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Today's Topics:
1. Re: Gai-tronics RedAlert callbox analog with ATA-187 ? (Mike King)
2. dial peers and translations (Erick Wellnitz)
3. FXO Hunt Group, can't pickup 2nd call (Matthew Loraditch)
4. Re: dial peers and translations (Nick Matthews)
5. Re: FXO Hunt Group, can't pickup 2nd call (Jason Aarons (AM))
6. Re: dial peers and translations (Robert Kulagowski)
7. Re: FXO Hunt Group, can't pickup 2nd call (Norton, Mike)
8. Media Resources for Dummies ([email protected])
9. Re: dial peers and translations (Erick Wellnitz)
10. CME WITH AUTO ATTENDANT requirement (Vincent Dao)
11. Re: CME WITH AUTO ATTENDANT requirement (Gregory Wenzel)
12. Re: CME WITH AUTO ATTENDANT requirement (Vincent Dao)
13. Re: CME WITH AUTO ATTENDANT requirement (Gregory Wenzel)
14. Re: CME WITH AUTO ATTENDANT requirement (Vincent Dao)
----------------------------------------------------------------------
Message: 1
Date: Wed, 29 May 2013 12:26:51 -0400
From: Mike King <[email protected]>
To: "Jason Aarons (AM)" <[email protected]>
Cc: "cisco-voip \([email protected]\)"
<[email protected]>
Subject: Re: [cisco-voip] Gai-tronics RedAlert callbox analog with
ATA-187 ?
Message-ID:
<cantppk6p8rrq3-y3ixnhzru6u6cnoozsnb7nagzivb_qm0u...@mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"
Never used Gaitronics, Mostly just Ramtel phones. I would try the
polarity reversal, if you can enable it on the ATA187. This is what I had
to do for the Ramtel phones. (Note, I never used Ramtel phones on a cisco
phone system, this mostly dates from when I ran a Nortel Meridain CS1000
and had to use OPX cards to get the polarity reversal)
Mike
On Wed, May 29, 2013 at 8:14 AM, Jason Aarons (AM) <
[email protected]> wrote:
> FXS in general has a lack of Supervisory Disconnect, that said has anyone
> gotten a Gai-troncis RedAlert 393XX callbox to work with an ATA-187
> registered to CallManager?****
>
> ** **
>
> Problem is you press the callbox button, the public safety answers and
> tells you help is on the way. Public Safety hangs up and the callbox
> blairs out re-order tone for 3-4 minutes. (You may recall a home phone
> left offhook will play a loud re-order tone if you don?t hang it up due to
> lack of Supervisory Disconnect).****
>
> ** **
>
> I found the ATA-187 parameter Reorder Delay = Never turns off the sounds,
> but doesn?t really hang up the phone. The ATA-187 has polarity reversal,
> does this work with Gai-troncis callboxes?****
>
> ** **
>
> In short can anyone say they have had success getting the Gai-tronics to
> work with the ATA-187? Maybe there are some settings in the Gai-troncs
> device?****
>
> Page 36 doesn?t get me much to work with
> http://www.gai-tronics.com/manuals/42004-438i.pdf****
>
> ** **
>
> I?m thinking the VG-2XX is the better way to go as it has more Supervisory
> Disconnect options, but I can?t guarantee that will work without testing it.
> ****
>
> ** **
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 2
Date: Wed, 29 May 2013 11:43:18 -0500
From: Erick Wellnitz <[email protected]>
To: cisco-voip <[email protected]>
Subject: [cisco-voip] dial peers and translations
Message-ID:
<CAK0wOsCfj9fvOA6jiAWVb3S+R6EKDoYvaoLOWpjS0LhHP_S=y...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Simple question I just need a refresher.
On my CUBE, if I translate-outgoing called xxxx on the dial peer does my
destination pattern need to match the original number or the translated
number? The dial peer guides are not quite clear in my mind.
Here is my dial peer and my translation rule.
voice translation-rule 4000
rule 1 /^\+/ /9/
***replacing the + globalization with 9 for outbound PSTN access
dial-peer voice 4000 voip
description Inbound Toll Bypass from XXXXX
translate-outgoing called 4000
preference 1
destination-pattern +16304###### <-----should this be 916304######?
session protocol sipv2
session-taaget ipv4:172.16.7.12:5070
incoming called-number +16304######
dtmf-relay rtp-nte
codec g711ulaw
Thanks in advance!
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Message: 3
Date: Wed, 29 May 2013 17:43:30 +0000
From: Matthew Loraditch <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] FXO Hunt Group, can't pickup 2nd call
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="iso-8859-1"
Small site with 3 FXO lines in a carrier hunt. Gateway is MGCP controlled.
Lines are setup loop start. Calls come in and ring to a broadcast hunt. First
call comes in and rings and is answerable. Second call comes in, Caller hears
normal ringing. My users get ? ring try to pickup but can never get the call. ?
ring will continue indefinitely.
Anyone seen anything similar? I've not ever had the pleasure of this sort of
setup before so I'm not sure what to look at.
Thanks!
Matthew G. Loraditch - CCNP-Voice, CCNA-R&S, CCDA
1965 Greenspring Drive
Timonium, MD 21093
voice. 410.252.8830
fax. 410.252.9284
Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email
Support<mailto:[email protected]?subject=Technical%20Support%20Request>
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Message: 4
Date: Wed, 29 May 2013 13:58:28 -0400
From: Nick Matthews <[email protected]>
To: Erick Wellnitz <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] dial peers and translations
Message-ID:
<cam-k-nonaa6ukgutlefdq1dz6sur90bfvctbqh5yomldf12...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
For outbound no. Inbound it would be that case. If this outbound call
failed, it would try to re-route on the original non-translated number as
well.
-nick
On Wed, May 29, 2013 at 12:43 PM, Erick Wellnitz <[email protected]>wrote:
> Simple question I just need a refresher.
>
> On my CUBE, if I translate-outgoing called xxxx on the dial peer does my
> destination pattern need to match the original number or the translated
> number? The dial peer guides are not quite clear in my mind.
>
> Here is my dial peer and my translation rule.
>
> voice translation-rule 4000
> rule 1 /^\+/ /9/
> ***replacing the + globalization with 9 for outbound PSTN access
>
> dial-peer voice 4000 voip
> description Inbound Toll Bypass from XXXXX
> translate-outgoing called 4000
> preference 1
> destination-pattern +16304###### <-----should this be 916304######?
> session protocol sipv2
> session-taaget ipv4:172.16.7.12:5070
> incoming called-number +16304######
> dtmf-relay rtp-nte
> codec g711ulaw
>
> Thanks in advance!
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 5
Date: Wed, 29 May 2013 13:58:32 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: Matthew Loraditch <[email protected]>,
"[email protected]" <[email protected]>
Subject: Re: [cisco-voip] FXO Hunt Group, can't pickup 2nd call
Message-ID:
<4e38db0a1959b04c8c83edcf069b53ed0d3551a...@usispclexdb01.na.didata.local>
Content-Type: text/plain; charset="windows-1252"
Debug ccm-manager backhaul packets
Debug mgcp packet
When you answer on the IP phone do you see the MDCX come in from callmanager to
the gateway, etc ?
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080174804.shtml
I suspect your need both a capture from back of phone, ccm trace logs
(translator), and the debugs from gateway.
Did you issue a mgcp bind xxxxxx command twice? If you look in CCM Admin does
the ip address of the MGCP endpoint match your bindings? If seen this when
they mis-matched.
From: cisco-voip [mailto:[email protected]] On Behalf Of
Matthew Loraditch
Sent: Wednesday, May 29, 2013 1:44 PM
To: [email protected]
Subject: [cisco-voip] FXO Hunt Group, can't pickup 2nd call
Small site with 3 FXO lines in a carrier hunt. Gateway is MGCP controlled.
Lines are setup loop start. Calls come in and ring to a broadcast hunt. First
call comes in and rings and is answerable. Second call comes in, Caller hears
normal ringing. My users get ? ring try to pickup but can never get the call. ?
ring will continue indefinitely.
Anyone seen anything similar? I've not ever had the pleasure of this sort of
setup before so I'm not sure what to look at.
Thanks!
Matthew G. Loraditch - CCNP-Voice, CCNA-R&S, CCDA
1965 Greenspring Drive
Timonium, MD 21093
voice. 410.252.8830
fax. 410.252.9284
Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email
Support<mailto:[email protected]?subject=Technical%20Support%20Request>
itevomcid
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Message: 6
Date: Wed, 29 May 2013 12:53:24 -0500
From: Robert Kulagowski <[email protected]>
To: Erick Wellnitz <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] dial peers and translations
Message-ID:
<cage0w3tv2hmqnpndt66db+r9e43x17a+csxj9amvgwnaten...@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
On Wed, May 29, 2013 at 11:43 AM, Erick Wellnitz
<[email protected]> wrote:
> Simple question I just need a refresher.
>
> On my CUBE, if I translate-outgoing called xxxx on the dial peer does my
> destination pattern need to match the original number or the translated
> number? The dial peer guides are not quite clear in my mind.
>
> Here is my dial peer and my translation rule.
>
> voice translation-rule 4000
> rule 1 /^\+/ /9/
> ***replacing the + globalization with 9 for outbound PSTN access
>
> dial-peer voice 4000 voip
> description Inbound Toll Bypass from XXXXX
> translate-outgoing called 4000
> preference 1
> destination-pattern +16304###### <-----should this be 916304######?
> session protocol sipv2
> session-taaget ipv4:172.16.7.12:5070
> incoming called-number +16304######
> dtmf-relay rtp-nte
> codec g711ulaw
The translation happens once you've matched a dial-peer: (we dial "8"
to get out so that people don't accidentally dial "911" by accident)
dial-peer voice 10000 voip
description Outbound SIP to Bandwidth.com
translation-profile outgoing SIP-OUT
preference 1
destination-pattern 8T
session protocol sipv2
session target dns:ot.bandwidth.com
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 200
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
voice translation-profile SIP-OUT
translate called 8
voice translation-rule 8
rule 1 /^81\(.*\)/ /+1\1/
rule 2 /^\*81\(.*\)/ /+1\1/
rule 3 /^8011\(.*\)/ /+\1/
rule 4 /^\*8011\(.*\)/ /+\1/
rule 10 /^91\(.*\)/ /+1\1/
rule 11 /^9011\(.*\)/ /+\1/
------------------------------
Message: 7
Date: Wed, 29 May 2013 18:32:23 +0000
From: "Norton, Mike" <[email protected]>
To: Matthew Loraditch <[email protected]>,
"[email protected]" <[email protected]>
Subject: Re: [cisco-voip] FXO Hunt Group, can't pickup 2nd call
Message-ID:
<2c4107b204f8e74eb0a63ef59b0f970212bbd...@pwsdexchange06.pwsb33.ab.ca>
Content-Type: text/plain; charset="iso-8859-1"
I would swap the FXO connections around and check if the problem stays with the
line or stays with the FXO port.
If the problem stays with the specific FXO port, then it's probably something
with your config. Confirm trunk type is correct. I've had half-ring
unanswerable calls when caller-id was enabled on the port but the line didn't
actually have caller ID service. I've also had that sort of behaviour with
fried FXO cards. Can also happen if CUCM DB replication is wonky, causing
gateway registration to be messed up.
If the problem moves when you move the line to a different port, then it's
probably something with the line. Bad connection that craps out upon loop
current flowing is possible. Or there is something else sharing the line that
you don't know about. Building security system, HVAC controller, utility meter
etc. sometimes get spliced onto a line by people who don't necessarily
understand the implications of doing so. These devices often attempt to have
"intelligent" line-sharing capabilities, which tends to get confused by IOS
gateway taking the line off-hook the instant it senses ringing.
-mn
From: cisco-voip [mailto:[email protected]] On Behalf Of
Matthew Loraditch
Sent: May-29-13 11:44 AM
To: [email protected]
Subject: [cisco-voip] FXO Hunt Group, can't pickup 2nd call
Small site with 3 FXO lines in a carrier hunt. Gateway is MGCP controlled.
Lines are setup loop start. Calls come in and ring to a broadcast hunt. First
call comes in and rings and is answerable. Second call comes in, Caller hears
normal ringing. My users get ? ring try to pickup but can never get the call. ?
ring will continue indefinitely.
Anyone seen anything similar? I've not ever had the pleasure of this sort of
setup before so I'm not sure what to look at.
Thanks!
Matthew G. Loraditch - CCNP-Voice, CCNA-R&S, CCDA
1965 Greenspring Drive
Timonium, MD 21093
voice. 410.252.8830
fax. 410.252.9284
Twitter<http://twitter.com/heliontech> |
Facebook<http://www.facebook.com/#!/pages/Helion/252157915296> |
Website<http://www.heliontechnologies.com/> | Email
Support<mailto:[email protected]?subject=Technical%20Support%20Request>
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Message: 8
Date: Wed, 29 May 2013 13:35:47 -0500
From: [email protected]
To: [email protected]
Subject: [cisco-voip] Media Resources for Dummies
Message-ID:
<ofd7cea52d.6f01d0a2-on86257b7a.0063008b-86257b7a.00662...@assurant.com>
Content-Type: text/plain; charset="us-ascii"
I've been doing this awhile now and while I've learned a ton, I still
can't quite grasp some of the Media Resource usage and why. We recently
put in a SIP proxy and are now routing a dozen+ call centers through it
using 1 ICM instance and 3 Call Manager clusters. I'm very excited to
turn on significant digits but I want to iron out all of these new issues
that popped up. With the SIP proxy, we now have separate trunks to Call
Manager based on location. This is what we wanted to help iron out some
other issues with MOH and internal transfers. My plan was to throw
resources at these trunks and let them get used when needed. This didn't
work out at all like I planned. Calls started using these resources
because they could not because they needed too. After cleaning it all up,
I've still got a few intermittent issues that I'm working with TAC on.
One-way audio and conference calls get dropped when the original agent
leaves the call. These are fun and we'll eventually figure out what is
causing them. What I ask of you is this. Is there a Media Resource for
Dummies document out there that describes what resources get used and
where?
For example:
The SIP trunk to the SIP Proxy in Call Manager. This trunk is used for
two things. To deliver a call to an agent extension from ICM and to
deliver the label from ICM via a CTI Route point back thru CVP to ICM. In
our case, I know that I need a Unicast MOH resource on this trunk for when
an agent puts a customer on hold. (Unicast because the ingress gateways
are 5400's)
Internal transfers back into ICM. I believe the CTI Route Point device
pool/MRGL will need software MTP's built using a DSPFARM on a router local
to the ingress gateway to allow the internal transfer to work correctly.
My biggest hurdle is that these locations all do something a little
differently when it comes to call routing. Add Cisco Mobile Agent to this
mess and the need for MTP is required checked on some gateways is deadly.
The smallest of changes always seem to test well with a handful of users
after hours but when the real work force comes in, there are intermittent
issues that always pop up. A majority of the locations receive incoming
call center calls at one of our main locations and since the agents are
across the WAN the call needs to be G.729. So it's not only Media
Resources but Region settings
that crush the very fabric of my inner soul. That is a stretch but I'm
hoping your seeing where I'm coming from. It has to be easier than this.
Looking for that Dummies book and fast!
Steve Siltman
[email protected]
This e-mail message and all attachments transmitted with it may
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Message: 9
Date: Wed, 29 May 2013 13:55:53 -0500
From: Erick Wellnitz <[email protected]>
To: Robert Kulagowski <[email protected]>, Nick Matthews
<[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] dial peers and translations
Message-ID:
<CAK0wOsCa074Xv2VzWH4fuk=ovtzzlrhv56d1cebd_qpve_v...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Thanks guys! Makes perfect sense now.
I'm in the process of cleaning up a CUBE so that the dial peer logic is
easier to follow. It isn't complex but it is production so I have to be
sure of my footsteps.
On Wed, May 29, 2013 at 12:53 PM, Robert Kulagowski <[email protected]>wrote:
> On Wed, May 29, 2013 at 11:43 AM, Erick Wellnitz
> <[email protected]> wrote:
> > Simple question I just need a refresher.
> >
> > On my CUBE, if I translate-outgoing called xxxx on the dial peer does my
> > destination pattern need to match the original number or the translated
> > number? The dial peer guides are not quite clear in my mind.
> >
> > Here is my dial peer and my translation rule.
> >
> > voice translation-rule 4000
> > rule 1 /^\+/ /9/
> > ***replacing the + globalization with 9 for outbound PSTN access
> >
> > dial-peer voice 4000 voip
> > description Inbound Toll Bypass from XXXXX
> > translate-outgoing called 4000
> > preference 1
> > destination-pattern +16304###### <-----should this be 916304######?
> > session protocol sipv2
> > session-taaget ipv4:172.16.7.12:5070
> > incoming called-number +16304######
> > dtmf-relay rtp-nte
> > codec g711ulaw
>
> The translation happens once you've matched a dial-peer: (we dial "8"
> to get out so that people don't accidentally dial "911" by accident)
>
> dial-peer voice 10000 voip
> description Outbound SIP to Bandwidth.com
> translation-profile outgoing SIP-OUT
> preference 1
> destination-pattern 8T
> session protocol sipv2
> session target dns:ot.bandwidth.com
> voice-class sip dtmf-relay force rtp-nte
> voice-class sip profiles 200
> dtmf-relay rtp-nte
> codec g711ulaw
> ip qos dscp cs5 media
> ip qos dscp cs4 signaling
> no vad
>
> voice translation-profile SIP-OUT
> translate called 8
>
> voice translation-rule 8
> rule 1 /^81\(.*\)/ /+1\1/
> rule 2 /^\*81\(.*\)/ /+1\1/
> rule 3 /^8011\(.*\)/ /+\1/
> rule 4 /^\*8011\(.*\)/ /+\1/
> rule 10 /^91\(.*\)/ /+1\1/
> rule 11 /^9011\(.*\)/ /+\1/
>
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Message: 10
Date: Thu, 30 May 2013 08:17:37 -0700
From: "Vincent Dao" <[email protected]>
To: <[email protected]>
Subject: [cisco-voip] CME WITH AUTO ATTENDANT requirement
Message-ID: <EA5787B307FC4D2D90E8C87B9FD3683B@xw8200>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original
Hello All,
would someone please give me a quick brief what is my requirement to make
Auto Attendant work when call in from external source.
i have the voicemail working but not the AA, not sure what am i missing.
Thank you very much for your time and very appreciate.
.........People First..........
Best Regards,
Vincent Dao
------------------------------
Message: 11
Date: Thu, 30 May 2013 15:26:49 +0000
From: Gregory Wenzel <[email protected]>
To: Vincent Dao <[email protected]>, "[email protected]"
<[email protected]>
Subject: Re: [cisco-voip] CME WITH AUTO ATTENDANT requirement
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Do you have the dial peer that points the inbound AA number to CUE?
Can you dial the AA number from the phones?
-----Original Message-----
From: cisco-voip [mailto:[email protected]] On Behalf Of
Vincent Dao
Sent: Thursday, May 30, 2013 11:18 AM
To: [email protected]
Subject: [cisco-voip] CME WITH AUTO ATTENDANT requirement
Hello All,
would someone please give me a quick brief what is my requirement to make Auto
Attendant work when call in from external source.
i have the voicemail working but not the AA, not sure what am i missing.
Thank you very much for your time and very appreciate.
.........People First..........
Best Regards,
Vincent Dao
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip
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proprietary. If you are not an intended recipient, please notify the sender,
and then please delete and destroy all copies and attachments, and be advised
that any review or dissemination of, or the taking of any action in reliance
on, the information contained in or attached to this message is prohibited.
Unless specifically indicated, this message is not an offer to sell or a
solicitation of any products.
------------------------------
Message: 12
Date: Thu, 30 May 2013 08:34:41 -0700
From: "Vincent Dao" <[email protected]>
To: "Gregory Wenzel" <[email protected]>,
<[email protected]>
Subject: Re: [cisco-voip] CME WITH AUTO ATTENDANT requirement
Message-ID: <5A49D98762D44B269570CFED1BAD4ECA@xw8200>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original
HI Gregory,
i have dial peer to voice mail, which is 50., for 500, and my AA is 502. do
i need another dial peer for it AA? yes, when i dial 502 on the phone, my
AA is working OK
.........People First..........
Best Regards,
Vincent Dao
----- Original Message -----
From: "Gregory Wenzel" <[email protected]>
To: "Vincent Dao" <[email protected]>; <[email protected]>
Sent: Thursday, May 30, 2013 8:26 AM
Subject: RE: [cisco-voip] CME WITH AUTO ATTENDANT requirement
Do you have the dial peer that points the inbound AA number to CUE?
Can you dial the AA number from the phones?
-----Original Message-----
From: cisco-voip [mailto:[email protected]] On Behalf Of
Vincent Dao
Sent: Thursday, May 30, 2013 11:18 AM
To: [email protected]
Subject: [cisco-voip] CME WITH AUTO ATTENDANT requirement
Hello All,
would someone please give me a quick brief what is my requirement to make
Auto Attendant work when call in from external source.
i have the voicemail working but not the AA, not sure what am i missing.
Thank you very much for your time and very appreciate.
.........People First..........
Best Regards,
Vincent Dao
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip
This message w/attachments (message) is solely for the use of the intended
recipient(s) and may contain information that is privileged, confidential or
proprietary. If you are not an intended recipient, please notify the sender,
and then please delete and destroy all copies and attachments, and be
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Unless specifically indicated, this message is not an offer to sell or a
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------------------------------
Message: 13
Date: Thu, 30 May 2013 15:36:53 +0000
From: Gregory Wenzel <[email protected]>
To: Vincent Dao <[email protected]>, "[email protected]"
<[email protected]>
Subject: Re: [cisco-voip] CME WITH AUTO ATTENDANT requirement
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Yes another dial peer for Ext 502 should point to the same as the vm sip peer.
HTH
-----Original Message-----
From: Vincent Dao [mailto:[email protected]]
Sent: Thursday, May 30, 2013 11:35 AM
To: Gregory Wenzel; [email protected]
Subject: Re: [cisco-voip] CME WITH AUTO ATTENDANT requirement
HI Gregory,
i have dial peer to voice mail, which is 50., for 500, and my AA is 502. do i
need another dial peer for it AA? yes, when i dial 502 on the phone, my AA is
working OK
.........People First..........
Best Regards,
Vincent Dao
----- Original Message -----
From: "Gregory Wenzel" <[email protected]>
To: "Vincent Dao" <[email protected]>; <[email protected]>
Sent: Thursday, May 30, 2013 8:26 AM
Subject: RE: [cisco-voip] CME WITH AUTO ATTENDANT requirement
Do you have the dial peer that points the inbound AA number to CUE?
Can you dial the AA number from the phones?
-----Original Message-----
From: cisco-voip [mailto:[email protected]] On Behalf Of
Vincent Dao
Sent: Thursday, May 30, 2013 11:18 AM
To: [email protected]
Subject: [cisco-voip] CME WITH AUTO ATTENDANT requirement
Hello All,
would someone please give me a quick brief what is my requirement to make
Auto Attendant work when call in from external source.
i have the voicemail working but not the AA, not sure what am i missing.
Thank you very much for your time and very appreciate.
.........People First..........
Best Regards,
Vincent Dao
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Message: 14
Date: Thu, 30 May 2013 08:42:00 -0700
From: "Vincent Dao" <[email protected]>
To: "Gregory Wenzel" <[email protected]>,
<[email protected]>
Subject: Re: [cisco-voip] CME WITH AUTO ATTENDANT requirement
Message-ID: <3FE6370CC96A47BB8DC12FF8F1CFFB2E@xw8200>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=original
HI Gregory,
so i already have a dial peer below, with 5..
dial-peer voice 160 voip
description Voicemail
destination-pattern 5..
session protocol sipv2
session target ipv4:10.99.x.x
dtmf-relay sip-notify
codec g711ulaw
no vad
I still need another dial peer, let say, 170 voip and destination to 5.., or
502 again? please clarify and thank you very much for your time.
do i need to use TCL scrip at all?
.........People First..........
Best Regards,
Vincent Dao
----- Original Message -----
From: "Gregory Wenzel" <[email protected]>
To: "Vincent Dao" <[email protected]>; <[email protected]>
Sent: Thursday, May 30, 2013 8:36 AM
Subject: RE: [cisco-voip] CME WITH AUTO ATTENDANT requirement
Yes another dial peer for Ext 502 should point to the same as the vm sip
peer.
HTH
-----Original Message-----
From: Vincent Dao [mailto:[email protected]]
Sent: Thursday, May 30, 2013 11:35 AM
To: Gregory Wenzel; [email protected]
Subject: Re: [cisco-voip] CME WITH AUTO ATTENDANT requirement
HI Gregory,
i have dial peer to voice mail, which is 50., for 500, and my AA is 502. do
i need another dial peer for it AA? yes, when i dial 502 on the phone, my
AA is working OK
.........People First..........
Best Regards,
Vincent Dao
----- Original Message -----
From: "Gregory Wenzel" <[email protected]>
To: "Vincent Dao" <[email protected]>; <[email protected]>
Sent: Thursday, May 30, 2013 8:26 AM
Subject: RE: [cisco-voip] CME WITH AUTO ATTENDANT requirement
Do you have the dial peer that points the inbound AA number to CUE?
Can you dial the AA number from the phones?
-----Original Message-----
From: cisco-voip [mailto:[email protected]] On Behalf Of
Vincent Dao
Sent: Thursday, May 30, 2013 11:18 AM
To: [email protected]
Subject: [cisco-voip] CME WITH AUTO ATTENDANT requirement
Hello All,
would someone please give me a quick brief what is my requirement to make
Auto Attendant work when call in from external source.
i have the voicemail working but not the AA, not sure what am i missing.
Thank you very much for your time and very appreciate.
.........People First..........
Best Regards,
Vincent Dao
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip
This message w/attachments (message) is solely for the use of the intended
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proprietary. If you are not an intended recipient, please notify the sender,
and then please delete and destroy all copies and attachments, and be
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reliance on, the information contained in or attached to this message is
prohibited.
Unless specifically indicated, this message is not an offer to sell or a
solicitation of any products.
This message w/attachments (message) is solely for the use of the intended
recipient(s) and may contain information that is privileged, confidential or
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Unless specifically indicated, this message is not an offer to sell or a
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------------------------------
Subject: Digest Footer
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------------------------------
End of cisco-voip Digest, Vol 115, Issue 29
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