20 Analog Trunks on CME 2911

Hi guys.
We have a current requirement that has the following specifications.
Minimum of 20 Analog trunks, 1 x E1 PRI ISDN trunk, with SIP license (60 licenses), with IVR, call data recording
I have configured a 2911 with the appropriate licenses in CCW. But, how can I come up with a solution for 20 analog trunks? I am not yet familiar with this.
Thanks in advance.

EVM-HD-8FXS/DID High density voice/fax extension module -8 FXS/DID
EVM-HD-8FXS/DID
The Cisco High-Density Extension Module for voice and fax
has 8 FXS and DID ports. Individual ports on the basebo
ard
module can be configured for FXS or DID signaling. Adj
acent ports should share the same configuration to avoid i
mpedance
setting conflicts. A change to the impedance setting for o
ne port, changes the setting on the adjacent port. Pair
ed ports are: 0
and 1; 2 and 3; 4 and 5; 6 and 7.
An on-premises FXS interface connects directly to a standar
d telephone, fax machine, or similar device and supplies
ring,
voltage, and dial tone. Signaling support available in
FXS mode includes loop-start and ground-start. DID tru
nks from the
central office can be connected to the Cisco High-Density E
xtension Module baseboard for off-premises connections.
Signaling support available in DID mode includes immedi
ate, delay dial, and wink start.
Users plug in up to two expansion modules in any combina
tion to increase the voice and fax capacity of the Cisco Hi
gh-
Density Extension Module baseboard.
or
EM3-HDA-8FXS/DID 8-port voice/fax expansion module-FXS/DID
EM3-HDA-8FXS/DID
This 8-port FXS/DID Voice and Fax Expansion Module pro
vides on-premises FXS signaling to connect directly to a sta
ndard
telephone, fax machine, or similar device and supplies ri
ng, voltage, and dial tone. Signaling support availabl
e in FXS mode
includes loop-start and ground-start. This expansion modu
le also works with the Cisco High-Density Analog Voice an
d Fax
Network Module (part number NM-HDA-4FXS).
DID trunks from the central office can be connected to t
he Cisco Voice and Fax Expansion Module for off-premises
connections. Signaling support available in DID mode incl
udes immediate, delay dial, and wink start. It support
s up to eight
ports in DID mode, both expansion slots combined (with ei
ght DIDs on EVM baseboard, up to 16 DIDs maximum).
Additionally, it supports robust front end protection.

Similar Messages

Analog trunk problem

I have an analog trunk plugged into a 2811 with a VIC2-4FXO. When the line is in use, if a call comes in on it, the caller hears one ring, then a busy signal (not just a busy signal). What would be causing this? It happens to be the last line in a hunt group that is set at the CO to forward on busy to another hunt group, but the forward isn't happening, probably because of that first ringback.

Likely the problem is not due to the router. If you do "debug vpm signal" you will see the incoming ring and port going off-hook. You should see that for engaged ports, no new call is presented, and is the router does nothing in generating ringback, as you said the line is busy already when the call comes in.
hope this helps, please rate post if it does!

How to Remove port number for SIP trunk in CME

Hi,
I trying to set a SIP trunk with SIP provider, I have CME 7.1
The trunk is registered now but I can´t make calsl via SIP provider. After some debbugs sip provider's staff told me that the solution is not
not append the port in the INVITE.
Is it possible to do this?, How?
I have found some info about normalization but is relating to CM server not CME.
regards

what port number does your provider use for signalling? They need to provide you the port number if its different from the standard 5060..
You can then configure the signalling ports on your dial-peer as shown in example below..where port 5081 is used here
dial-peer voice 1 voip
destination-pattern .T
session protocol sipv2
session target ipv4:10.10.10.24:5081
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"

Which files should a CME 2911 router shipped with ?

we have installation of CME on 2911 but on the flash we have only this
yourname#sh flash
-#- --length-- -----date/time------ path
1     62662920 Sep 25 2010 01:42:54 c2900-universalk9-mz.SPA.150-1.M3.bin
2         2903 Sep 25 2010 01:57:40 cpconfig-29xx.cfg
3      2915328 Sep 25 2010 01:57:52 cpexpress.tar
4         1038 Sep 25 2010 01:58:02 home.shtml
5       115712 Sep 25 2010 01:58:08 home.tar
6      1697952 Sep 25 2010 01:58:22 securedesktop-ios-3.1.1.45-k9.pkg
7       415956 Sep 25 2010 01:58:34 sslclient-win-1.1.4.176.pkg
shouldn't be shipped with CME and phone load files ? or is it my responsibility to install and load them on the router ?!!

The CME files are IOS version specific so they don't ship on new routers. They always have to be downloaded separately. You can find instructions on how to install the files in the CME Admin Guide,
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeinstl.html#wp1077212
-Felipe

Best practice about dial-peer creating when using analog lines

Hi,
I am trying to find out what is the best practice when creating dial-peer for analog lines on CME, should I use trunk group or create separate dial-peer for each FXO ports? If I use trunk group, is there any advantage ( lesser dial-peer) or disadvantage?
Thanks!

The advantage of trunk groups is that a single dial peer can point to for instance PSTN, rather then multiple dialpeers, with varying preference, each pointing to a separate FXO. Funtionally I can't see much difference. So I guess it also comes down to personal preference.
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Trunk lines

Hi All ,
Just a quick one, i need to connect 32 Analog trunk lines to a PABX and 4 PRI's to the PSTN. any Suggestions on which Cards i could use on the ISR to achieve this
Thanks in Advance

Hi,
from the PBX point of view, trunk lines are FXO, that is, analog lines that normally go to the pstn. This means the voice gateway will need FXS ports.
If this is your requirement, be aware that the integration may suffer of problems, for example you will not be able to give DID to the PBX.
Anyway, considering the density, the minimum you can use is a 2821 with EVM-HD-8FXS/DID + 2 x EM-HDA-8FXS, 2 x VIC-4FXS, 2 x VWIC2-2T1E1, 1 x PVDM2-32.
Hope this helps, please rate post if it does!

FXS versus FXO cards in 2821

I am new to Cisco CME and VOIP and am trying to configure a new 2821 with CME 4.1. I have two 4 port FXS cards. The intent was to use the ports on these cards to connect to PSTN POTS lines for inbounce and outbound cards. Can I use these FXS cards for this purpose or do they need to be FXO cards? According to the description they do as the FXS is intended to have phones connected and supplies dial-tone and the FXO does not supply dial-tone and is intended to be connected to PSTN. The reason that I am a little confused is that I have configured the system to dial out but was having issues where it would not hang up the line.

Hi Michael,
Here is some background info to add to Paolos always Great info (hey P.);
Analog Telephony Protocols
Analog telephony signaling, the original signaling protocol, provides the method for connecting or disconnecting calls on analog trunks. By using direct current (DC) over two-wire or four-wire circuits to signal on-hook and off-hook conditions, each analog trunk connects analog endpoints or devices such as a PBX or analog phone.
To provide connections to legacy analog central offices and PBXs, Cisco CallManager uses analog signaling protocols over analog trunks that connect voice gateways to analog endpoints and devices . Cisco CallManager supports these types of analog trunk interfaces:
Foreign Exchange Office (FXO) Analog trunks that connect a gateway to a central office (CO) or private branch exchange (PBX).
Foreign Exchange Station (FXS) Analog trunks that connect a gateway to plain old telephone service (POTS) device such as analog phones, fax machines, and legacy voice-mail systems.
From this good doc;
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00801ec5cc.html#1134121
FXS and FXO Interfaces
An FXS interface connects the router or access server to end-user equipment such as telephones, fax machines, or modems. The FXS interface supplies ring, voltage, and dial tone to the station and includes an RJ-11 connector for basic telephone equipment, keysets, and PBXs.
An FXO interface is used for trunk, or tie line, connections to a PSTN CO or to a PBX that does not support E&M signaling (when local telecommunications authority permits). This interface is of value for off-premise station applications. A standard RJ-11 modular telephone cable connects the FXO voice interface card to the PSTN or PBX through a telephone wall outlet.
FXO and FXS interfaces indicate on-hook or off-hook status and the seizure of telephone lines by one of two access signaling methods: loop start or ground start. The type of access signaling is determined by the type of service from the CO; standard home telephone lines use loop start, but business telephones can order ground start lines instead.
Loop-start is the more common of the access signaling techniques. When a handset is picked up (the telephone goes off-hook), this action closes the circuit that draws current from the telephone company CO and indicates a change in status, which signals the CO to provide dial tone. An incoming call is signaled from the CO to the handset by sending a signal in a standard on/off pattern, which causes the telephone to ring.
Loop-start has two disadvantages, however, that usually are not a problem on residential telephones but that become significant with the higher call volume experienced on business telephones. Loop-start signaling has no means of preventing two sides from seizing the same line simultaneously, a condition known as glare. Also, loop start signaling does not provide switch-side disconnect supervision for FXO calls. The telephony switch (the connection in the PSTN, another PBX, or key system) expects the router's FXO interface, which looks like a telephone to the switch, to hang up the calls it receives through its FXO port. However, this function is not built into the router for received calls; it only operates for calls originating from the FXO port.
Another access signaling method used by FXO and FXS interfaces to indicate on-hook or off-hook status to the CO is ground start signaling. It works by using ground and current detectors that allow the network to indicate off-hook or seizure of an incoming call independent of the ringing signal and allow for positive recognition of connects and disconnects.
From this very descriptive doc;
http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_configuration_guide_chapter09186a0080080afd.html
Hope this helps!
Rob

How to connect keysystem/pbx

hi guys i m new to voip and just started my cvoice, i have a confusion plz help me out
1) if i want to connect a pbx/keysystem with a CO switch over analog lines then wat type of interfaces will be used at both end ?? i m asking this coz in cvoice book its written that if u want to connect keysystem to CO switch thru analog trunk lines the signalling used will be winkstart and all does that mean that the cards will also be E&m trunk cards ??? am i right in this ??
2) wat are the possible ways of connecting pbx/keysystem with a CO switch ( i mean physical interfaces and the signalling )
plz tell me in a bit detail coz i m very much confused
thanks in advance

hi sir i m needing this for cvoice, see i have completed my first chapter but still not able to understand the analog connections b/w pbx and co switch, i m narrowing my question, can u plz tell me that if i want to connect a pbx with a co switch what are the possible ways of doing it ?? like wat interfaces will be used at both end ? wat signalling ? somewhere its written that groundstart signalling is used and somehere its written that i have to use E&M interfaces so wat shall i conclude, i really dont have any other book besides cvoice and voip fundamentals, if u could plz refer me a site or answer me anyhow i will really be grateful
thanks in advance

Cisco VS 3com in voice solution

what is the main difference between 3com and Cisco in ip telephony solution?

Tell us more about your network, goals, and requirements. How many phones? How many sites? How many voice mail boxes will you need? Do you have a separate voice mail system already that you'd like to keep?
What sort of connections to your voice provider do you require, e.g. PRI, analog trunks, etc.? Do you need to connect to PBXs from other vendors?
What Cisco solution was suggested, CallManager or CallManager Express?
We need more information in order to make an informed recommendation.

UC320 questions

Hello,
I have installed 2 UC320s. On my 2nd install the customer wants to be able to use an external transfer on the analog trunk. This requires a hook flash. I have not been able to figure out how to do this.
3 analog trunks
uc320 in blend mode
5 spa 504G sets
I have tried with direct line appearances and with no line appearances (dial 9 to access trunks). The spa 504 does not show a hook/flash button or key anywhere that I can find. Is the UC320 able to produce a hook/flash ?
also, is it possible for the auto attendant to produce a hook/flash.. for instance, customer wants after hours calls to transfer to an on call techs cell phone, can the UC320 produce a hook/flash in the instance to access the central office external transfer function on the analog trunk ? This is to not tie up 2 lines, but access the external transfer feature on the analog trunk itself
The customer has a 4th line (fax line) they would like this line to be accessible from the UC320 to make a 4th call only if the other 3 lines are busy. How do I setup the UC320 to call out on line 4 only when the other 3 lines are busy? I can not make 4 line appearances as the spa504 sets require one intercom set button.
Also, is it possible to have line appearances, and still have those line accessible from the dial out group when 9 is pressed?
Also, is it possible to break up the dial plan, for instance, have dial 8 go to analog trunks 1 or 2 and have dial 9 go to analog trunks 3 and 4 ?
I like the simplicity of the interface, however so far, it looks like the system is too dumbed down... please make an advanced tab or something that allows for dial plan manipulation (trunk call out order ect...), set VLANS, DSCP marks, cos etc... and other higher level functions. This thing works great for basic vannila stuff but if you need to do anything else good luck.
How do you check which voip encoding is being used, g.711 or g.729... can you force the phones to use one of the other? Does it support T.38... g.729 with VAD ?
Does the UC320 support Central office voice mail idication lamp on the analog trunks ? How about call waiting on the analog trunks... how do you do a hook/flash (redundant question, I know im just sort of ranting at this point)
Also, how do you create call restrictions.. for example, limit a lobby phone from dialing long distance, 900 #s, but still allow calling cards and 800 numbers ?
Any help would be great,
CTL Jake

Jake,
Let me categorize your questions and answer them
1. Hook Flash Support: Currently the UC320 / SPA Phone do not have any hook flash support or Centrex feature integration. Centrex capability is likely in the next release at the end of this year, however hook flash would still not be supported in the near term.
2. Fax Line: You could put a splitter on the 4th line and send one end to Fax and the other end to UC320. In the PBX mode ( 9+number) the 4th line would be selected last for outbound calls. On Shared FXOs offcourse it is user choice. They would need to be told not to use the Fax Line.
3. Simplicity: Yes this is intentional for a High Volume product. If you like to tweak things we would recommend proposing the UC500.
4. Codec: Yes preffered codec can be chose on the SIP trunk. T38 is supported.
5. MWI form CO: Again unsupported Centrex feature. In Key system mode you should be able to hear stutter dial tone.
6. Call Restriction: It is likely that this feature will be in our next major release at the end of this year.

Reg:help on E&m config

Dear experts,
My setup is like this
router with E&M (2800)----ip cloud----router with fxs--analog phone.
now i want to dial a call through e/m it should land on fxs
and from fxs to call land on pbx(e/m)
so plz help me how can i configure the routers.
Any body can help i am very thankfull to him,and also iam attaching the diag also.
Tha ks&Regards
srini

2600 router with FXS:
dial-peer voice 1 pots
destination pattren 101
port 1/0/0
dial-peer voice 2 pots
destination pattren 102
port 1/0/1
dial-peer voice 3 voip
destination-pattren 2..
session-target ipv4:192.168.1.1
2800 router connected to pbx:
Configuring an E&M analog trunk is straightforward. Three key options have to be set:
ô The signaling E&M signaling type
ô Two- or four-wire operation
ô The E&M type
Both sides of the trunk need to have a matching configuration. This example configuration
shows an E&M trunk using wink-start signaling, E&M type 5, and four-wire operation.
Because E&M supports inbound and outbound DNIS, DID is also configured on the
corresponding dial peer:
voice-port 0/2/0
signal wink-start ! default
operation 4-wire
type 5
voice-port 0/2/1
signal wink-start
operation 4-wire
type 5
so make sure u have the config in the router similer to the pbx E&M .. OK
then make the following dialpeers
dial-peer voice 9 voip
destination-pattren 1..
session target ipv4:192.168.1.2
dial-peer voice 10 pots
destination-pattern 2..
direct-inward-dial
port 0/2/0
forward-digits all
dial-peer voice 11 pots
destination-pattern 2..
preference 2
direct-inward-dial
port 0/2/1
forward-digits all
prefrence word mean if port 0/2/0 no available the dial-peer 11 over port 0/2/1 will takover
Specify how many wires are used for voice transmission using the operation {2-wire | 4-wire} command. Note that this defines the wires used for the audio path only, not for signaling. The default is two-wire operation.
Specify the E&M interface type to which this port is connected using the type {1 | 2 | 3 | 5} command. On the VIC-2E/M or VIC2-2E/M interface module, you must set both ports to the same interface type
If the port configuration is correct and you are still having problems placing or receiving calls on the trunk, the first place to look is at the wiring between the PBX and the voice gateway. As mentioned, the most common problem encountered with E&M trunk is incorrect wiring
good luck
If helpfule Rate

PBX interoperatibility with router

Dear ALl
I have a confusion regarding the PBX integrations with Router.
Now if i am integrating my 2801 router with any PBX trunk ports (by trunk ports i mean the ports used to connect with PSTN with normal RJ-11 connector) , do i really have to worry about the compatibility between router and PBX.
The reason is that if the PBX can terminate the lines from the PSTN (RJ-11) than it should work with my router too .It is with e1 when we should worry about compatibility .Am i correct.
So what ports do i need on the router side?FXS.i thnok so as when we connect pstn with PBX than PSTN provides the tone ,so now my router should provide the tone to PBX trunks.
My scenario is that eight lines from PSTN are terminated in the norstar PBX and as i am proposing the IPT system the lines will now be terminated into the router and than the router should forward the call to the PBX if it is for any analog phone .
So my real question is do i really need to worry about the PBX interoperability when i am dealing with analog trunks.
Backbone

Hi,
the short answer is if even FXO is barely acceptable to connect to the PSTN (you should really get at least ISDN BRI), FXS/FXO connnection to the PBX will be certainly too limiting. Many problems of double dialing, potential stuck ports, no passing of information, etc. etc. The very miniumum is E&M to the PBX but real good would be again, ISDN BRI or PRI.
If the PBX is missing the necessary cards and is not worth to buy them, the you should at some point look at replacing the PBX completely.
The answer to real question is yes, you should really worry about analog trunks if you want the integration be of a professional level.
Hope this helps, please rate post if it does!

Callmanager 5.04 basic configuration

Does anyone have a basic "step through" guide on the required steps to get a CCM 5.0 & 2801 gateway setup to make simple call via analog trunk? I need to setup a quick demo...

http://www.firewall.cx/cisco-technical-knowledgebase/cisco-voice/369-cisco-ccme-gui-part-1.html

Lab setup- 2911/CME, aironet AP, and 7926

I need to setup a lab for testing / demo 2 7926 phones to call each other.
Do you have a simple sample CME config to accomplish this (ie, ephone command, realtive show commands to prove it connected or not)?
Can I plug my AP directly into the ethernet port on the 2911? If not will any switch do?
Also, there will be a VMXL server added later, where does this get connected physically?
Let me know on this sample config what else I would need to setup the phones. Thanks so much.

You have to plug your AP to PoE switch port configured as trunk port (assuming multiple vlan trunk to wireless network) if you are using it standalone mode (without WLC). If your switch is not POE then you need power injector to power the AP.
Then you have to configure necessary SSIDs to phones to join via wirelessly & register to your CME. In DHCP scope for the phones you need to provide option 150 which is CME IP address.
Do not know about VMXL server connetion.
HTH
Rasika

Convert analog DID trunk to a PRI?

Hi, I'm trying to figure out the least expensive way of converting an analog DID trunk coming from the phone company to a PRI that I can feed into our ISDN switch. I realize that the analog DID trunk only equates out to one B-channel + one D-channel for signaling but that is fine, the other 22 channels can go unused/unmapped. I'm not too familiar with Cisco gear but figured maybe two MC3810s back to back or a 2610 with the proper add-ons (VIC-2DID + ?). I also need to make sure that the dialed phone number is preserved within the signaling (DID->D-channel). Any low end hardware suggestions would be greatly appreciated. Thanks for any info, Brian.

When receiving an inbound call from a plain old telephone service (POTS) interface, the DID feature in dial peers enables the router/gateway to use the called number (DNIS) to directly match an outbound dial peer. When DID is configured on the inbound POTS dial peer, the called number is automatically used to match the destination pattern for the outbound call leg.
To configure a POTS dial peer for DID, enter the following Cisco IOS commands beginning in global configuration mode:
Router(config)#dial-peer voice number pots
Router(config-dial-peer)#direct-inward-dial
This URL should help you:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00801142f8.shtml

20 Analog Trunks on CME 2911

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