protected]<mailto:[email protected]> [mailto:[email protected]] On
Behalf Of Tim Smith
Sent: Friday, 10 August 2012 11:05 AM
To: Aaron Riemer
Cc: JP Senior; [email protected]<mailto:[email protected]>
Subject: Re: [cisco-voip] 7965's running SCCP45.9-3-1-1S auto dial bug
Hi Guys,
In any of your cases did the phones require stimulus? i.e. the speaker button,
or did they just decide to randomly dial numbers at other times by themselves?
I have a 9971 (SIP) on CME, that decides to make calls by itself at random
times.
Completely different call control and protocol, but just curious really.
On the firmware. Ask TAC if the fix in 9.2(1)S is included in the version of
firmware you are running.
Cheers,
Tim
On 9 August 2012 21:10, Aaron Riemer
<[email protected]<mailto:[email protected]>> wrote:
Wow ok. We are running the very latest firmware though and I don't want to
roll back.
I think we need to go to 8.6 to resolve perhaps.
Thanks for posting.
Aaron.
-----Original Message-----
From: JP Senior
[mailto:[email protected]<mailto:[email protected]>]
Sent: Thursday, 9 August 2012 1:11 AM
To: Aaron Riemer; [email protected]<mailto:[email protected]>
Subject: RE: [cisco-voip] 7965's running SCCP45.9-3-1-1S auto dial bug
Yes, I have had the identical problem here on 7975G and CUCM 8.0.3 - random
users would pick up the phone, it would auto-dial the most recent caller in
the call history list. The fix was to move us to to 9.1(2)S. We're now
running 9.2(1)S in a stable environment on 8.6.
Cisco TAC quoted me CSCtj76712 - "Call is auto dialed when pressing speaker
in an idle phone".
As you mentioned, the only workaround was for a user to reset their own
phone.
Sometimes a hard reset (pulling power) would last a few days without the
problem compared to a soft-reset.
-JP Senior
From:
[email protected]<mailto:[email protected]>
[mailto:[email protected]<mailto:[email protected]>]
On Behalf Of Aaron Riemer
Sent: 06 August 2012 7:57 PM
To: [email protected]<mailto:[email protected]>
Subject: [cisco-voip] 7965's running SCCP45.9-3-1-1S auto dial bug
Hi Guys,
Has anyone yet run into a bug in the new SCCP firmware (SCCP45.9-3-1-1S)
where randomly a phone will attempt to auto dial a number when the handset
is lifted / speaker button is pressed? Looking at the directory it was a
number in the missed call list (5th in the list to be precise).
I have seen this with my own eyes and can confirm the directory is not open.
The only fix seems to be a reset of the phone.
We have had multiple phones report of this issue but so far we can't
manually replicate the problem.
We are running CUCM 8.5 and 7965's.
I found a similar bug that I hope has not been brought back to life with the
new code.
CSCtz29414 Bug Details
7942/7962 - Call History is Auto-Dialed When Pressing Speaker Button
Symptom:
When a 7942/62 phone is populated with call history records (e.g.
received/placed/missed calls), the endpoint will auto-dial the number
contained in its call history when the speaker button is pressed.
Conditions:
IP Phone Call History contains Received/Placed/Missed Phone number record
prior to pressing speaker button to go off-hook. Phone will continue to
auto-dial the number listed in Call History once error condition is
experienced.
Workaround:
Reset IP Phone to clear error condition
1st Found-In
9.2(3)
Fixed-In
9.2(3)ES7
9.3(1)TH2.2
9.2(3)MN1.22
Anyone had similar issues? Would be great to hear from you.
Cheers,
Aaron.
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Message: 3
Date: Mon, 29 Oct 2012 10:30:04 -0400
From: Ryan Ratliff <[email protected]>
To: "Linsemier, Matthew" <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Calling Name lost inbound from PSTN over
Intercluster Trunk with QSIG
Message-ID: <[email protected]>
Content-Type: text/plain; charset="windows-1252"
What was the answer you got in 2008? In order for calling name to be
delivered to the remote cluster the originating cluster is going to have to
send it, and the remote is going to have to honor it.
How is the name delivered on the 6.1 side and what do ccm traces show for what
it sends on to the 8.6 cluster?
-Ryan
On Oct 28, 2012, at 4:26 PM, "Linsemier, Matthew" <[email protected]>
wrote:
All,
I asked this question back in 2008 and I am wondering if anything has changed
since then.
I have two Unified Communication clusters, one on 6.1 (existing) and a new one
on 8.6 (new greenfield cluster). They are currently connected via a
Intercluster Trunk using QSIG. Everything is working well across the trunk
from internal calling name and number as well as MWI.
The issue we are having is as we move users from the old cluster to the new
cluster, inbound calls from the PSTN on the old cluster (via MGCP gateways that
are still registered there) destined for users on the new cluster will lose
calling name. This means that during the migration (which will be going on for
3-4 weeks), users on the new cluster will lose Calling Name from outside PSTN
users.
PSTN GW ? MGCP --> UC6.1 <--- ICT ---> UC8.6 --- Phone
Is there any way to fix this?
Sincerely,
<image001.gif>
Matthew M. Linsemier, CCNP / CCDP
Senior Network Engineer
The Doctors Company
( 517.324.6695
* [email protected]
Confidentiality Notice: This message and any attachments hereto may contain
confidential and privileged communications or information and/or attorney
client communications or work-product protected by law. The information
contained herein is transmitted for the sole use of the intended recipient(s).
If you are not the intended recipient or designated agent of the recipient of
such information, you are hereby notified that any use, dissemination, copying
or retention of this e-mail or the information contained herein is strictly
prohibited and may subject you to penalties under federal and/or state law. If
you received this e-mail in error, please notify the sender immediately and
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------------------------------
Message: 4
Date: Mon, 29 Oct 2012 07:39:52 -0700
From: "Linsemier, Matthew" <[email protected]>
To: Ryan Ratliff <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Calling Name lost inbound from PSTN over
Intercluster Trunk with QSIG
Message-ID: <[email protected]>
Content-Type: text/plain; charset="utf-8"
Ryan,
The answer in 2008 from Wes had to do with setting up a phone in a corner with
a xxxx extension in different CSS and then answered the calls on the 6.1 side
then did a CFA via over the ICT to the other cluster. From what I understand
the way that it comes in from the PSTN on the 6.1 server and how the UC acts is
that it is immediatly processed by the route pattern to the ICT and the 6.1
cluster does not process the Calling Name and it gets lost in transit.
What traces should I take a look at? I thought perhaps others had seen this
type of behavior before.
Matt
Sent from my iPad
On Oct 29, 2012, at 10:30 AM, "Ryan Ratliff"
<[email protected]<mailto:[email protected]>> wrote:
What was the answer you got in 2008? In order for calling name to be
delivered to the remote cluster the originating cluster is going to have to
send it, and the remote is going to have to honor it.
How is the name delivered on the 6.1 side and what do ccm traces show for what
it sends on to the 8.6 cluster?
-Ryan
On Oct 28, 2012, at 4:26 PM, "Linsemier, Matthew"
<[email protected]<mailto:[email protected]>> wrote:
All,
I asked this question back in 2008 and I am wondering if anything has changed
since then.
I have two Unified Communication clusters, one on 6.1 (existing) and a new one
on 8.6 (new greenfield cluster). They are currently connected via a
Intercluster Trunk using QSIG. Everything is working well across the trunk
from internal calling name and number as well as MWI.
The issue we are having is as we move users from the old cluster to the new
cluster, inbound calls from the PSTN on the old cluster (via MGCP gateways that
are still registered there) destined for users on the new cluster will lose
calling name. This means that during the migration (which will be going on for
3-4 weeks), users on the new cluster will lose Calling Name from outside PSTN
users.
PSTN GW ? MGCP --> UC6.1 <--- ICT ---> UC8.6 --- Phone
Is there any way to fix this?
Sincerely,
<image001.gif>
Matthew M. Linsemier, CCNP / CCDP
Senior Network Engineer
The Doctors Company
? 517.324.6695
? [email protected]<mailto:[email protected]>
Confidentiality Notice: This message and any attachments hereto may contain
confidential and privileged communications or information and/or attorney
client communications or work-product protected by law. The information
contained herein is transmitted for the sole use of the intended recipient(s).
If you are not the intended recipient or designated agent of the recipient of
such information, you are hereby notified that any use, dissemination, copying
or retention of this e-mail or the information contained herein is strictly
prohibited and may subject you to penalties under federal and/or state law. If
you received this e-mail in error, please notify the sender immediately and
permanently delete this e-mail.
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
Confidentiality Notice: This message and any attachments hereto may contain
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------------------------------
Message: 5
Date: Mon, 29 Oct 2012 10:43:00 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: Ryan Ratliff <[email protected]>, "Linsemier, Matthew"
<[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Calling Name lost inbound from PSTN over
Intercluster Trunk with QSIG
Message-ID:
<4e38db0a1959b04c8c83edcf069b53ed0d2b536...@usispclexdb01.na.didata.local>
Content-Type: text/plain; charset="windows-1252"
Depending on switch/telco could also be a Facility IE vs Display IE issue, I
would pull debug q931/debug ccm-manager backhaul packets/ CCM traces and look.
Do 7900 phones on both sides show name on incoming call from PSTN? It's just
the ICT that isn't passing name across. Does the ICT work 7900 Phone to 7900
phone with name delivery?
From: [email protected]
[mailto:[email protected]] On Behalf Of Ryan Ratliff
Sent: Monday, October 29, 2012 10:30 AM
To: Linsemier, Matthew
Cc: [email protected]
Subject: Re: [cisco-voip] Calling Name lost inbound from PSTN over Intercluster
Trunk with QSIG
What was the answer you got in 2008? In order for calling name to be
delivered to the remote cluster the originating cluster is going to have to
send it, and the remote is going to have to honor it.
How is the name delivered on the 6.1 side and what do ccm traces show for what
it sends on to the 8.6 cluster?
-Ryan
On Oct 28, 2012, at 4:26 PM, "Linsemier, Matthew"
<[email protected]<mailto:[email protected]>> wrote:
All,
I asked this question back in 2008 and I am wondering if anything has changed
since then.
I have two Unified Communication clusters, one on 6.1 (existing) and a new one
on 8.6 (new greenfield cluster). They are currently connected via a
Intercluster Trunk using QSIG. Everything is working well across the trunk
from internal calling name and number as well as MWI.
The issue we are having is as we move users from the old cluster to the new
cluster, inbound calls from the PSTN on the old cluster (via MGCP gateways that
are still registered there) destined for users on the new cluster will lose
calling name. This means that during the migration (which will be going on for
3-4 weeks), users on the new cluster will lose Calling Name from outside PSTN
users.
PSTN GW - MGCP --> UC6.1 <--- ICT ---> UC8.6 --- Phone
Is there any way to fix this?
Sincerely,
<image001.gif>
Matthew M. Linsemier, CCNP / CCDP
Senior Network Engineer
The Doctors Company
* 517.324.6695
* [email protected]<mailto:[email protected]>
Confidentiality Notice: This message and any attachments hereto may contain
confidential and privileged communications or information and/or attorney
client communications or work-product protected by law. The information
contained herein is transmitted for the sole use of the intended recipient(s).
If you are not the intended recipient or designated agent of the recipient of
such information, you are hereby notified that any use, dissemination, copying
or retention of this e-mail or the information contained herein is strictly
prohibited and may subject you to penalties under federal and/or state law. If
you received this e-mail in error, please notify the sender immediately and
permanently delete this e-mail.
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
itevomcid
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------------------------------
Message: 6
Date: Mon, 29 Oct 2012 07:46:23 -0700
From: "Linsemier, Matthew" <[email protected]>
To: "'Jason Aarons (AM)'" <[email protected]>, Ryan
Ratliff <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Calling Name lost inbound from PSTN over
Intercluster Trunk with QSIG
Message-ID:
<11191743fbfab44995641ce6e5fa2e7f04601...@npexchmb102.tdc.internal>
Content-Type: text/plain; charset="utf-8"
Jason,
All other Calling Name works correctly back and forth over the ICT? so phone to
phone, Unity, Meetingplace Express, UCCX, all works as expected.
Since inbound Calling Name 6.1 cluster works fine to a phone ON that cluster, I
know it works. I suspect that because the way that UC is processing that call
and pushing it directly to the ICT that it doesn?t / can?t process these
features . The ICT with QSIG enabled does not support setting the Facility /
Display IE features.
Matt
From: Jason Aarons (AM) [mailto:[email protected]]
Sent: Monday, October 29, 2012 10:43 AM
To: Ryan Ratliff; Linsemier, Matthew
Cc: [email protected]
Subject: RE: [cisco-voip] Calling Name lost inbound from PSTN over Intercluster
Trunk with QSIG
Depending on switch/telco could also be a Facility IE vs Display IE issue, I
would pull debug q931/debug ccm-manager backhaul packets/ CCM traces and look.
Do 7900 phones on both sides show name on incoming call from PSTN? It?s just
the ICT that isn?t passing name across. Does the ICT work 7900 Phone to 7900
phone with name delivery?
From:
[email protected]<mailto:[email protected]>
[mailto:[email protected]]<mailto:[mailto:[email protected]]>
On Behalf Of Ryan Ratliff
Sent: Monday, October 29, 2012 10:30 AM
To: Linsemier, Matthew
Cc: [email protected]<mailto:[email protected]>
Subject: Re: [cisco-voip] Calling Name lost inbound from PSTN over Intercluster
Trunk with QSIG
What was the answer you got in 2008? In order for calling name to be
delivered to the remote cluster the originating cluster is going to have to
send it, and the remote is going to have to honor it.
How is the name delivered on the 6.1 side and what do ccm traces show for what
it sends on to the 8.6 cluster?
-Ryan
On Oct 28, 2012, at 4:26 PM, "Linsemier, Matthew"
<[email protected]<mailto:[email protected]>> wrote:
All,
I asked this question back in 2008 and I am wondering if anything has changed
since then.
I have two Unified Communication clusters, one on 6.1 (existing) and a new one
on 8.6 (new greenfield cluster). They are currently connected via a
Intercluster Trunk using QSIG. Everything is working well across the trunk
from internal calling name and number as well as MWI.
The issue we are having is as we move users from the old cluster to the new
cluster, inbound calls from the PSTN on the old cluster (via MGCP gateways that
are still registered there) destined for users on the new cluster will lose
calling name. This means that during the migration (which will be going on for
3-4 weeks), users on the new cluster will lose Calling Name from outside PSTN
users.
PSTN GW ? MGCP --> UC6.1 <--- ICT ---> UC8.6 --- Phone
Is there any way to fix this?
Sincerely,
<image001.gif>
Matthew M. Linsemier, CCNP / CCDP
Senior Network Engineer
The Doctors Company
? 517.324.6695
? [email protected]<mailto:[email protected]>
Confidentiality Notice: This message and any attachments hereto may contain
confidential and privileged communications or information and/or attorney
client communications or work-product protected by law. The information
contained herein is transmitted for the sole use of the intended recipient(s).
If you are not the intended recipient or designated agent of the recipient of
such information, you are hereby notified that any use, dissemination, copying
or retention of this e-mail or the information contained herein is strictly
prohibited and may subject you to penalties under federal and/or state law. If
you received this e-mail in error, please notify the sender immediately and
permanently delete this e-mail.
_______________________________________________
cisco-voip mailing list
[email protected]<mailto:[email protected]>
https://puck.nether.net/mailman/listinfo/cisco-voip
itevomcid
Confidentiality Notice: This message and any attachments hereto may contain
confidential and privileged communications or information and/or attorney
client communications or work-product protected by law. The information
contained herein is transmitted for the sole use of the intended recipient(s).
If you are not the intended recipient or designated agent of the recipient of
such information, you are hereby notified that any use, dissemination, copying
or retention of this e-mail or the information contained herein is strictly
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------------------------------
Message: 7
Date: Mon, 29 Oct 2012 16:19:57 +0100
From: Cristina Petre <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] Unity 8.5.1
Message-ID: <[email protected]>
Content-Type: text/plain; charset=us-ascii
Hi all,
If i want to use call handlers in unity how is the connection between cucm and
unity?
The goal should be to dial a number from external to reach the cucm after that
the unity to get a welcome message and after that to go back to cucm to a pilot
point.
I've created a directory number with voice mail pilot and voice mail profile,
on unity I've created a system call handler with a transfer rule back to cucm
and recorded a greeting. It's works but the problem is after I dial the number
I get an "sorry.. Is not available" message after that phones in hunt lists
ringing. I just want to hear my message nothing else.
What's wrong on my configuration? Many thanks in advance.
Von meinem iPhone gesendet
------------------------------
Message: 8
Date: Mon, 29 Oct 2012 11:20:44 -0400
From: Ed Leatherman <[email protected]>
To: Cristina Petre <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Unity 8.5.1
Message-ID:
<cafc4dspdapea3yk2vwk3c3txe3vumbiobp_yefquq6efmgh...@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
The way I do this is setup a pilot number in CUCM and point to your unity
connection line group, and then setup direct routing rule in unity
connection to send the call to the call handler greeting that you want it
to go to.
On Mon, Oct 29, 2012 at 11:19 AM, Cristina Petre
<[email protected]>wrote:
> Hi all,
>
> If i want to use call handlers in unity how is the connection between cucm
> and unity?
> The goal should be to dial a number from external to reach the cucm after
> that the unity to get a welcome message and after that to go back to cucm
> to a pilot point.
>
> I've created a directory number with voice mail pilot and voice mail
> profile, on unity I've created a system call handler with a transfer rule
> back to cucm and recorded a greeting. It's works but the problem is after I
> dial the number I get an "sorry.. Is not available" message after that
> phones in hunt lists ringing. I just want to hear my message nothing else.
>
> What's wrong on my configuration? Many thanks in advance.
>
> Von meinem iPhone gesendet
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
--
Ed Leatherman
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Message: 9
Date: Mon, 29 Oct 2012 10:26:46 -0500
From: Erick Wellnitz <[email protected]>
To: cisco-voip <[email protected]>
Subject: [cisco-voip] unified attendant console
Message-ID:
<cak0wosdegsdvkmucyov4jxzf2q44thvy2tpppuk+rfkrfjm...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Does anyone know what the minimum requirements are for a PC to run the
Unified Attendant Console? I'm having problems locating this information.
Thanks!
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Message: 10
Date: Mon, 29 Oct 2012 15:29:23 +0000
From: "Jamie Gale -X (jamgale - Arc Solutions at Cisco)"
<[email protected]>
To: Erick Wellnitz <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] unified attendant console
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="iso-8859-1"
Erick,
You can refer to the installation guide at the following link for this
information:
http://www.cisco.com/en/US/products/ps7282/prod_installation_guides_list.html
Kind Regards
Jamie Gale
Technical Marketing Engineer, Cisco Unified Attendant Consoles
Arc Solutions, onsite at Cisco
[email protected]
D +1 919 392 4671
M +1 919 699 4910
Find our new Cisco Unified Attendant Console End User training videos at
http://www.arcsolutions.com/north_america/solutions/products/cisco_oem_consoles.aspx
or https://www.youtube.com/channel/UC3jC1gmgsRWPR4PLR2VWBhA?feature=CCQQwRs%3D
Join the Cisco Unified Attendant Console Forum at Arc Solutions!
http://forum.arcsolutions.com/forumdisplay.php?f=4
On Oct 29, 2012, at 11:26 AM, Erick Wellnitz <[email protected]>
wrote:
Does anyone know what the minimum requirements are for a PC to run the Unified
Attendant Console? I'm having problems locating this information.
Thanks!
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip
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Message: 11
Date: Mon, 29 Oct 2012 16:35:16 +0100
From: Cristina Petre <[email protected]>
To: Ed Leatherman <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Unity 8.5.1
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"
Hi Ed,
Thanks for answering. Ive also created a pilot number but my pilot number is
assigned to forward all on the line to reach the unity. Do you mean it that
way?
Could you explain me the steps I need to do? Thanks.
Von meinem iPhone gesendet
Am 29.10.2012 um 16:20 schrieb Ed Leatherman <[email protected]>:
> The way I do this is setup a pilot number in CUCM and point to your unity
> connection line group, and then setup direct routing rule in unity connection
> to send the call to the call handler greeting that you want it to go to.
>
> On Mon, Oct 29, 2012 at 11:19 AM, Cristina Petre <[email protected]>
> wrote:
> Hi all,
>
> If i want to use call handlers in unity how is the connection between cucm
> and unity?
> The goal should be to dial a number from external to reach the cucm after
> that the unity to get a welcome message and after that to go back to cucm to
> a pilot point.
>
> I've created a directory number with voice mail pilot and voice mail profile,
> on unity I've created a system call handler with a transfer rule back to cucm
> and recorded a greeting. It's works but the problem is after I dial the
> number I get an "sorry.. Is not available" message after that phones in hunt
> lists ringing. I just want to hear my message nothing else.
>
> What's wrong on my configuration? Many thanks in advance.
>
> Von meinem iPhone gesendet
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> --
> Ed Leatherman
>
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Message: 12
Date: Mon, 29 Oct 2012 11:46:36 -0400
From: Ed Leatherman <[email protected]>
To: Cristina Petre <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Unity 8.5.1
Message-ID:
<cafc4dsoompnq3obqyod4xc1n_9wq7spmecjur7rp5mpiwcz...@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
I am meaning a Hunt Pilot in call routing, rather than the voice mail pilot
number:
In CUCM 7.1 (version i'm running):
Call Routing -> Route/Hunt -> Hunt Pilot
On Mon, Oct 29, 2012 at 11:35 AM, Cristina Petre
<[email protected]>wrote:
> Hi Ed,
>
> Thanks for answering. Ive also created a pilot number but my pilot number
> is assigned to forward all on the line to reach the unity. Do you mean it
> that way?
> Could you explain me the steps I need to do? Thanks.
>
> Von meinem iPhone gesendet
>
> Am 29.10.2012 um 16:20 schrieb Ed Leatherman <[email protected]>:
>
> The way I do this is setup a pilot number in CUCM and point to your unity
> connection line group, and then setup direct routing rule in unity
> connection to send the call to the call handler greeting that you want it
> to go to.
>
> On Mon, Oct 29, 2012 at 11:19 AM, Cristina Petre <
> [email protected]> wrote:
>
>> Hi all,
>>
>> If i want to use call handlers in unity how is the connection between
>> cucm and unity?
>> The goal should be to dial a number from external to reach the cucm after
>> that the unity to get a welcome message and after that to go back to cucm
>> to a pilot point.
>>
>> I've created a directory number with voice mail pilot and voice mail
>> profile, on unity I've created a system call handler with a transfer rule
>> back to cucm and recorded a greeting. It's works but the problem is after I
>> dial the number I get an "sorry.. Is not available" message after that
>> phones in hunt lists ringing. I just want to hear my message nothing else.
>>
>> What's wrong on my configuration? Many thanks in advance.
>>
>> Von meinem iPhone gesendet
>> _______________________________________________
>> cisco-voip mailing list
>> [email protected]
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
>
> --
> Ed Leatherman
>
>
--
Ed Leatherman
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Message: 13
Date: Mon, 29 Oct 2012 16:59:39 +0100
From: Cristina Petre <[email protected]>
To: Ed Leatherman <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] Unity 8.5.1
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"
My construct is as following:
DN with voice mail profile --> on DN forward all to voice mail --> on unity
call handler, transfer rule back to cucm to line group.
This construct is working but the message sorry "my recorded message" is not
available disturbs me.
Is my config ok?
Von meinem iPhone gesendet
Am 29.10.2012 um 16:46 schrieb Ed Leatherman <[email protected]>:
> I am meaning a Hunt Pilot in call routing, rather than the voice mail pilot
> number:
>
> In CUCM 7.1 (version i'm running):
> Call Routing -> Route/Hunt -> Hunt Pilot
>
>
> On Mon, Oct 29, 2012 at 11:35 AM, Cristina Petre <[email protected]>
> wrote:
> Hi Ed,
>
> Thanks for answering. Ive also created a pilot number but my pilot number is
> assigned to forward all on the line to reach the unity. Do you mean it that
> way?
> Could you explain me the steps I need to do? Thanks.
>
> Von meinem iPhone gesendet
>
> Am 29.10.2012 um 16:20 schrieb Ed Leatherman <[email protected]>:
>
>> The way I do this is setup a pilot number in CUCM and point to your unity
>> connection line group, and then setup direct routing rule in unity
>> connection to send the call to the call handler greeting that you want it to
>> go to.
>>
>> On Mon, Oct 29, 2012 at 11:19 AM, Cristina Petre <[email protected]>
>> wrote:
>> Hi all,
>>
>> If i want to use call handlers in unity how is the connection between cucm
>> and unity?
>> The goal should be to dial a number from external to reach the cucm after
>> that the unity to get a welcome message and after that to go back to cucm to
>> a pilot point.
>>
>> I've created a directory number with voice mail pilot and voice mail
>> profile, on unity I've created a system call handler with a transfer rule
>> back to cucm and recorded a greeting. It's works but the problem is after I
>> dial the number I get an "sorry.. Is not available" message after that
>> phones in hunt lists ringing. I just want to hear my message nothing else.
>>
>> What's wrong on my configuration? Many thanks in advance.
>>
>> Von meinem iPhone gesendet
>> _______________________________________________
>> cisco-voip mailing list
>> [email protected]
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>>
>> --
>> Ed Leatherman
>>
>
>
>
> --
> Ed Leatherman
>
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