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Today's Topics:
1. Re: recommendations for handing off sip trunks as pri for
legacy (Joel Perez)
2. Re: Integrating CUPS with External Domain: Google Talk
(Eric Pedersen)
3. Cisco Business Edition 3000 and 7911 (Nikolay Shopik)
4. meetme - use specific conference resource (Erick Wellnitz)
5. Re: meetme - use specific conference resource (Wes Sisk)
6. Re: meetme - use specific conference resource (Erick Wellnitz)
7. Re: meetme - use specific conference resource (Erick Wellnitz)
8. Re: meetme - use specific conference resource (Wes Sisk)
9. Re: meetme - use specific conference resource (Erick Wellnitz)
10. External calls not able to transfer (Jimmy Thomas)
11. Cisco UCS community? (Wes Sisk)
12. SCCP 6945 to CME 9.1 fails (Jason Aarons (AM))
13. Re: recommendations for handing off sip trunks as pri for
legacy ([email protected])
14. 911 Conference call (Sandy Lee)
15. Re: 911 Conference call (Scott Voll)
16. call disconnect - trace file message (Erick Wellnitz)
----------------------------------------------------------------------
Message: 1
Date: Mon, 1 Apr 2013 12:09:33 -0400
From: Joel Perez <[email protected]>
To: Anthony Holloway <[email protected]>
Cc: Mike <[email protected]>, Cisco VoIP Group
<[email protected]>
Subject: Re: [cisco-voip] recommendations for handing off sip trunks
as pri for legacy
Message-ID:
<CABdWoUEkSME8yZXBvL6Jf2=jicz9ihauoamv_v+4gkmhz1k...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Anthony put it more eloquently than i ever could have. All his points are
right on and those are the rules we try to follow as often as possible.
Joel P.
On Sun, Mar 31, 2013 at 10:43 PM, Anthony Holloway <
[email protected]> wrote:
> Unless I missed part of your requirements, I don't see where you'll need a
> CUBE license. You are not doing IP-to-IP call legs (E.g., SIP-to-SIP or
> H323-to-SIP or H323-to-H323), but instead doing SIP-to-POTS.
>
> The config is very simple, but just like configuring a PRI for PSTN
> connectivity, you need to sync up your settings with the telco for SIP.
> I.e., Call control ip address and port, proxy settings, expected diversion
> header, authentication mechanisms, early offer, etc.
>
> The complete SIP guide on a router can be found in the CVOICE book/course
> material. I highly recommend you read this book.
>
>
> http://www.amazon.com/Implementing-Unified-Communications-Foundation-Learning/dp/1587204193/ref=sr_1_1?ie=UTF8&qid=1364782855&sr=8-1&keywords=cvoice+642
>
> Below are some of my personal notes on SIP on a voice gateway.
>
> ! SIP Gateway Configurations
> !
> ==============================================================================
>
> ! SIP gateways have two things:
> ! 1. SIP-UA (Optionally the destination can be set at the DP level)
> ! 2. VoIP (SIP) Dial Peers
>
> ! To configure the SIP UA which contains:
> ! Authentication (Optional)
> ! SIP Servers (Registrar and Proxy)
> sip-ua
> ! To specify a sip-server (you can also type this at the DP level)
> ! This is how you configure the SIP proxy you point at
> sip-server ipv4:192.168.3.1:5060
> !
>
> ! SIP VoIP Dial Peers have these two things:
> ! 1. Session Protocol defined as SIP
> ! 2. A session target pointing at the SIP UA
>
> ! The first thing you need to do is change the default call control protocol
> ! from H.323 to SIP, then you need to add a session target
> dial-peer voice 10 voip
> session protocol sipv2
> session target ipv4:192.168.3.1:5060
> ! OR if you specified the server in the sip-ua section
> session target sip-server
> !
>
> ! SIP uses UDP for outbound signaling by default, but you can change it if you
> ! want to or if you have to (ITSP standard)
> voice service voip
> sip
> session transport tcp
> ! Or if you want to UDP it
> ! session transport udp
> !
>
> ! Or at the DP level
> dial-peer voice 10 voip
> session transport tcp
> !
>
> ! To bind to a source address globally (only way)
> voice service voip
> sip
> bind control source-interface Loopback0
> bind media source-interface Loopback0
> ! Or bind both in a single command (think of this like a macro for the
> above two commands)
> ! bind all source-interface Loopback0
> !
>
> ! The only way you can tune SIP timers is in SIP-UA mode
> sip-ua
> ! To cut the default timer in half for how long an INVITE is valid
> timers expires 90000
> !
>
> ! If you have an ISDN->SIP gateway, and you would like Calling Name display
> voice service voip
> signaling forward unconditional
> !
> inteface Serial0/0/0:23
> isdn supp-service name calling
> !
>
> ! If you wish to enable CLID privacy (aka blocking)
> dial-peer voice 10 voip
> clid strip pi-restrict
> !
>
> ! If you wish to enable mapping the calling number into the display name field
> voice service voip
> clid substitute name
> !
>
> ! DTMF will be inband with the audio stream unless you say otherwise
>
> dial-peer voice 10 voip
> ! This will try to use SIP NOTIFY first, then RTP-NTE second and prevent
> ! any DTMF from ever being inserted into the voice stream
> dtmf-relay sip-notify rtp-nte digit-drop
> !
>
> ! Fax support is on by default and is cisco fax relay
> ! To change this to another type, you could
> voice service voip
> ! This configures t38 followed by pass-through globally
> fax protocol t38 fallback pass-through g711ulaw
> ! You could also disable ECM
> fax-relay ecm disable
> ! Or even prevent SG3 faxing (downgrades to G3)
> fax-relay sg3-to-g3
> !
>
> ! By default the fax rate is "voice" which means "as fast a possible"
> ! If you wanted to slow it down, you have to do it at the DP level
> dial-peer voice 10 voip
> ! This slows down the transmission to 14.4kbps
> fax rate 14400
> ! If you want the dial-peer to take on the fax-relay config from global
> fax-relay system
> !
>
> ! Modem passthrough is a breakthrough
> voice service voip
> modem passthrough nse codec g711ulaw
> !
>
> ! If you want a dial-peer to use the global settings you need to...
> dial-peer voice 10 voip
> modem passthrough system
> !
>
> ! Modem relay is better, and will auto fallback to passthrough.
> voice service voip
> modem relay nse codec g711ulaw
> !
>
> ! If you want a dial-peer to use the global settings...oh wait, they can't
> dial-peer voice 10 voip
> modem relay nse codec g711ulaw
> !
>
> ! If you're looking to change the codec of a voip call, you can do it globally
> ! with the added benefit of having a prioritized list...
> voice class codec 1
> codec preference 1 g711ulaw
> codec preference 2 g729r8
> !
> dial-peer voice 10 voip
> voice-class codec 1
> !
>
> ! Or directly on the dial-peer, but at the cost of only a single codec
> dial-peer voice 10 voip
> codec g711ulaw
> !
>
> ! For CUBE, you can actually just tell the DP to pass whatever it was given
> voice class codec 1
> codec preference 1 transparent
> codec preference 2 g711ulaw
> codec preference 3 g729r8
> !
> dial-peer voice 10 voip
> voice-class codec 1
> !
>
> ! Verifying SIP
>
> !
> ==============================================================================
>
> ! Verify that the SIP feature is UP
>
> show sip-ua service ! SIP Service is up
>
> ! debug the SIP messages
> debug ccsip messages
>
>
>
> On Sun, Mar 31, 2013 at 3:34 PM, Mike <[email protected]> wrote:
>
>> Chris,
>>
>> I've done this several times. You will need CUBE licenses and a multi-flex
>> T1 card (MFT), but you easily convert the SIP handoff to PRI.
>>
>> -----Original Message-----
>> From: [email protected]
>> [mailto:[email protected]] On Behalf Of chris
>> Sent: Sunday, March 31, 2013 9:28 AM
>> To: [email protected]
>> Subject: [cisco-voip] recommendations for handing off sip trunks as pri
>> for
>> legacy
>>
>> We have recently acquired a new location which has a legacy analog pbx
>> and a
>> carrier who was providing service as a PRI.
>>
>> The carrier has now stated they will no longer be supporting PRI and are
>> recommending a switch to SIP trunks, which would be fine if they provided
>> adtran or similar to handle the conversion.
>>
>> The carrier says they do not get involved in this situation and its up to
>> the customer, so we are left to fend for ourselves :)
>>
>> This loction already has a 2851 with ipvoice, and I was thinking I should
>> be
>> able to do everything I need with that?
>>
>> Googling seems to turn up lots of configs which simply terminate local
>> calls
>> to a physically connected PRI, when in actuality I want to do the inverse
>> and use the cisco to hand off a traditional PRI to the analog system.
>>
>> Anyone gone down this path before? Have any config examples of what I
>> described that I can reference?
>>
>> Also what type of linecard would I need to handoff the PRI? Does it need
>> to
>> be a MFT T1 card or plain t1 dsu/csu?
>>
>> thanks in advance
>> chris
>> _______________________________________________
>> cisco-voip mailing list
>> [email protected]
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>> _______________________________________________
>> cisco-voip mailing list
>> [email protected]
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 2
Date: Mon, 1 Apr 2013 16:32:44 +0000
From: Eric Pedersen <[email protected]>
To: Kenneth Hayes <[email protected]>, Cisco VoIPoE List
<[email protected]>
Subject: Re: [cisco-voip] Integrating CUPS with External Domain:
Google Talk
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Hi Kenneth,
Google already has the gmail.com SRV records so you don't need to add those to
your DNS. All you need to add to your DNS are the _xmpp-server SRV records for
your CUPS servers. The federated domain you need to put in CUPS should just be
gmail.com. The dialback secret confused me too. You can put any string in
there. As I recall, when you send an XMPP message to another domain, their XMPP
server will send a verification request message back to your XMPP server with
that string to make sure your original message wasn't forged.
Eric
From: [email protected]
[mailto:[email protected]] On Behalf Of Kenneth Hayes
Sent: 28 March 2013 10:37 AM
To: Cisco VoIPoE List
Subject: [cisco-voip] Integrating CUPS with External Domain: Google Talk
Hello all,
This morning I've been working on my integration for Google Talk and
CUPS/Jabber. From what I was reading I need to add the following information in
my public DNS:
_xmpp-server._tcp.gmail.com<http://tcp.gmail.com>. IN SRV 5 0 5269
xmpp-server.l.google.com<http://xmpp-server.l.google.com>.
_xmpp-server._tcp.gmail.com<http://tcp.gmail.com>. IN SRV 20 0 5269
alt1.xmpp-server.l.google.com<http://alt1.xmpp-server.l.google.com>.
_xmpp-server._tcp.gmail.com<http://tcp.gmail.com>. IN SRV 20 0 5269
alt2.xmpp-server.l.google.com<http://alt2.xmpp-server.l.google.com>.
_xmpp-server._tcp.gmail.com<http://tcp.gmail.com>. IN SRV 20 0 5269
alt3.xmpp-server.l.google.com<http://alt3.xmpp-server.l.google.com>.
_xmpp-server._tcp.gmail.com<http://tcp.gmail.com>. IN SRV 20 0 5269
alt4.xmpp-server.l.google.com<http://alt4.xmpp-server.l.google.com>.
Now in my DNS provider I enter the SRV=xmpp-server and the protocol= tcp, etc.
etc. but the target I'm assuming I need to add the
"xmpp-server.l.google.com<http://xmpp-server.l.google.com>, just need
confirmation on that...here's the thing the section
"xmpp-server._tcp.gmail.com<http://tcp.gmail.com>" I'm suppose to replace the
".gmail.com<http://gmail.com>" with my domain, but I can't place that
information anywhere (so I'm assuming this is a provider issue?).
Now on to the Presence configuration, I'm confused on the dialback secret, I'm
not sure what I'm suppose to put there so I need some claifications there. When
Presence wants me to configure the Federated Domain do I put
"xmpp-server.l.google.com<http://xmpp-server.l.google.com>" or do I simply put
l.google.com<http://l.google.com>?
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Message: 3
Date: Mon, 01 Apr 2013 20:42:47 +0400
From: Nikolay Shopik <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] Cisco Business Edition 3000 and 7911
Message-ID: <[email protected]>
Content-Type: text/plain; charset=UTF-8
Hey,
We currently using CME with 2 sites and thinking about upgrade to Cisco
Business Edition 3000, which seems enough for us at moment. But most our
phones are 7911, and datasheet for Cisco Business Edition 3000 doesn't
mention them as supported.
Anyone could confirm/deny we could use 7911 with Cisco Business Edition
3000? As I may believe this is just marketing bs.
------------------------------
Message: 4
Date: Mon, 1 Apr 2013 12:14:05 -0500
From: Erick Wellnitz <[email protected]>
To: cisco-voip <[email protected]>
Subject: [cisco-voip] meetme - use specific conference resource
Message-ID:
<cak0wosb686r9elp7vcgvbsvwi9afvekhwt2wmasdpmeqm-p...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hello all!
I have a situation where i need to make my meet-me numbers use a specific
CFB resource, specifically a software CFB. My problem arises because I
want ad-hoc conferences to keep using the hardware CFB so changing the
order of MRG in the MRGL assigned to phones is out.
Any ideas are greatly appreciated.
Thanks in advance!
Erick
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Message: 5
Date: Mon, 1 Apr 2013 13:19:59 -0400
From: Wes Sisk <[email protected]>
To: Erick Wellnitz <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] meetme - use specific conference resource
Message-ID: <[email protected]>
Content-Type: text/plain; charset=iso-8859-1
For some reason I thought this was possible by initiating the meet me from a
dedicated device (phone/cipc) that uses a specific MRGL?
This is quite fuzzy to me so I am more than willing to be wrong. Just sharing
an idea.
-wes
On Apr 1, 2013, at 1:14 PM, Erick Wellnitz <[email protected]> wrote:
Hello all!
I have a situation where i need to make my meet-me numbers use a specific CFB
resource, specifically a software CFB. My problem arises because I want ad-hoc
conferences to keep using the hardware CFB so changing the order of MRG in the
MRGL assigned to phones is out.
Any ideas are greatly appreciated.
Thanks in advance!
Erick
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip
------------------------------
Message: 6
Date: Mon, 1 Apr 2013 12:34:27 -0500
From: Erick Wellnitz <[email protected]>
To: Wes Sisk <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] meetme - use specific conference resource
Message-ID:
<cak0wosd3gozgajgyowzig-akignnrweakkgwa+w5en7rpb5...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
I think that would work but we have to be able to initiate from the phone
of any IT person.
My ideal outcome is:
>From any phone go off hook, press meetme, dial conference number -
conference uses software CFB.
>From any phone initiate an ad-hoc conference - conference uses hardware CFB.
I haven't found a way to do this but it was an idea that I thought I should
investigate.
I can always set the maximum conference-participants on the IOS CFB higher
but that would also allow an ad-hoc conference to have more participants.
On Mon, Apr 1, 2013 at 12:19 PM, Wes Sisk <[email protected]> wrote:
> For some reason I thought this was possible by initiating the meet me from
> a dedicated device (phone/cipc) that uses a specific MRGL?
>
> This is quite fuzzy to me so I am more than willing to be wrong. Just
> sharing an idea.
>
> -wes
>
> On Apr 1, 2013, at 1:14 PM, Erick Wellnitz <[email protected]>
> wrote:
>
> Hello all!
>
> I have a situation where i need to make my meet-me numbers use a specific
> CFB resource, specifically a software CFB. My problem arises because I
> want ad-hoc conferences to keep using the hardware CFB so changing the
> order of MRG in the MRGL assigned to phones is out.
>
> Any ideas are greatly appreciated.
>
> Thanks in advance!
>
> Erick
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 7
Date: Mon, 1 Apr 2013 12:37:01 -0500
From: Erick Wellnitz <[email protected]>
To: Wes Sisk <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] meetme - use specific conference resource
Message-ID:
<CAK0wOsDquQw56KH6hHGYp4XjS-SZHd==hbuhxdrvm1a8f4y...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Actually, it won't increase the number for ad-hoc if I leave the service
parameter for maximum ad-hoc participants as-is.
On Mon, Apr 1, 2013 at 12:34 PM, Erick Wellnitz <[email protected]>wrote:
> I think that would work but we have to be able to initiate from the phone
> of any IT person.
>
> My ideal outcome is:
> From any phone go off hook, press meetme, dial conference number -
> conference uses software CFB.
> From any phone initiate an ad-hoc conference - conference uses hardware
> CFB.
> I haven't found a way to do this but it was an idea that I thought I
> should investigate.
>
> I can always set the maximum conference-participants on the IOS CFB
> higher but that would also allow an ad-hoc conference to have more
> participants.
>
>
>
>
> On Mon, Apr 1, 2013 at 12:19 PM, Wes Sisk <[email protected]> wrote:
>
>> For some reason I thought this was possible by initiating the meet me
>> from a dedicated device (phone/cipc) that uses a specific MRGL?
>>
>> This is quite fuzzy to me so I am more than willing to be wrong. Just
>> sharing an idea.
>>
>> -wes
>>
>> On Apr 1, 2013, at 1:14 PM, Erick Wellnitz <[email protected]>
>> wrote:
>>
>> Hello all!
>>
>> I have a situation where i need to make my meet-me numbers use a specific
>> CFB resource, specifically a software CFB. My problem arises because I
>> want ad-hoc conferences to keep using the hardware CFB so changing the
>> order of MRG in the MRGL assigned to phones is out.
>>
>> Any ideas are greatly appreciated.
>>
>> Thanks in advance!
>>
>> Erick
>> _______________________________________________
>> cisco-voip mailing list
>> [email protected]
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>
>>
>
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Message: 8
Date: Mon, 1 Apr 2013 13:58:12 -0400
From: Wes Sisk <[email protected]>
To: Erick Wellnitz <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] meetme - use specific conference resource
Message-ID: <[email protected]>
Content-Type: text/plain; charset="iso-8859-1"
I'm not aware of anything inside CUCM that allows the same device to
differentiate MRGL for adhoc vs. meet me.
I have worked with a few patterns who wanted this in the past. We sent them
down the route of a custom application. Something like a UCCX custom
application. Use a telephony trigger that invokes a script that remote controls
a dedicated phone and initiates the conference.
-wes
On Apr 1, 2013, at 1:34 PM, Erick Wellnitz <[email protected]> wrote:
I think that would work but we have to be able to initiate from the phone of
any IT person.
My ideal outcome is:
>From any phone go off hook, press meetme, dial conference number - conference
>uses software CFB.
>From any phone initiate an ad-hoc conference - conference uses hardware CFB.
I haven't found a way to do this but it was an idea that I thought I should
investigate.
I can always set the maximum conference-participants on the IOS CFB higher but
that would also allow an ad-hoc conference to have more participants.
On Mon, Apr 1, 2013 at 12:19 PM, Wes Sisk <[email protected]> wrote:
For some reason I thought this was possible by initiating the meet me from a
dedicated device (phone/cipc) that uses a specific MRGL?
This is quite fuzzy to me so I am more than willing to be wrong. Just sharing
an idea.
-wes
On Apr 1, 2013, at 1:14 PM, Erick Wellnitz <[email protected]> wrote:
Hello all!
I have a situation where i need to make my meet-me numbers use a specific CFB
resource, specifically a software CFB. My problem arises because I want ad-hoc
conferences to keep using the hardware CFB so changing the order of MRG in the
MRGL assigned to phones is out.
Any ideas are greatly appreciated.
Thanks in advance!
Erick
_______________________________________________
cisco-voip mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/cisco-voip
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Message: 9
Date: Mon, 1 Apr 2013 13:24:03 -0500
From: Erick Wellnitz <[email protected]>
To: Wes Sisk <[email protected]>
Cc: cisco-voip <[email protected]>
Subject: Re: [cisco-voip] meetme - use specific conference resource
Message-ID:
<cak0wosafadbzxenp+jj1d+ccc5tgiskoa3xzis_oqhabdyw...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
I'm going to end up increasing the maximum participants on the hardware
CFB. It works and there is no messing around with MRGLs.
I can think of a number of ways to do what I want to do in a secure manner
but it requires on premesis conferencing or UCCX as suggested.
On Mon, Apr 1, 2013 at 12:58 PM, Wes Sisk <[email protected]> wrote:
> I'm not aware of anything inside CUCM that allows the same device to
> differentiate MRGL for adhoc vs. meet me.
>
> I have worked with a few patterns who wanted this in the past. We sent
> them down the route of a custom application. Something like a UCCX custom
> application. Use a telephony trigger that invokes a script that remote
> controls a dedicated phone and initiates the conference.
>
> -wes
>
> On Apr 1, 2013, at 1:34 PM, Erick Wellnitz <[email protected]>
> wrote:
>
> I think that would work but we have to be able to initiate from the phone
> of any IT person.
>
> My ideal outcome is:
> From any phone go off hook, press meetme, dial conference number -
> conference uses software CFB.
> From any phone initiate an ad-hoc conference - conference uses hardware
> CFB.
> I haven't found a way to do this but it was an idea that I thought I
> should investigate.
>
> I can always set the maximum conference-participants on the IOS CFB
> higher but that would also allow an ad-hoc conference to have more
> participants.
>
>
>
>
> On Mon, Apr 1, 2013 at 12:19 PM, Wes Sisk <[email protected]> wrote:
>
>> For some reason I thought this was possible by initiating the meet me
>> from a dedicated device (phone/cipc) that uses a specific MRGL?
>>
>> This is quite fuzzy to me so I am more than willing to be wrong. Just
>> sharing an idea.
>>
>> -wes
>>
>> On Apr 1, 2013, at 1:14 PM, Erick Wellnitz <[email protected]>
>> wrote:
>>
>> Hello all!
>>
>> I have a situation where i need to make my meet-me numbers use a specific
>> CFB resource, specifically a software CFB. My problem arises because I
>> want ad-hoc conferences to keep using the hardware CFB so changing the
>> order of MRG in the MRGL assigned to phones is out.
>>
>> Any ideas are greatly appreciated.
>>
>> Thanks in advance!
>>
>> Erick
>> _______________________________________________
>> cisco-voip mailing list
>> [email protected]
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Message: 10
Date: Mon, 1 Apr 2013 15:12:43 -0400
From: Jimmy Thomas <[email protected]>
To: [email protected]
Subject: [cisco-voip] External calls not able to transfer
Message-ID:
<CAAOGEkwcjvUeSzmUVW40apgHpS_4zJdT5xc00Ry30prC0Wp=n...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
All,
I've the following issue with a customer of mine. The CME is configured
with 1 PRI circuit and everything seems to be working except the external
call transfers. The version of CME is 8.8. Any time I have an external call
coming in, my users cannot transfer the call internally to another
extension. They are using 4 digit extentions. The following is the
telephony-services configuration. Your assistance is greatly appreciated.
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 2 register MTPacf2c55d1d50
associate profile 1 register CNFacf2c55d1d50
!
dspfarm profile 2 transcode
codec g729r8
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 1
associate application SCCP
!
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 1
associate application SCCP
!
telephony-service
sdspfarm units 2
sdspfarm transcode sessions 4
sdspfarm tag 1 CNFacf2c55d1d50
sdspfarm tag 2 MTPacf2c55d1d50
authentication credential CMEXXadmin xxxxxx
xml user xmluser password xxxxxx
max-ephones 50
max-dn 150
ip source-address <CME IP> port 2000
auto assign 1 to 50
service dnis dir-lookup
system message Federal Public Defender
url services http://<unity IP>/voiceview/common/login.do
url authentication http://<<unity IP>>/CCMCIP/authenticate.asp
load 7915-12 B015-1-0-4.SBN
load 7945 term45.default.loads
load 7965 term65.default.loads
time-zone 8
dialplan-pattern 1 405-609-59.. extension-length 4
keepalive 120
voicemail 8900
max-conferences 4 gain -6
call-forward pattern .T
moh "flash:/music-on-hold.au"
multicast moh 239.10.16.4 port 16384 route 10.193.110.2
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 9405.......
transfer-pattern 59..
transfer-pattern .T
transfer-pattern 1001 blind
transfer-pattern 7001 blind
secondary-dialtone 9
directory last-name-first
create cnf-files version-stamp 7960 Mar 29 2013 08:23:42
!
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn send-alerting
isdn sending-complete
isdn outgoing display-ie
isdn outgoing ie redirecting-number
no isdn gtd
no cdp enable
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Message: 11
Date: Mon, 1 Apr 2013 16:20:41 -0400
From: Wes Sisk <[email protected]>
To: "cisco-voip ([email protected])"
<[email protected]>
Subject: [cisco-voip] Cisco UCS community?
Message-ID: <[email protected]>
Content-Type: text/plain; charset=us-ascii
I really appreciate the interactions and exchanges on cisco-voip.
Is there a similar community for Cisco UCS or virtualization? Where do folks
go for good discussions, well rounded responses, and answers?
Regards,
Wes
------------------------------
Message: 12
Date: Mon, 1 Apr 2013 20:58:15 -0400
From: "Jason Aarons (AM)" <[email protected]>
To: "cisco-voip ([email protected])"
<[email protected]>
Subject: [cisco-voip] SCCP 6945 to CME 9.1 fails
Message-ID:
<4e38db0a1959b04c8c83edcf069b53ed0d2ecef...@usispclexdb01.na.didata.local>
Content-Type: text/plain; charset="windows-1252"
Out of box 6945s don't register to CME 9.1 (15.2.4M). Other SCCp phones (eg
8945)work fine.
Debug ephone register show nothing with these 6945s...I see debug tftp packets,
I can get to phone's web page.
I'll get more details tomorrow, wondering if anyone else has seen this or if
it's a known issue.
-jason
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Message: 13
Date: Tue, 2 Apr 2013 08:16:42 +0000
From: <[email protected]>
To: <[email protected]>, <[email protected]>,
<[email protected]>
Subject: Re: [cisco-voip] recommendations for handing off sip trunks
as pri for legacy
Message-ID:
<f8e0cc3253a10c4cb137f12f568dad061b1facb...@gblonz-pmsgem02.emrsn.org>
Content-Type: text/plain; charset="us-ascii"
Now here's a question:
CUBE, as far as I know, is IP to IP gateway. So why would one need CUBE
licenses for SIP to PRI?
Your IPVoice feature set should do the trick just fine (not as if CUBE would be
anywhere enforced, but I digress. I actually don't think this setup needs CUBE
licensing).
However, keep in mind that you will also need DSPs (PVDM modules) for the T1, a
PVDM2-32 per T1.
You would probably need to act as network side on the PRI for your PBX.
Here's some sample config, although your switch-type may vary, as well as you
may need to use different patterns, number translations etc.:
card type t1 0 0
!
network-clock-participate wic 0
network-clock-select 1 T1 0/0/0
!
isdn switch-type primary-qsig
!
controller T1 0/0/0
cablelength long 0db
pri-group timeslots 1-24
!
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn incoming-voice voice
no cdp enable
!
voice-port 0/0/0:23
!
dial-peer voice 2 voip
description SIPTrunk
destination-pattern 9T
session protocol sipv2
session target ipv4:10.20.30.40
incoming called-number .
voice-class sip options-keepalive up-interval 20 down-interval 10
dtmf-relay rtp-nte
no vad
supplementary-service pass-through
!
dial-peer voice 3 pots
description PBX-PRI
destination-pattern +T
incoming called-number .
direct-inward-dial
port 0/0/0:23
!
Cheers,
Zoltan Kelemen
ETS & Information Security
Implementation Engineering
w: +40 374 132356 | m: +40 757 039093
-----Original Message-----
From: [email protected]
[mailto:[email protected]] On Behalf Of Mike
Sent: Sunday, March 31, 2013 11:35 PM
To: 'chris'; [email protected]
Subject: Re: [cisco-voip] recommendations for handing off sip trunks as pri for
legacy
Chris,
I've done this several times. You will need CUBE licenses and a multi-flex
T1 card (MFT), but you easily convert the SIP handoff to PRI.
-----Original Message-----
From: [email protected]
[mailto:[email protected]] On Behalf Of chris
Sent: Sunday, March 31, 2013 9:28 AM
To: [email protected]
Subject: [cisco-voip] recommendations for handing off sip trunks as pri for
legacy
We have recently acquired a new location which has a legacy analog pbx and a
carrier who was providing service as a PRI.
The carrier has now stated they will no longer be supporting PRI and are
recommending a switch to SIP trunks, which would be fine if they provided
adtran or similar to handle the conversion.
The carrier says they do not get involved in this situation and its up to the
customer, so we are left to fend for ourselves :)
This loction already has a 2851 with ipvoice, and I was thinking I should be
able to do everything I need with that?
Googling seems to turn up lots of configs which simply terminate local calls to
a physically connected PRI, when in actuality I want to do the inverse and use
the cisco to hand off a traditional PRI to the analog system.
Anyone gone down this path before? Have any config examples of what I described
that I can reference?
Also what type of linecard would I need to handoff the PRI? Does it need to be
a MFT T1 card or plain t1 dsu/csu?
thanks in advance
chris
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------------------------------
Message: 14
Date: Tue, 2 Apr 2013 10:05:36 -0400
From: Sandy Lee <[email protected]>
To: "[email protected]" <[email protected]>
Subject: [cisco-voip] 911 Conference call
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Hi,
Our on-site security service actually receives all the 911 calls and everything
is good. But now they want the 911 calls sent directly to the public 911 AND be
able to listen to the call. They only need to listen to what's going on and
where the emergency is from and what's the nature of the call, so that they can
help the local services when they come in the campus. I was thinking of a
conference call, but not sure how to do this automatically without a 3rd party
app. Any ideas would be appreciated.
Thanks and regards.
Sandy.
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Message: 15
Date: Tue, 2 Apr 2013 07:29:29 -0700
From: Scott Voll <[email protected]>
To: Sandy Lee <[email protected]>
Cc: "[email protected]" <[email protected]>
Subject: Re: [cisco-voip] 911 Conference call
Message-ID:
<cahgd+38uxjgu0fmj0f-qbzai+ra4qbt4hyr2amc_qqtjpqj...@mail.gmail.com>
Content-Type: text/plain; charset="windows-1252"
The only things I can think of would be third party apps. I believe 911
enable can do this. redsky might also.
Non E911 solutions might be a third party recording / monitoring app, but I
would suggest a true e911 app before I would do that.
Wish I could think of something better......
YMMV
Scott
On Tue, Apr 2, 2013 at 7:05 AM, Sandy Lee <[email protected]> wrote:
> Hi,****
>
> ** **
>
> Our on-site security service actually receives all the 911 calls and
> everything is good. But now they want the 911 calls sent directly to the
> public 911 AND be able to listen to the call. They only need to listen to
> what?s going on and where the emergency is from and what?s the nature of
> the call, so that they can help the local services when they come in the
> campus. I was thinking of a conference call, but not sure how to do this
> automatically without a 3rd party app. Any ideas would be appreciated.****
>
> ** **
>
> Thanks and regards.****
>
> ** **
>
> Sandy.****
>
> ** **
>
> ** **
>
> _______________________________________________
> cisco-voip mailing list
> [email protected]
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
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Message: 16
Date: Tue, 2 Apr 2013 10:58:04 -0500
From: Erick Wellnitz <[email protected]>
To: cisco-voip <[email protected]>
Subject: [cisco-voip] call disconnect - trace file message
Message-ID:
<CAK0wOsA-7Lfy8vD6ZMT7-Z=q0jeh4s5qmozfuk89rstsfyv...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
I have a rather strange issue. At random, calls are disconnected to one
particular number. When this happens, I see the following in the traces:
|CCM_PI: CPIClass::incValue(), Caller's error, inappropriate
InstanceIndex[298] > m_nMaxCreatedInstanceIndex[6] or
m_nMaxCreatedInstanceIndex == -1. PIClass(Enum=2)|*^*^*
Anyone know what this message means? MGCP and Q.931 see it as a normal
call clearing.
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End of cisco-voip Digest, Vol 114, Issue 2
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