Otto,
Could hunt stop channel be used in combination with max-calls-per-button? In
other words - if I had a shared line at site C of 3500 - I want one phone with
this appearance to be able to accept 2 calls on the shared line - I want the
second phone to be able to accept 3 calls on the shared line - with an
additional stipulation that only 4 simultaneous calls to the shared line should
be possible - could I do it as follows:
ephone-dn 10
number 3500
huntstop channel 4
ephone 1
max-calls-per-button 2
button 1:10
ephone 2
max-calls-per-button 3
button 1:10
I thought this would work but when I have tried it does not.
Thanks
Scott
Date: Tue, 16 Mar 2010 09:02:34 -0430
From: o...@ipexpert.com
To: aar...@packet360.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] CME busy-trigger-per-button
Hi,
Have you tried to use huntstop channel 2 on the 1001 dn?, this will limit the
number of incoming calls the 1001 dn will receive, busy-trigger-per-button
command takes into account incoming and outgoing calls,
On Sat, Mar 13, 2010 at 12:43 AM, Aman Arora <aar...@packet360.com> wrote:
Hey Folks
Is it possible to set different busy-trigger-per-button for each button (line)
on a phone on CME.
For example :
If I have line 1 : 1000
And line 2 : 1001
I need to limit 4 incoming calls on line 1 and limit 2 incoming calls on line 2.
How can I achieve this. I guess busy-trigger-per-button sets limits for all the
buttons on the particular ephone.
Thanks
Aman
-----Original Message-----
From: ccie_voice-boun...@onlinestudylist.com
[mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of
ccie_voice-requ...@onlinestudylist.com
Sent: Friday, March 12, 2010 8:05 PM
To: ccie_voice@onlinestudylist.com
Subject: CCIE_Voice Digest, Vol 49, Issue 85
Send CCIE_Voice mailing list submissions to
ccie_voice@onlinestudylist.com
To subscribe or unsubscribe via the World Wide Web, visit
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When replying, please edit your Subject line so it is more specific
than "Re: Contents of CCIE_Voice digest..."
Today's Topics:
1. Re: UC and cme sip integration (Omotayo)
2. Re: SIP Hardware Transcoder (Jeff Cotter)
3. Re: SIP Hardware Transcoder (Omotayo)
----------------------------------------------------------------------
Message: 1
Date: Fri, 12 Mar 2010 23:01:11 +0100
From: Omotayo <adefilabi...@gmail.com>
Subject: Re: [OSL | CCIE_Voice] UC and cme sip integration
To: Flemming Ortvald <f...@netdesign.dk>
Cc: OSL Group <ccie_voice@onlinestudylist.com>
Message-ID:
<3082f9d41003121401o3d85ff29id1e503233e21d...@mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"
Hello,
it work ok now
I was using the wrong ip address on the unity connection all the while
Thanks
On Fri, Mar 12, 2010 at 8:30 AM, Flemming Ortvald <f...@netdesign.dk> wrote:
> Unity connection can do both g729 and g711, you can use ?voice class
> codec? on ?voice register dn? to expand codec support for sip.
>
>
>
> Med venlig hilsen
>
> Flemming Ortvald
> Network System Eng.
> NetDesign A/S
> +45 4435 8346
>
> T?nk p? milj?et inden udskrivning af denne e-post og tilknyttede
> vedh?ftninger
>
>
> *From:* Omotayo [mailto:adefilabi...@gmail.com]
> *Sent:* 11 March, 2010 20:58
> *To:* Flemming Ortvald
>
> *Subject:* Re: [OSL | CCIE_Voice] UC and cme sip integration
>
>
>
>
> Hello,
>
>
>
> I have to configure a transcoder on the br2 router?
>
>
>
> Unity connection support g729 only?
>
>
>
> Rgd
>
> On Thu, Mar 11, 2010 at 8:24 PM, Flemming Ortvald <f...@netdesign.dk> wrote:
>
> You will need a transcoder or chnage the sip endpoints to support g.711,
> natively it only supports g.729
>
>
>
> Best regards
>
> Flemming Ortvald
> Network System Eng.
> NetDesign A/S
> +45 4435 8346
>
> T?nk p? milj?et inden udskrivning af denne e-post og tilknyttede
> vedh?ftninger
>
>
> *From:* ccie_voice-boun...@onlinestudylist.com [mailto:
> ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Omotayo
> *Sent:* 11 March, 2010 20:07
> *To:* OSL Group
> *Subject:* Re: [OSL | CCIE_Voice] UC and cme sip integration
>
>
>
> Hello all,
>
>
>
> As anyone been able to get the SIP integration between Unity Connection and
> Cme to work? I followed the Proctorlabs Guide
>
>
>
> I posted this sometime lat week and revised as advised but keep getting a
> reorder tone( Number Unknown) when the message button is pressed
>
> Below is the relevant configuration
>
>
>
> voice service voip
> allow-connections h323 to h323
> allow-connections h323 to sip
> allow-connections sip to h323
> allow-connections sip to sip
> no supplementary-service sip moved-temporarily
> no supplementary-service sip refer
> sip
> bind control source-interface Loopback0
> bind media source-interface Loopback0
> registrar server expires max 600 min 60
>
>
>
>
>
>
>
> voice register global
> mode cme
> source-address 10.10.110.3 port 5060
> max-dn 3
> max-pool 6
> authenticate register
> mwi reg-e164
> voicemail 3600
> tftp-path flash:
> create profile sync 0006855418337003
> !
> voice register dn 1
> number 3002
> call-forward b2bua busy 3600
> call-forward b2bua mailbox 3002
> call-forward b2bua noan 3600 timeout 12
> name br2 phone 2
> no-reg
> label br2 phone 2
> mwi
> !
> voice register dn 2
> number 3003
> call-forward b2bua busy 3600
> call-forward b2bua mailbox 3003
> call-forward b2bua noan 3600 timeout 12
> name br2 phone 3
> no-reg
> label br2 phone 3
> mwi
> !
> voice register pool 1
> id mac 1111.1111.1111
> type 7941
> number 1 dn 1
> dtmf-relay rtp-nte
> username 3002 password cisco
> !
> voice register pool 2
> id mac 001F.6C7E.D6FE
> type 7941
> number 1 dn 2
> dtmf-relay rtp-nte
> username 3003 password cisco
>
>
>
>
>
> dial-peer voice 200 voip
> max-conn 1
> destination-pattern 3600
> session protocol sipv2
> session target ipv4:10.10.210.13
> dtmf-relay rtp-nte
> codec g711ulaw
> !
> !
>
>
>
> telephony-service
> no auto-reg-ephone
> em logout 0:0 0:0 0:0
> max-ephones 8
> max-dn 8
> ip source-address 10.10.202.1 port 2000
> voicemail 3600
> mwi relay
> max-conferences 8 gain -6
> transfer-system full-consult
> transfer-pattern .T
> create cnf-files version-stamp 7960 Mar 10 2010 15:22:39
> !
> !
> ephone-dn 1 dual-line
> number 3001 no-reg primary
> label Br2 pHone 1
> name Br2 Phone 1
> call-forward busy 3600
> call-forward noan 3600 timeout 12
> !
> !
>
> sip-ua
> mwi-server ipv4:10.10.210.13 expires 3600 port 5060 transport udp
> unsolicited
>
> !
> !
> ephone 1
> device-security-mode none
> mac-address 001E.EC15.996D
> type CIPC
> button 1:1
> !
>
>
>
> Thanks for the anticipated support
>
>
>
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Message: 2
Date: Fri, 12 Mar 2010 14:33:41 -0800
From: Jeff Cotter <jcot...@voxns.com>
Subject: Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
To: Omotayo <adefilabi...@gmail.com>, Otto Sanchez <o...@ipexpert.com>
Cc: "ccie_voice@onlinestudylist.com" <ccie_voice@onlinestudylist.com>
Message-ID: <54cc1bd3093b6e41b86926c1657432f187a06...@ssfex1>
Content-Type: text/plain; charset="us-ascii"
FYI, I was only able to get this to work using transcoder on CME. Had to match
the codec between UCM trunk and incoming dial-peer on CME...then xcoder would
engage on CME for the SIP phone. I have a hardware limitation in my home lab
so I am not able to configure a xcoder on both UCM and CME simultaneously.
From: Omotayo [mailto:adefilabi...@gmail.com]
Sent: Friday, March 12, 2010 6:33 AM
To: Otto Sanchez
Cc: Jeff Cotter; ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
Hello Otto,
i had same issue
The transcoder can be on the trunk?
When i did the transcoder on the br2 router, i get a busy tone when the sip
phone is being called from the hq phone
REgards
On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez
<o...@ipexpert.com<mailto:o...@ipexpert.com>> wrote:
Hi Jeff,
Would you please tell us more about the call flow and the end to end codec
requirements for this call. If doing g.729 over the wan, and your sip phone is
using g.711 you should transcode at br2,
Please let us know,
Thanks,
On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter
<jcot...@voxns.com<mailto:jcot...@voxns.com>> wrote:
Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM.
Can't seem to get a call from Call Manager to CME sip phone working. I can
call from CME to UCM but not the other way around. Rings but disconnects when
answered. Transcoder shows registered in Call manager. Thanks
Jeff
_______________________________________________
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com<http://www.ipexpert.com/>
--
Regards,
Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com<http://www.ipexpert.com/>
_______________________________________________
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com<http://www.ipexpert.com/>
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------------------------------
Message: 3
Date: Sat, 13 Mar 2010 02:04:50 +0100
From: Omotayo <adefilabi...@gmail.com>
Subject: Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
To: Jeff Cotter <jcot...@voxns.com>
Cc: "ccie_voice@onlinestudylist.com" <ccie_voice@onlinestudylist.com>
Message-ID:
<3082f9d41003121704j6e368e5egc989092a7343e...@mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"
Hello Jeff,
All calls worked when i configure the xcoder on the cme
The question says use the hq router resources- that is where i have issues
thanks
On Fri, Mar 12, 2010 at 11:33 PM, Jeff Cotter <jcot...@voxns.com> wrote:
> FYI, I was only able to get this to work using transcoder on CME. Had to
> match the codec between UCM trunk and incoming dial-peer on CME?then xcoder
> would engage on CME for the SIP phone. I have a hardware limitation in my
> home lab so I am not able to configure a xcoder on both UCM and CME
> simultaneously.
>
>
>
>
>
>
>
>
>
> *From:* Omotayo [mailto:adefilabi...@gmail.com]
> *Sent:* Friday, March 12, 2010 6:33 AM
> *To:* Otto Sanchez
> *Cc:* Jeff Cotter; ccie_voice@onlinestudylist.com
>
> *Subject:* Re: [OSL | CCIE_Voice] SIP Hardware Transcoder
>
>
>
> Hello Otto,
>
>
>
> i had same issue
>
>
>
> The transcoder can be on the trunk?
>
>
>
> When i did the transcoder on the br2 router, i get a busy tone when the sip
> phone is being called from the hq phone
>
>
>
> REgards
>
> On Fri, Mar 12, 2010 at 12:59 PM, Otto Sanchez <o...@ipexpert.com> wrote:
>
> Hi Jeff,
>
> Would you please tell us more about the call flow and the end to end codec
> requirements for this call. If doing g.729 over the wan, and your sip phone
> is using g.711 you should transcode at br2,
>
> Please let us know,
>
> Thanks,
>
> On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter <jcot...@voxns.com> wrote:
>
> Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder
> on UCM. Can?t seem to get a call from Call Manager to CME sip phone
> working. I can call from CME to UCM but not the other way around. Rings but
> disconnects when answered. Transcoder shows registered in Call manager.
> Thanks
>
>
>
>
>
> Jeff
>
>
>
> _______________________________________________
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
>
>
> --
> Regards,
>
> Otto Sanchez
> CCIE #25592 (Voice)
> Support Engineer - IPexpert, Inc.
> URL: http://www.IPexpert.com <http://www.ipexpert.com/>
>
>
> _______________________________________________
> For more information regarding industry leading CCIE Lab training, please
> visit www.ipexpert.com
>
>
>
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End of CCIE_Voice Digest, Vol 49, Issue 85
******************************************
_______________________________________________
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
--
Regards,
Otto Sanchez
CCIE #25592 (Voice)
Support Engineer - IPexpert, Inc.
URL: http://www.IPexpert.com
_________________________________________________________________
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