[cisco-voip] cisco-voip Digest, Vol 58, Issue 49
Darren Smith (AU)
Darren.Smith2 at didata.com.au
Wed Aug 6 05:46:49 EDT 2008
Darren Smith
Solutions Architect - Connectivity
Dimension Data
Tel: +61 (0) 7 3292 1999
Mob: +61 (0) 417 491 055
Fax: +61 (0) 7 3292 1301
e-mail: darren.smith2 at didata.com.au
For more information about Dimension Data, please go to www.dimensiondata.com
----- Original Message -----
From: cisco-voip-bounces at puck.nether.net <cisco-voip-bounces at puck.nether.net>
To: cisco-voip at puck.nether.net <cisco-voip at puck.nether.net>
Sent: Wed Aug 06 18:22:24 2008
Subject: cisco-voip Digest, Vol 58, Issue 49
Send cisco-voip mailing list submissions to
cisco-voip at puck.nether.net
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
cisco-voip-request at puck.nether.net
You can reach the person managing the list at
cisco-voip-owner at puck.nether.net
When replying, please edit your Subject line so it is more specific
than "Re: Contents of cisco-voip digest..."
Today's Topics:
1. OT: I wonder if this utilizes in-band DTMF or not.
(Lelio Fulgenzi)
2. VTvantage issues (Tom Mc)
3. cucc 7 device limit on ccm (Kent Roberts)
4. Alerting Name being passed to PBX (William Roy)
5. Marking voicemail as private. (Yujin Tan)
6. Re: Alerting Name being passed to PBX (Tim Smith)
7. Re: how to configure which DID go out with mobility (Tim Smith)
8. Re: Alerting Name being passed to PBX (William Roy)
----------------------------------------------------------------------
Message: 1
Date: Tue, 5 Aug 2008 21:19:59 -0400
From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
Subject: [cisco-voip] OT: I wonder if this utilizes in-band DTMF or
not.
To: <cisco-voip at puck.nether.net>
Message-ID: <000e01c8f762$8b5c0b20$6501a8c0 at pdp11>
Content-Type: text/plain; charset="windows-1252"
http://www.youtube.com/watch?v=O_Ixg9G8DKM
--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"Bad grammar makes me [sic]" - Tshirt
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20080805/d18ba5ee/attachment-0001.html>
------------------------------
Message: 2
Date: Wed, 6 Aug 2008 12:35:53 +1000
From: "Tom Mc" <tomdmc at gmail.com>
Subject: [cisco-voip] VTvantage issues
To: cisco-voip at puck.nether.net
Message-ID:
<172436590808051935v5ee4b8c0y475574a2212752d6 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hello all,
We have a client which we have deployed call manager across about 20 sites
at this stage.
They want to use vtvantage for video conferencing.
Within the head office LAN, video works fine. Calls are made between video
enabled phones and vtvantage displays the video instantly.
Between the head office and a remote site, video is not displayed... Until
you click to Softphone in vtvantage and then back to hardphone.
Its odd, after then getting a video call working hardphone to hardphone, if
you hold a call or transfer you loose video. This part is normal I know, but
after resuming the call, you cannot get video back until you click to the
softphone and back again.
Starting the vtvantage software after the call has been established has the
same effect. Likewise if you hold the call, on resumption if you restart the
vtvantage software up pops video.
Like most problems, the ASA firewall at the head office was blamed. It was
thought to be an issue in the signaling.
But it has been tested between two phones at a remote site and it works
fine. (signaling would still be going through the head office ASA to the
call manager).
It has been tested from remote site to remote site also and the problem is
present.
We have tested setting 'location' bandwidth to unlimited and increasing the
'Region' video call bandwidth, but it does not seem to help.
We originally deployed vtvantage ver 2.1.1 and we have downgraded to 2.0.3
which has not helped.
Any ideas?
Cheers,
Tom
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20080806/1dcb07b0/attachment-0001.html>
------------------------------
Message: 3
Date: Tue, 05 Aug 2008 21:50:34 -0600
From: Kent Roberts <kent at fredf.org>
Subject: [cisco-voip] cucc 7 device limit on ccm
To: cisco-voip at puck.nether.net
Message-ID: <48991F8A.8030707 at fredf.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Question, I have hit the device limit on ccm of 2500 per user. I have
a new pg setup to talk to ccm as a new user. It registers and goes on
line. Does anyone know if I need to setup something so this new user's
devices will talk to the ctios and cad server, or if I have to setup a
new cad server/ctios for this new link?
------------------------------
Message: 4
Date: Wed, 6 Aug 2008 14:21:37 +0800
From: William Roy <William.Roy at l7.com.au>
Subject: [cisco-voip] Alerting Name being passed to PBX
To: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Message-ID:
<FF8A2D6EA63DD54A8D5FC59FD33D1E71101AEC0207 at auperaex01.l7.local>
Content-Type: text/plain; charset="iso-8859-1"
I have NEC IPX2400 PBX connected to a ISR2851 via an QSIG E1 link. The ISR is setup as an MGCP gateway with a CUCM 6.1 cluster. I am not seeing Calling name being passed either from an NEC phone to a Cisco IP phone or visa versa. Is there any debug I can run on the ISR to see if the cisco ip phone is sending Calling name?
regards
Wil
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20080806/cf6666af/attachment-0001.html>
------------------------------
Message: 5
Date: Wed, 6 Aug 2008 15:01:45 +0800
From: Yujin Tan <Yujin.Tan at l7.com.au>
Subject: [cisco-voip] Marking voicemail as private.
To: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Message-ID:
<FF8A2D6EA63DD54A8D5FC59FD33D1E71101AF8E81E at auperaex01.l7.local>
Content-Type: text/plain; charset="us-ascii"
Hi,
Is there an options on unity where you can mark a voice mail as private when you leave a message (just like when you mark priority)?? Seems you can only do that when you dial unity and choose send message.
Regards
YuJin
------------------------------
Message: 6
Date: Wed, 6 Aug 2008 09:54:56 +0200
From: "Tim Smith" <thsglobal at gmail.com>
Subject: Re: [cisco-voip] Alerting Name being passed to PBX
To: "William Roy" <William.Roy at l7.com.au>
Cc: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Message-ID:
<202613020808060054i2bb3f46difb3fede0aff1f976 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi Wil,
What do you see with a debug isdn q931?
Do you get connected name once the call is established?
Cheers,
Tim.
On 8/6/08, William Roy <William.Roy at l7.com.au> wrote:
>
> I have NEC IPX2400 PBX connected to a ISR2851 via an QSIG E1 link. The
> ISR is setup as an MGCP gateway with a CUCM 6.1 cluster. I am not seeing
> Calling name being passed either from an NEC phone to a Cisco IP phone or
> visa versa. Is there any debug I can run on the ISR to see if the cisco ip
> phone is sending Calling name?
>
> regards
> Wil
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20080806/509bea6f/attachment-0001.html>
------------------------------
Message: 7
Date: Wed, 6 Aug 2008 10:15:22 +0200
From: "Tim Smith" <thsglobal at gmail.com>
Subject: Re: [cisco-voip] how to configure which DID go out with
mobility
To: "James Grace" <grace.jd at gmail.com>
Cc: CiscosupportUpuck <cisco-voip at puck.nether.net>
Message-ID:
<202613020808060115q3a9bf3ffme861c0c69b8622d1 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi James,
Assuming it is going out an ISDN PRI?
First of all check the gateway - debug isdn q931 - See what calling number
you are sending
It's come up a few times before on this list I think.
But it really depends where you are.. and your telco.
Some telco's let you send any CLI you want, some only let you send numbers
from your DID range.
Some will let you add other numbers / ranges to an allowed list of CLI's.
If you send something invalid.. i.e. a 4 digit extension number without the
ext number mask - a lot of telco's will just overwrite it with the main
pilot number.
Best thing to do first is to check what you are sending.
Then you can work your way from the phone to the network.
i.e. have you got an external number mask set on the extension, are you
sending this on the route lists
Cheers,
Tim
On 8/6/08, James Grace <grace.jd at gmail.com> wrote:
>
> we are running cm 6.0 using mobility SNR and all call going out to PSTN is
> the main number. This is controlled by us. How do i configure mobility to
> send out my DID
>
> --
> James Grace
> CCVP CCNP CCNA MCSE MCDBA
> System Engineer / Consultant
> Email: grace.jd at gmail.com
>
> MSN IM: grace.jd at gmail.com
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20080806/18c856bd/attachment-0001.html>
------------------------------
Message: 8
Date: Wed, 6 Aug 2008 16:23:07 +0800
From: William Roy <William.Roy at l7.com.au>
Subject: Re: [cisco-voip] Alerting Name being passed to PBX
To: Tim Smith <thsglobal at gmail.com>
Cc: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Message-ID:
<FF8A2D6EA63DD54A8D5FC59FD33D1E71101AEC0212 at auperaex01.l7.local>
Content-Type: text/plain; charset="iso-8859-1"
No we do not get connected name once the call is connected only calling number. This is a debug from an IP phone to a NEC phone
ho-vgate-1#
ho-vgate-1#
*Aug 6 08:21:39.106: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x001E
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Calling Party Number i = 0x0081, '6005'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '4723'
Plan:Unknown, Type:Unknown
*Aug 6 08:21:39.166: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x801E
Channel ID i = 0xA9839F
Exclusive, Channel 31
*Aug 6 08:21:39.170: ISDN Se0/0/0:15 Q931: RX <- ALERTING pd = 8 callref = 0x801E
*Aug 6 08:21:43.158: ISDN Se0/0/0:15 Q931: RX <- CONNECT pd = 8 callref = 0x801E
Progress Ind i = 0x8082 - Destination address is non-ISDN
Connected Number i = 0x0081, '4723'
*Aug 6 08:21:43.162: ISDN Se0/0/0:15 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x001E
________________________________
From: smithsonianwa at gmail.com [smithsonianwa at gmail.com] On Behalf Of Tim Smith [thsglobal at gmail.com]
Sent: Wednesday, 6 August 2008 3:54 PM
To: William Roy
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Alerting Name being passed to PBX
Hi Wil,
What do you see with a debug isdn q931?
Do you get connected name once the call is established?
Cheers,
Tim.
On 8/6/08, William Roy <William.Roy at l7.com.au<mailto:William.Roy at l7.com.au>> wrote:
I have NEC IPX2400 PBX connected to a ISR2851 via an QSIG E1 link. The ISR is setup as an MGCP gateway with a CUCM 6.1 cluster. I am not seeing Calling name being passed either from an NEC phone to a Cisco IP phone or visa versa. Is there any debug I can run on the ISR to see if the cisco ip phone is sending Calling name?
regards
Wil
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>
https://puck.nether.net/mailman/listinfo/cisco-voip
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20080806/59fa8041/attachment.html>
------------------------------
_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip
End of cisco-voip Digest, Vol 58, Issue 49
******************************************
More information about the cisco-voip
mailing list