Cisco VG 224 Analog Phone Gateway

Hi
Can i use cisco vg224 gateway in a scenario like -
cisco ISR+CCM express -->WAN LINK --> cisco ISR -> VG224 gateway --> analog phones.
If this scenario is good then, can i do call transfer and forwarding with analog phones with any extensions digital/analog on that setup?
thanks
regards
Rakesh
=====

The Cisco VG 224 is an analog voice gateway that manages up to 24 FXS ports for analog devices. Call control of the analog FXS ports is provided by Cisco CallManager or by a Cisco CallManager Express (Cisco CME) system. An SCCP telephony control (STC) application on the Cisco VG 224 functions as a a proxy to translate call-control messages between the call-control system (Cisco CallManager or Cisco CME) and the voice gateway.
This URL should help you:
http://www.cisco.com/en/US/products/ps6441/products_feature_guide09186a0080483a76.html
http://www.cisco.com/en/US/products/ps6441/products_feature_guide09186a0080483a76.html

Similar Messages

Pick up group between analog Phone connected on VG224 with Cisco 7911 does not work?

Hey guys,
I had a problem, during a demo. My team was design a cisco callmanager 7.0 with 7911 IP Phone and with VG224 analog devices phones. I would like to know if there are some a problem of capability when configured a Analog Phone on VG224 in the same pickup group with Cisco IP Phone 7911.
I got pickup between Analog phones but with I cannot pick up, between IP Phones and analog phones.
Does anybody knows if has capability problem or could be a configuration problem.
The feature has already enbled for pick up **3 (stcapp feature access-code).
best regards
Daniel

Hi Daniel,
Did you try **4
The default settings for the VG224 feature codes are as follows:
Call Forward All (CFA) **1
Call Forward All Cancel **2
Call Park Directed Call Pickup Directed **6
Call Pickup Group **4
Call Pickup Local **3
Call Transfer = Hookflash
Redial *#
Speed Dial *01 to *99 for two-digit codes
Speed Dial to Voice Mail Default prefix and code is *0 for one-digit codes, and *00 for two-digit codes
http://www.cisco.com/en/US/docs/ios/12_4t/12_4t2/ht1vg224.html#wp1178205
Hope this helps!
Rob

Analog phone with ring light

We have analog phones in certain applications that use a neon light instead of a ring tone. It requires 90v to activate the neon light. We have VG224 analog gateways which appear to not be able to push 90v. Does anyone have a solution or is there another Cisco gateway that will support analog phones that use a neon light instead of a ring?

BlackBerrys have a flashing red LED indicator light for new messages. It's hard to miss. And with the sleeve cases, there is a gap for keeping this notification light visible, even when holstered.

SPA9000 + SPA492 + analogic phone: Cancel Request doesn't work

Hi all, I'm build up a LVS in the following configuration: SPA9000 + SPA942 + analogic phone on FXS1; PBX connected to LAN with WAN interface and IP phones on the same LAN at all; "Force Media Proxy" parameter = "yes"; "Line 1" in trunk with an ITSP; no NAT parameter changed from default; WAG54GP2 on ADSL as Internet gateway in standard configuration. All seems running fine, except for one small but really fundamental function: I start an outbound call from SPA942 or analogic phone, the addressed phone rings and if I hung up before the call is accepted, the phone on the other side keeps the ringing up. In my opinion, it seems like a failed "Cancel" request. If I try to do the same with the analogic phone conected to WAG54GP2 all runs well, so I can exclude a signalling problem of the ITSP. Thanks in advance for your help.

Several things to try -
1) Set NAT mapping and NAT Keep Alive to yes on the SPA9000 line connected to the ITSP.
2) Setup port forwarding on the WAG54GP2 to forward port 5060 to the IP address of your SPA9000.
3) Replace the WAG54GP2 with a standard modem and connect the SPA9000 directly to it.

H.323 endpoint to analog phone call problem

I have Tandberg H.323 endpoints registered in gatekeeper, Callmanager 4.2(1) and MGCP voice gateway to classic telephony. There is a problem, when I try to set up a call from H.323 endpoint to the analog phone through voice gateway. The call is set up but disconnets after about 3 seconds alerting signal. However you can have normal connection, when you answer the phone during theese three seconds of alerting. When it disconnects it says: 'Resource unavailable, unsecified'.
What can cause such problem?
Thanks in advance.

NO, I was using an Sx-20 as the H323 endpoint. I tried a different video confernece unit.The other confernce unit worked with out any issues. Sorry I wish I new what was wrong with the sx-20. What kind of h323 endpoint are you using?

MWI on analog phone using stutter dial on VG248

Someone is telling me that you can use stutter dial or some other method to light up an anolog phone Message waiting indicator using on a VG248. This is with an analog phone that has an Message light on it. Is this possible and if so is it possible using and ATA with Call Manager?

Hi Shane,
This does work with the VG248, we have used the "Stutter Dial Tone" on some analog phones and used the actual MWI Lamp on other analog phones with good success. Have a look;
Choosing Message Waiting Indicator Type
The VG248 supports several types of methods for sending MWI messages to analog phones. Because you might have different types of analog phones connected to the VG248, you can modify the MWI type on a per-port basis. So, if you have some analog phones that have MWI lamps on them, you can notify users of awaiting messages using the lamp. Or, you can choose to play a tone when users pick up their phones.
Keep in mind that the VG248 only sends this information to the phones if it is received from Cisco CallManager. If Cisco CallManager is not integrated with your voice mail system, it does not send this information to the VG248.
Step 1 From the main menu, choose Configure.
Step 2 Choose Telephony.
Step 3 Choose Port specific parameters.
Step 4 Use the arrow keys to select the port to configure and press Enter.
Step 5 Choose MWI type.
Step 6 Choose from the following options:
Lampilluminates lamp on phone
Caller IDuses caller ID mechanism to send MWI messages to the LCD screen on phone
Stutterplays tones when user picks up the phone
Lamp + stutterilluminates lamp and plays tone
Caller ID + stuttersends message to LCD screen and plays tone
Nonedoes not send MWI information
From this doc;
http://www.cisco.com/en/US/products/hw/gatecont/ps2250/products_configuration_guide_chapter09186a0080087de4.html#xtocid13
Hope this helps!
Rob
Please remember to rate helpful posts..........

Setup Hotline Analog Phone with SCCP

Anyone out there can help me out how to setup a hotline in CUCM 8.6?
I've followed this procedure below, but I'm still getting an error message when I lift the handset such as "your dial cannot be completed...".
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a0080232b9f.shtml
Just in a nutshell, here is what I've done:
Extension: 1111 PLAR Analog SCCP Phone with: CSS_Hotdial and PT_Allowed (line settings)
TP: blank, PT_Hotdial and CSS_Hotdial, and called PTM: 2222
Extension: 2222 CSS_Allowed, and PT_Allowed - no changes on this extension...
Added the PT_Allowed into CSS_Hotdial. The order is: PT_Hotdial/PT_Allowed
Both lines are on the same VG204-vg20x-ipvoice-mz.124-22.T5.bin. When I selected hotline feature it says is supported Analog Phone SCCP, but I did not see the VG. It looks like is not supported hotline on VGs. Can someone confirm that?
Am I missing something? Do I need to enable something under Service/Enterprise parameters?
Thank you!

Chris, thanks for your prompt help.
I thought the TP needs a CSS with access to the 1111 DN, not to the 2222. Because when someone lifts the handset on phone 1111 this phone should invoke the TP which translate and ring to the 2222 phone. Thus, I put this CSS_Hotdial on phone 1111 under line.
And my TP looks like this: PT_Hotdial and CSS_Hotdial, and called party transformation mask 2222
The CSS_Hotdial has only access to one PT_Hotdial.
Also the two phones 1111 and 2222 are on the same partition PT_Allowed, but they are on different CSS.
Anyway I also tried what you said but did not work. Are you sure this hotline works on VG204? I think it is not working because this feature is not supported on VG204 nor in any VG model.
If you are sure it should work. Could you please elaborate more this solution with PT/CSS/TP?
Once again thanks

Re: CME and Analog Phones

I have CME running on a Cisco 2801 router and I have installed a 4FXS card. I have an analog phone connected to the port 0/2/0 of the FXS card and the following is configured.
dial-peer voice 201 pots
destination-pattern 4015
port 0/2/0
I am able to initiate calls from this unit however calls to this DN (analog) phone I receive a fast busy.
Also how do I transfer a call from an analog phone in this setup?

This is the feature explained:
http://www.cisco.com/en/US/docs/ios/12_4t/12_4t2/ht1vg224.html
After the basic is working, you can configure a fac to access various functions:
http://www.cisco.com/en/US/docs/ios/12_4/12_4x/12_4_6xe/htfeatmd.html

What solution for implement voicemail for analog phone?

if want to add voicemail for analog phone via VG224, how to notify voicemail to analog phone?
we knew there is a light flash on IP phone when any voicemail on waiting, but how to do it to analog phone?
I fround one document about this, but don't know how to do it.
http://www.cisco.com/en/US/docs/ios/12_4t/12_4t2/ht1ccmqs.html#wp34936

It depends if you phones have a MWI lamp, or just rely on stuttered dialtone. Check:
http://www.cisco.com/en/US/docs/ios/voice/sip/configuration/guide/sip_cg-mwi_support_TSD_Island_of_Content_Chapter.html#wp1069088

CCM MGCP configuration for an analog phone

Hi all,
I am in trouble with between CCM and FXS port. When I pick an analog phone up on FXS port, It always sounds busy-tone, so I can not dial everytimes. Strangely it is possible to call from ip phone to analog phone, but not from the analog phone to IP phone. Of course, I made sure that both of FXS port and phone were registered to CCM properly.
In this case, Do I need specific configuration for FXS on IOS gateway, based on 2600xx, or on CCM ?
can anybody else know about those kinds of situation or experienced people ?
My simple config is the following
-------- the following on 26xxXM IOS ---------
MGCP
mgcp
mgcp dtmf-relay voip codec all mode out-of-band
mgcp call-agent 192.168.10.10
mgcp sdp simple
ccm-manager mgcp
dial-peer voice 101 pots
application MGCPAPP
port 1/0/0
-------- the end --------------

Knew codec's and their usages.
But in an actual deploying environement haven't encountered (if there is any thing like guaranteed bandwidth; not much of as said by bevilacqua; but may be consumed by an ip phone to make its fresh registration), as very well we knew that the bandwidth 64k is enough to establish call using G729 WAN end. More over the 64k pipeline is used for voice and data.
Thanks for sharing the link. Went through the topic. It's useful.
Regds,
Karthik

Can't transfer to an analog phone on FXS port (CME2811)

I have six IP phones with locals 501 to 506. Additionally, I have four analog phones (locals 541 to 544) connected to a VIC2-4FXS card on a 2811 with CME and CUE module.
All phones can call each other. But if say, a user on an IP phone calls a 2nd user on another IP phone, the 2nd user cannot transfer to the 54x locals. She hears a busy or reorder tone (still need to verify which) after keying in the 2nd digit (ie. 54 then "busy" tone)
Am I missing something?

Under telephony-service configuration enter the transfer-pattern command. For example:
router(config)#telephony-service
router(config)#transfer-pattern 5..
If no transfer pattern is set, transfers are only permitted to other IP phones.
Hope this helps. If so, please rate the post.
Brandon

No ring on analog phone

Hi I have a router 1750 with an analog phone connected through an FXS port on the router. The voice works but when the extension number is dialed the phone does not ring. What can the problem be?
Regards.

What is your dial-peer configuration ?
If yo want to see how the router rings the phone, check "debug vpm signal" with "term mon".
Hope this helps, please rate post if it does!

Analog phones couldn't communicate with ip phones in an internal lan

As you can see in the above picture, i have a astrick server 1.6.2 and cucm 7.0, spa and isr router.
when im calling from ip phone to analog phone call is successful , but when Im calling from analog phones to ip phones call is not successful.
only one way communication is happenig in this scenerio i.e from ip phones to analog phones.but vice versa is not working why ?
can any one slove my problem i will glad and thanks for reading my post.

Kiran,
1. Who handles those analog phones ( CUCM or Astrick ) ?
2. Did you check the routing on the ISR ? Usually one way audio is experienced due to missing ip route. Please ensure you have the route in place for both directions.
3. What is the integration type between CUCM & Asterisk ?
GP.

Vg224 don't auto disconnect with analog phone

Dear all ,
why vg224 don't auto disconnect with analog phone? is the busy frequency dismatch between vg224 and analog phone?
can you have any suggest to me ?
thks.

HI rrjoas45,
thks your answer, I try to add a command with supervisory . but I discovery it isn't effective . the port status follow:
sh voice port 2/0
Foreign Exchange Station 2/0 Slot is 2, Port is 0
Type of VoicePort is FXS
Operation State is DORMANT
Administrative State is UP
The Last Interface Down Failure Cause is Administrative Shutdown
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 64 ms
Echo Cancel worst case ERL is set to 6 dB
Playout-delay Mode is set to adaptive
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 250 ms
Playout-delay Minimum mode is set to default, value 40 ms
Playout-delay Fax is set to 300 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call Disconnect Time Out is set to 60 s
Supervisory Disconnect Time Out is set to 750 ms
Ringing Time Out is set to 180 s
Wait Release Time Out is set to 30 s
Companding Type is u-law
Region Tone is set for CN
Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Analog interface A-D gain offset = -3.0 dB
Analog interface D-A gain offset = -3.0 dB
FXS idle voltage set to high
Ring DC offset set to 0 volt
Station name None, Station number None
Translation profile (Incoming):
Translation profile (Outgoing):
Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 25 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is active
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Hookflash-in Timing is set to max=1000 ms, min=150 ms
Hookflash-out Timing is set to 400 ms
No disconnect acknowledge
Ring Cadence is defined by CPTone Selection
Ring Cadence are [10 40] * 100 msec
Ringer Equivalence Number is set to 1
thks.

Equium A60-191: Modem issue - Only analog phone lines are supported

My Equium A60 was powered up when lightning struck nearby and caused the electricity supply to go off and back on.
The OS is XP Home and my ISP is AOL. When I try to access the Net by DUN, I receive the repeated error message:
"Connected phone line is not compatible with this modem. Only analog phone lines are supported."
My desktop computer, which also uses AOL, still connects to the same phone line without any problem.
I have rebooted the A60 several times; changed and tried out 2 phone connnection cables - both work; the phone line works fine; I have reinstalled the driver provided on this site; removed the tick from the box that says to wait for the dial phone; checked out that the modem is in working and on the Device Manager list. All of this is to no avail.
Troubleshooter is suggesting that COM 3 is not enabled. I cannot see COM 3 listed in Device Manager, under Ports.
I have tried Google and ASK with the error message. There are not many posts and it appears that the Toshiba modem is in fact an Agere. This makes seems to be the subject of some of the complaints.
Finally I have uninstalled the driver/modem, AOL and one other DUN connection. I rebooted and went through the process of installing "New Hardware" and AOL. Whilst the modem is enabled and allegedly working... I am still unable to connect to the Net. I reinstalled an additional 'pay as you go' ISP. When I try to dial out on that it generates the error 680. It does this even when the box is unchecked, for wait for a dial tone.
I would be most grateful if anybody could help. I have run out of ideas.
Many thanks.

Hi
You are right, the modem support the Agere Modem Chip.
You will not find the COM3 in the device manager because the modem uses not real com port. Its a virtual com port.
I think you should start the diagnostic tool test and check if the Modem has no malfunction.
Additional you should delete the connection in the Region Select Utility and after new reboot create a new one.
I found also the useful Microsoft article about the Troubleshooting network and dial-up connections.
http://www.microsoft.com/resources/documentation/windows/xp/all/proddocs/en-us/trouble_all.mspx?mfr=true
and here you will find info about the DUNS error 680:
http://www.modemsite.com/56k/duns680.asp
http://php.iupui.edu/~aamjohns/trouble.html#No Dial Tone

Cisco VG 224 Analog Phone Gateway

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