[cisco-voip] help
Ken Vizena
Kvizena at ursi.com
Tue Feb 8 14:39:12 EST 2005
Please remove
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
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cisco-voip-request at puck.nether.net
Sent: Tuesday, February 08, 2005 1:36 PM
To: cisco-voip at puck.nether.net
Subject: cisco-voip Digest, Vol 24, Issue 24
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Today's Topics:
1. Re: Problems with CallerID on a PRI connected to PA-VXC-2TE1
(John Lange)
2. Recording Phones (Eric McKinney)
3. Re: Problems with CallerID on a PRI connected to PA-VXC-2TE1
(peter.casanave at us.army.mil)
4. RE: Recording Phones (Voll, Scott)
5. Unity VM Light (Voll, Scott)
6. RE: Unity VM Light (Walenta, Phil)
----------------------------------------------------------------------
Message: 1
Date: Tue, 08 Feb 2005 12:27:09 -0600
From: John Lange <john.lange at open-it.ca>
Subject: Re: [cisco-voip] Problems with CallerID on a PRI connected to
PA-VXC-2TE1
To: Dave Temkin <dave at ordinaryworld.com>
Cc: cisco VoIP <cisco-voip at puck.nether.net>
Message-ID: <1107887229.9204.153.camel at ws103.darkcore.net>
Content-Type: text/plain; charset=iso-8859-1
Thanks Dave.
Unfortunately we tried that setting but it had no effect.
Its very confusing to me how the Name information could be passing
properly but not the number info.
If you have any other suggestions I'd be glad to hear them.
Thanks.
John
On Mon, 2005-02-07 at 12:22, Dave Temkin wrote:
> Try:
>
> isdn outgoing display-ie
> on your serial interface for the PRI
>
>
>
> -Dave
>
> On Mon, 7 Feb 2005, John Lange wrote:
>
> > We have an Asterisk Phone system that places outbound calls using
SIP
> > via a Cisco PA-VXC-2TE1+ card connected to an Allstream PRI.
> >
> > Try as we might we can not get callerid number working on outbound
> > calls.
> >
> > Strangely, CallerID name display works perfectly fine.
> >
> > Allstream insists that their equipment is configured correctly and
that
> > they will accept the name/number we set on calls.
> >
> > So the question is, what is the setting on the Cisco so that it
passes
> > CID name and number?
> >
> > The following is the trace provided by Allstream on a sample
outbound
> > call from our equipment. As you can see DSP is set properly to the
name
> > but there is no CGN portion. (From my conversations with them my
> > understanding is the CGN is the number, DSP is the name?)
> >
> > Below this trace is the current configuration for the Cisco.
> >
> > Any suggestions are welcome.
> >
> > Thanks.
> >
> > --------------
> >
> > <== 03:12:51:05.11 (CM Time: 02:15:56:55.12).
> > <== Q931: SETUP: from S[7040] L[1,186,0] E[26,185,0] SPA[----]
> > CR: 0,01 6C
> > BC: speech
> > 64 kbit/s
> > circuit mode
> > mu-law speech
> > CID: 18
> > LENGTH: 03
> > Channel Selection Info: As Indicated in Following Octets
> > D-Channel Indicator: D-Channel NOT indicated
> > Preferred/Exclusive: Exclusive
> > Interface type = primary rate
> > Interface Identifier: IID Implicitly Identified
> > Channel Type: B - Channel Units (3).
> > Number Map: Channel is indicated by the number following.
> > Coding Standard: CCITT
> > Channel Number = 23
> > DSP: ?OpenIT
> > CDN: unknown
> > unknown
> > 14162045956
> > --------------
> >
> >
> > Current configuration : 3012 bytes
> > !
> > ! Last configuration change at 14:10:36 CST Thu Feb 3 2005
> > ! NVRAM config last updated at 13:54:27 CST Tue Feb 1 2005
> > !
> > version 12.2
> > service timestamps debug datetime msec
> > service timestamps log datetime msec
> > service password-encryption
> > !
> > hostname WPG-SIPgw-01
> > !
> > boot system slot0:c7200-is-mz.122-15.T14.bin
> > card type t1 1
> > logging queue-limit 100
> > enable secret 5 ***REMOVED***
> > !
> > clock timezone CST -6
> > clock summer-time CDT recurring 1 Sun Apr 2:00 4 Sun Oct 2:00
> > dspint DSPfarm1/0
> > !
> > ip subnet-zero
> > !
> > !
> > ip cef
> > !
> > isdn switch-type primary-dms100
> > !
> > !
> > !
> > voice service voip
> > sip
> > !
> > !
> > !
> > !
> > !
> > !
> > !
> > no voice hpi capture buffer
> > no voice hpi capture destination
> > !
> > !
> > mta receive maximum-recipients 0
> > !
> > !
> > controller T1 1/0
> > framing esf
> > linecode b8zs
> > cablelength long 0db
> > pri-group timeslots 1-24
> > !
> > controller T1 1/1
> > shutdown
> > framing esf
> > linecode b8zs
> > cablelength long 0db
> > !
> > !
> > !
> > interface FastEthernet0/0
> > ip address XXX.XXX.106.150 255.255.255.0
> > duplex full
> > !
> > interface Serial1/0:23
> > no ip address
> > no logging event link-status
> > isdn switch-type primary-dms100
> > isdn incoming-voice voice
> > no cdp enable
> > !
> > ip classless
> > ip route 0.0.0.0 0.0.0.0 XXX.XXX.106.1
> > no ip http server
> > !
> > !
> > !
> > access-list 10 permit XXX.XXX.106.4
> > access-list 10 permit XXX.XXX.106.5
> > !
> > !
> > call rsvp-sync
> > !
> > voice-port 1/0:23
> > !
> > !
> > mgcp profile default
> > !
> > dial-peer cor custom
> > !
> > !
> > !
> > dial-peer voice 100 voip
> > application session
> > destination-pattern XXXXX54235
> > progress_ind setup enable 3
> > session protocol sipv2
> > session target ipv4:XXX.XXX.99.50
> > session transport udp
> > !
> > dial-peer voice 101 voip
> > application session
> > destination-pattern XXXXX54236
> > progress_ind setup enable 3
> > session protocol sipv2
> > session target ipv4:XXX.XXX.102.59
> > session transport udp
> > !
> > dial-peer voice 5000 pots
> > application session
> > destination-pattern .......
> > direct-inward-dial
> > port 1/0:23
> > !
> > dial-peer voice 102 voip
> > application session
> > destination-pattern XXXXX74002
> > progress_ind setup enable 3
> > session protocol sipv2
> > session target ipv4:XXX.XXX.102.59
> > session transport udp
> > !
> > dial-peer voice 103 voip
> > application session
> > destination-pattern XXXXX74003
> > progress_ind setup enable 3
> > session protocol sipv2
> > session target ipv4:XXX.XXX.102.59
> > session transport udp
> > codec g711ulaw
> > !
> > dial-peer voice 104 voip
> > application session
> > destination-pattern XXXXX74004
> > progress_ind setup enable 3
> > session protocol sipv2
> > session target ipv4:XXX.XXX.102.59
> > session transport udp
> > !
> > dial-peer voice 105 voip
> > destination-pattern XXXXX74005
> > progress_ind setup enable 3
> > session protocol sipv2
> > session target ipv4:XXX.XXX.102.59
> > session transport udp
> > !
> > dial-peer voice 5001 pots
> > incoming called-number 1..........
> > direct-inward-dial
> > port 1/0:23
> > prefix 1
> > !
> > sip-ua
> > calling-info sip-to-pstn number set XXXXX74006
> > !
> > !
> > gatekeeper
> > shutdown
> > !
> > !
> > line con 0
> > stopbits 1
> > line aux 0
> > stopbits 1
> > line vty 0 4
> > access-class 10 in
> > exec-timeout 60 0
> > password ***REMOVED***
> > login
> > !
> > ntp clock-period 17180028
> > ntp server XXX.XXX.111.252
> > !
> > end
> >
> >
------------------------------
Message: 2
Date: Tue, 8 Feb 2005 14:00:32 -0500
From: "Eric McKinney" <emckinney at fuquay-varina.org>
Subject: [cisco-voip] Recording Phones
To: <cisco-voip at puck.nether.net>
Message-ID: <200502081856.j18Iu4Ol037870 at puck.nether.net>
Content-Type: text/plain; charset="us-ascii"
I have to record some of our phones and I have Mercom Audiolog for this.
I
was planning on using RSPAN to record all the voice traffic. Has
anybody
done this before? Any pitfalls for me to be aware of?
Thanks,
Eric McKinney
-------------- next part --------------
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------------------------------
Message: 3
Date: Tue, 08 Feb 2005 13:02:07 -0600
From: <peter.casanave at us.army.mil>
Subject: Re: [cisco-voip] Problems with CallerID on a PRI connected to
PA-VXC-2TE1
To: John Lange <john.lange at open-it.ca>
Cc: cisco VoIP <cisco-voip at puck.nether.net>
Message-ID: <cab4b02caaf24a.caaf24acab4b02 at us.army.mil>
Content-Type: text/plain; charset=iso-8859-1
This link may be of assistance:
http://www.cisco.com/en/US/customer/tech/tk652/tk653/technologies_config
uration_example09186a00800a9a49.shtml
V/R,
PETER CASANAVE
SFC, USA
BBN Platoon Sergeant
----- Original Message -----
From: John Lange <john.lange at open-it.ca>
Date: Tuesday, February 8, 2005 12:27 pm
Subject: Re: [cisco-voip] Problems with CallerID on a PRI connected to
PA-VXC-2TE1
> Thanks Dave.
>
> Unfortunately we tried that setting but it had no effect.
>
> Its very confusing to me how the Name information could be passing
> properly but not the number info.
>
> If you have any other suggestions I'd be glad to hear them.
>
> Thanks.
>
> John
>
>
> On Mon, 2005-02-07 at 12:22, Dave Temkin wrote:
> > Try:
> >
> > isdn outgoing display-ie
> > on your serial interface for the PRI
> >
> >
> >
> > -Dave
> >
> > On Mon, 7 Feb 2005, John Lange wrote:
> >
> > > We have an Asterisk Phone system that places outbound calls
> using SIP
> > > via a Cisco PA-VXC-2TE1+ card connected to an Allstream PRI.
> > >
> > > Try as we might we can not get callerid number working on outbound
> > > calls.
> > >
> > > Strangely, CallerID name display works perfectly fine.
> > >
> > > Allstream insists that their equipment is configured correctly
> and that
> > > they will accept the name/number we set on calls.
> > >
> > > So the question is, what is the setting on the Cisco so that
> it passes
> > > CID name and number?
> > >
> > > The following is the trace provided by Allstream on a sample
> outbound> > call from our equipment. As you can see DSP is set
> properly to the name
> > > but there is no CGN portion. (From my conversations with them my
> > > understanding is the CGN is the number, DSP is the name?)
> > >
> > > Below this trace is the current configuration for the Cisco.
> > >
> > > Any suggestions are welcome.
> > >
> > > Thanks.
> > >
> > > --------------
> > >
> > > <== 03:12:51:05.11 (CM Time: 02:15:56:55.12).
> > > <== Q931: SETUP: from S[7040] L[1,186,0] E[26,185,0] SPA[--
> --]
> > > CR: 0,01 6C
> > > BC: speech
> > > 64 kbit/s
> > > circuit mode
> > > mu-law speech
> > > CID: 18
> > > LENGTH: 03
> > > Channel Selection Info: As Indicated in Following Octets
> > > D-Channel Indicator: D-Channel NOT indicated
> > > Preferred/Exclusive: Exclusive
> > > Interface type = primary rate
> > > Interface Identifier: IID Implicitly Identified
> > > Channel Type: B - Channel Units (3).
> > > Number Map: Channel is indicated by the number following.
> > > Coding Standard: CCITT
> > > Channel Number = 23
> > > DSP: ?OpenIT
> > > CDN: unknown
> > > unknown
> > > 14162045956
> > > --------------
> > >
> > >
> > > Current configuration : 3012 bytes
> > > !
> > > ! Last configuration change at 14:10:36 CST Thu Feb 3 2005
> > > ! NVRAM config last updated at 13:54:27 CST Tue Feb 1 2005
> > > !
> > > version 12.2
> > > service timestamps debug datetime msec
> > > service timestamps log datetime msec
> > > service password-encryption
> > > !
> > > hostname WPG-SIPgw-01
> > > !
> > > boot system slot0:c7200-is-mz.122-15.T14.bin
> > > card type t1 1
> > > logging queue-limit 100
> > > enable secret 5 ***REMOVED***
> > > !
> > > clock timezone CST -6
> > > clock summer-time CDT recurring 1 Sun Apr 2:00 4 Sun Oct 2:00
> > > dspint DSPfarm1/0
> > > !
> > > ip subnet-zero
> > > !
> > > !
> > > ip cef
> > > !
> > > isdn switch-type primary-dms100
> > > !
> > > !
> > > !
> > > voice service voip
> > > sip
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > !
> > > no voice hpi capture buffer
> > > no voice hpi capture destination
> > > !
> > > !
> > > mta receive maximum-recipients 0
> > > !
> > > !
> > > controller T1 1/0
> > > framing esf
> > > linecode b8zs
> > > cablelength long 0db
> > > pri-group timeslots 1-24
> > > !
> > > controller T1 1/1
> > > shutdown
> > > framing esf
> > > linecode b8zs
> > > cablelength long 0db
> > > !
> > > !
> > > !
> > > interface FastEthernet0/0
> > > ip address XXX.XXX.106.150 255.255.255.0
> > > duplex full
> > > !
> > > interface Serial1/0:23
> > > no ip address
> > > no logging event link-status
> > > isdn switch-type primary-dms100
> > > isdn incoming-voice voice
> > > no cdp enable
> > > !
> > > ip classless
> > > ip route 0.0.0.0 0.0.0.0 XXX.XXX.106.1
> > > no ip http server
> > > !
> > > !
> > > !
> > > access-list 10 permit XXX.XXX.106.4
> > > access-list 10 permit XXX.XXX.106.5
> > > !
> > > !
> > > call rsvp-sync
> > > !
> > > voice-port 1/0:23
> > > !
> > > !
> > > mgcp profile default
> > > !
> > > dial-peer cor custom
> > > !
> > > !
> > > !
> > > dial-peer voice 100 voip
> > > application session
> > > destination-pattern XXXXX54235
> > > progress_ind setup enable 3
> > > session protocol sipv2
> > > session target ipv4:XXX.XXX.99.50
> > > session transport udp
> > > !
> > > dial-peer voice 101 voip
> > > application session
> > > destination-pattern XXXXX54236
> > > progress_ind setup enable 3
> > > session protocol sipv2
> > > session target ipv4:XXX.XXX.102.59
> > > session transport udp
> > > !
> > > dial-peer voice 5000 pots
> > > application session
> > > destination-pattern .......
> > > direct-inward-dial
> > > port 1/0:23
> > > !
> > > dial-peer voice 102 voip
> > > application session
> > > destination-pattern XXXXX74002
> > > progress_ind setup enable 3
> > > session protocol sipv2
> > > session target ipv4:XXX.XXX.102.59
> > > session transport udp
> > > !
> > > dial-peer voice 103 voip
> > > application session
> > > destination-pattern XXXXX74003
> > > progress_ind setup enable 3
> > > session protocol sipv2
> > > session target ipv4:XXX.XXX.102.59
> > > session transport udp
> > > codec g711ulaw
> > > !
> > > dial-peer voice 104 voip
> > > application session
> > > destination-pattern XXXXX74004
> > > progress_ind setup enable 3
> > > session protocol sipv2
> > > session target ipv4:XXX.XXX.102.59
> > > session transport udp
> > > !
> > > dial-peer voice 105 voip
> > > destination-pattern XXXXX74005
> > > progress_ind setup enable 3
> > > session protocol sipv2
> > > session target ipv4:XXX.XXX.102.59
> > > session transport udp
> > > !
> > > dial-peer voice 5001 pots
> > > incoming called-number 1..........
> > > direct-inward-dial
> > > port 1/0:23
> > > prefix 1
> > > !
> > > sip-ua
> > > calling-info sip-to-pstn number set XXXXX74006
> > > !
> > > !
> > > gatekeeper
> > > shutdown
> > > !
> > > !
> > > line con 0
> > > stopbits 1
> > > line aux 0
> > > stopbits 1
> > > line vty 0 4
> > > access-class 10 in
> > > exec-timeout 60 0
> > > password ***REMOVED***
> > > login
> > > !
> > > ntp clock-period 17180028
> > > ntp server XXX.XXX.111.252
> > > !
> > > end
> > >
> > >
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
------------------------------
Message: 4
Date: Tue, 8 Feb 2005 11:30:25 -0800
From: "Voll, Scott" <Scott.Voll at wesd.org>
Subject: RE: [cisco-voip] Recording Phones
To: "Eric McKinney" <emckinney at fuquay-varina.org>,
<cisco-voip at puck.nether.net>
Message-ID: <D713462ED535184D830F6F6301753E7E010E7E04 at VISHNU.wesd.org>
Content-Type: text/plain; charset="us-ascii"
As far as RSPAN the only Pitfall I can think of is not being able to do
it over a WAN connection.
Scott
________________________________
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Eric McKinney
Sent: Tuesday, February 08, 2005 11:01 AM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Recording Phones
I have to record some of our phones and I have Mercom Audiolog for this.
I was planning on using RSPAN to record all the voice traffic. Has
anybody done this before? Any pitfalls for me to be aware of?
Thanks,
Eric McKinney
-------------- next part --------------
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URL:
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cc/attachment-0001.html
------------------------------
Message: 5
Date: Tue, 8 Feb 2005 11:31:49 -0800
From: "Voll, Scott" <Scott.Voll at wesd.org>
Subject: [cisco-voip] Unity VM Light
To: <cisco-voip at puck.nether.net>
Message-ID: <D713462ED535184D830F6F6301753E7E010E7E05 at VISHNU.wesd.org>
Content-Type: text/plain; charset="US-ASCII"
I'm running CM 3.3.4sr2 and Unity 4.0.4 and I've had some issues with VM
lights not syncing up. Is there a global way to resync rather then
rebooting the servers?
TIA
Scott
------------------------------
Message: 6
Date: Tue, 8 Feb 2005 13:35:02 -0600
From: "Walenta, Phil" <philip.walenta at berbee.com>
Subject: RE: [cisco-voip] Unity VM Light
To: "Voll, Scott" <Scott.Voll at wesd.org>, <cisco-voip at puck.nether.net>
Message-ID:
<D078DAD578D48A46A67AB71E6D232494CD57A0 at CTG-MSNEXC01.staff.berbee.com>
Content-Type: text/plain; charset="us-ascii"
If you go into the Integration manager, under the CallManager
integration you should find a "resync now" button, in addition to the
ability to schedule a resynch.
-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Voll, Scott
Sent: Tuesday, February 08, 2005 1:32 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Unity VM Light
I'm running CM 3.3.4sr2 and Unity 4.0.4 and I've had some issues with VM
lights not syncing up. Is there a global way to resync rather then
rebooting the servers?
TIA
Scott
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------------------------------
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