MGCP FXS ports requires a license in CUCM9

Hello!
I am connecting Some analoge phones to VG350 FXS ports which is configured as a MGCP Gateway in CUCM. I beleive MGCP did not requires any license for it. Can some confirm this ? is there any Cisco doc on it ?
Thanks & Regards,

Hi Sambit,
Technically u are correct but legally I think u would be requiring license.
even the ordering guide says Analog devices are supported with Essential USer license and must be purchased through UCL.
http://www.cisco.com/web/partners/downloads/partner/WWChannels/technology/ipc/downloads/finalcopy.pdf
regds,
aman

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Adding MGCP FXS Ports to H323 Gateways

Currently all of our Gateways are H.323 gateways. Due to a business requirement we are now going to be enforcing our users to use forced authorization codes to place LD calls. In order to facilitate this on our analog phones it seems the only option is to use MGCP gateways.
From what I understand we can run multiple signalling protocols on voice gateways. We have a variety of gateway models but by and large most of these gateways are VG224 models. I think what I would like to do is keep the current h.323 dial-peer and voice-port settings for the PLAR emergency phones that we have on these gateways and only change the analog phones to MGCP.
Most of the route patterns to these h323 gateways look like this... 102[0-5] and then the dial peers on the individual gateways route to the appropriate voice port like this...
dial-peer voice 1020 pots
huntstop
destination-pattern 1002
port 2/21
The Voice port config looks like this...
voice-port 2/21
timeouts interdigit 7
description tie pr 1520
station-id name PTRM 1020
station-id number 1020
caller-id enable
My plan is to create the MGCP Gateways in CUCM as wells as the DN's... in this example x1020. I will then enable MGCP on the gateways. After that my assumption is that I can individually remove the Voice-port and dial-peer configurations and then add the MGCP dial peers with the port and "service MGCPAPP" commands.
My other option is to redo the entire gateway at the same time and schedule after-hours down-times to make the change. I want to avoid this if possible as we have 40+ gateways that need to be changed.
Basically I just need some guidance or confirmation if my plan will work or if there is a better way to do this? Are there any caveats or known issues I should look out for when running multiple signalling protocols on the same gateway?
Thanks,
Trav Moore

Thanks Aaron,
I was wondering about the MGCP ccm-config command but was worried it would re-write the entire h.323 gateway to MGCP. Good to know that it won't and that this is a potential option.
I actually do prefer the idea of only having one signalling protocol (I would like to go all SIP if not for the FAC codes needed). Unfortunately any maintenance that I do that impacts end-users requires a lot of after-hours scheduling and maintenance alerts. These gateways have a combination of fax-machines, PLAR's (emergency phones and overhead paging), and analog phones. Maybe eventually I can migrate all of these ports to MGCP. For now the analog phones are the only ones that must be converted and if I can quickly convert them without anyone noticing aside from the minimal reset in CUCM then this would be ideal.
Thanks!

CFwd on FXS-Ports (mgcp)

hi!
How can I forward the Calls on an Phone witch is connected to an FXS-Port (mgcp, vg200)?
How can I use the other Features, like PickUp etc.
vy Markus

Since there has been no response to your post, it appears to be either too complex or too rare an issue for other forum members to assist you. If you don't get a suitable response to your post, you may wish to review our resources at the online Technical Assistance Center (http://www.cisco.com/tac) or speak with a TAC engineer. You can open a TAC case online at http://www.cisco.com/tac/caseopen
If anyone else in the forum has some advice, please reply to this thread.
Thank you for posting.

Installing an analog polycom soundstation 2 on FXS port in CUCME

I apologize if this is a stupid question, I'm an Avaya voice (cisco data) guy, I'm still learning Cisco voice.
I've installed an analog polycom soundstation 2, I can make internal and external calls. However I can only receive one incoming call at at time (second call receives a busy signal) and I can't conference a second call.
From researching I think I need to change the FXS port from MGCP to SCCP (I have the license for it) but I'm not 100% sure that's correct and if it is I'm not sure how to do it.
Any advice would be much appreciated.

This should give you an idea where to start
http://www.icciev.com/1/post/2011/09/adding-vg224-to-cucm-80-as-sccp-or-mgcp-gateway-differences-and-configurations-part-2.html
Jorge Armijo
Please remember to rate helpful responses and identify helpful or correct answers.

CLID not shown on FXS port

I am using Call-manager 6.0.1b, a MGCP controlled Gateway.On the Gateway i have installed a NM-HD-2V with a Vic2-2FXS module. On this module i connect 2 analog phone with capability to display the Caller-ID. When i call the analog port from a ip phone or from the other analog phone the Dn is not shown. When i connect the phone directly to the PSTN and dial this nr via my cell phone the nr is shown so i expect that the nr format received is not correct. How can this performed that the correct format is shown to the FXS port connected phone ?

Hi,
Yes - you are correct. Looked at this one too quickly.
You will want to make sure the CPTONE defined on the port is for the country the phone is manufactured for, and that the voice-pport has the 'caller-id enable' command.
If those are both correct, and I'm guessing that they are since some caller-id works, then you need to inspect the gateways that take the calls to begin with.
Do you have caller-id trouble for internal calls also?
How do these calls come into your network?
hth,
nick

FXS Ports & Pickup Groups

Is there a way to make an analog phone connected to an FXS port a part of a call pickup group that contains both analog phones & IP phones? I setup a lab and used MGCP to add the gateway and I was able to add the DN associated with the FXS port to a call pickup group. However, I am unable to figure out how to answer the call from the analog phone when another IP phone in the call pickup group is ringing.
Thanks in advance

Hi
You are going down the right track with this.
http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Unified%20Communications%20and%20Video&topic=IP%20Telephony&CommCmd=MB%3Fcmd%3Dpass_through%26location%3Doutline%40%5E1%40%40.1dde5372/0#selected_message
See this other post I made (for a different purpose, but the principal is the same - it just opens up features available to IP phones for FXS ports by registering them using SCCP).
Regards
Aaron
Please rate helpful posts...

CLID presentation fails on FXS port

i have a mgcp gateway with a nm-hd-2v , and the modules vic2-2fxo ,vic2-2fxs
The CLID of the external call is presented normally on my fxo port and displayed on the IP (soft-)phones.
I dont see the nr getting presented on the FXS port when i debug.
Called number and calling nr remains empty.
I see a calling nr when i do a csim start <exention>
caller-id enable is configured on the fxs port configuration at the gateway.
tried several type of caller-id alerting methodes without success.
When connecting the same phone on the pstn, i get the clid normally so it is something in the configuration
regards

Hi,
Yes - you are correct. Looked at this one too quickly.
You will want to make sure the CPTONE defined on the port is for the country the phone is manufactured for, and that the voice-pport has the 'caller-id enable' command.
If those are both correct, and I'm guessing that they are since some caller-id works, then you need to inspect the gateways that take the calls to begin with.
Do you have caller-id trouble for internal calls also?
How do these calls come into your network?
hth,
nick

Cisco FXS Port - RJ11 to RJ45

Hi,
I have a VIC3-2FXS/DID in Switerzland and my customer has asked for it to be connected to a fax machine through Strucuted cabling and they would like to know what type of cables they required.
They want RJ11 (FXS Port) to RJ45 (Patch Panel) then RJ45 (Patch Panel) to RJ11(Fax Machine).
Could someone explain the pins out of the cables, as it is going to be two RJ11 to RJ45 cables do they need to rollover?
Thanks for any help.

All cabling on Cisco voice cards is straight through from the port to the device. RJ11 has 2 pins, for tip and ring. That's blue and white/blue. RJ45 has 8 pins, but if you want to stick with the 568B wiring standard on the RJ45 jacks,you would use this.
Blue: RJ11(pin1)----RJ45(pin 4)--------(pin4)RJ45-----(pin 1)RJ11
White/Blue: RJ11(pin2)----RJ45(pin 5)--------(pin 5)RJ45-----(pin 2)RJ11
If you actually have RJ14 jacks (4 pins), its pins 2 and 3 (the middle pins) which are used. If it is RJ25, it's still the middle pins, which would be 3 and 4.

FXS Port stucks in Off-Hook state on NM-HDA

Hi,
i have some trouble with an FXS Port on an NM-HDA which stucks in the Off-Hook state, after some time of normal operation. The Voice-Port is in Off-Hook state regardless of if there is an Enddevice attached or not. Even an reload of the router does not help.
The voice port is configured for loop-start. The NM-HDA is housed in an Cisco 3725 running IOS Vers. 12.3(14)T2 with H.323 signalling.
Is there anybody who has seen this before? Could this be caused by an IOS-Bug or is it more likley that the NM-HDA is faulty?
Regards
Robert

Here's the bug ID: CSCse15025
Doubt the NM is faulty - we had the same issue in a 2821 and migrated to 12.4(4)T4 to correct it. No hardware replacement required. All has worked fine since then.
HTH
Tom

MGCP FAX port not receiving incoming faxes

Hi Gurus'
I have a client and they have set up an FXS port with MGCP for their fax machine on VG 224 gateway. T 38 fax relay is enabled on the gateway page.
They can send the fax to any number on PSTN and if a phone is attached to that port it can make and receive calls to and from PSTN. The fax machine cannot receive the incoming faxes but if you call that number from PSTN you can hear the fax tone. They have a SIP provider for the PSTN calls.
Any thoughts on this would be helpful.
Thanks,
Asad Hanif

hello - I have moved your conversation from an obscure community to a more trafficked one, hopefully one of our experts will pick your discussion up to answer.

Ports required for voice gateway registration

Hi,
Currently our remote office voice gateway is trying to register to the CM and in between there is a firewall. We have opened port DNS, NTP, 2427 and 2428 but it still showing registering to the call manager. What other ports shall we open to make it works?
What about the port requirement for CUE?
Thanks.

For MGCP:
DNS
NTP
UDP 2427
TCP 2428
TFTP (UDP 69)
For CUE, here is a link you may find helpful:
http://www.cisco.com/en/US/partner/netsol/ns340/ns394/ns165/ns391/networking_solutions_design_guidance09186a00801f8e31.html#wp41149
hth,
nick

Port required for Veritas cluster implementation

hello there ,
i need to know what are the port required for veritas cluster implementation on Sun Messaging Server 6.2 . anybody care to help me on this ?
thanks

> We are planning a 2 node Oracle 9i RAC cluster on Sun
Cluster 3.Good. This is a popular configuration.
Can you please explain these 2 questions?
1)
If we have a hardware disk array RAID controller with
LUNs etc, then why do we need to have Veritas Volume
Manager (VxVM) if all the LUNS are configured at a
hardware level?VxVM is not required to run RAC. VxVM has an option (separately
licensable) which is specifically designed for OPS/RAC. But if
you have a highly reliable, multi-pathed, hardware RAID platform,
you are not required to have VxVM.
2)
Do we need to have VxFS? All our Oracle database
files will be on raw partitions.No.
IMHO, simplify is a good philosophy. Adding more software
and layers into a highly available design will tend to reduce
the availability. So, if you are going for maximum availabiliity,
you will want to avoid over-complicating the design. KISS.
In the case of RAC, or Oracle in general, many people do use
raw and Oracle has the ability to manage data in raw devices
pretty well. Oracle 10g further improves along these lines.
A tenet in the design of highly available systems is to keep
the data management as close to the application as possible.
Oracle, and especially 10g, are following this tenet. The only
danger here is that they could try to get too clever, and end up
following policies which are suboptimal as the underlying
technologies change. But even in this case, the policy is
coming from the application rather than the supporting platform.
-- richard

SPA2102 FXS port state monitoring

Hi all,
Is there any way to get FXS port state for LinkSys SPA2102 VoIP adapter (i.e. is there a phone that connected to it or not) remotely, for example, via SNMP ?
Thanks.

Here's the bug ID: CSCse15025
Doubt the NM is faulty - we had the same issue in a 2821 and migrated to 12.4(4)T4 to correct it. No hardware replacement required. All has worked fine since then.
HTH
Tom

Call Manager register fxs port with voice gateway- problem

I have a CUCM 6 and a Voice Gateway V224. I've configured the voice gateway's voice FXS ports as MGCP.
I have a Voip connected and registered to the CUCM and a Pots phone connected to the Voice Gateway.
If i dial from the Voip to the Pots phone it rings. The problem is that i cannot ring from the Pots to the Voip phone.
I have no dial tone.
If i write no shut down on the voice port i have a tone. If i configure mgcp on the voice port i have a busy ringtone.
I've entered no mgcp and mgcp commands and i've reset the voice gateway.
How can i call from the pots to the voip phone?
The ios version on the voice gateway is Version 12.4(22)T4.
Here is an outghtput from the Voice gateway.
ccm-manager mgcp
ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 10.1.1.33
ccm-manager config
mgcp
mgcp call-agent CCMIOSS 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp validate domain-name
mgcp rtp payload-type g726r16 static
mgcp profile default
timeout tone busy 600
timeout tone dial 600
dial-peer voice 999223 pots
service mgcpapp
port 2/23
dial-peer voice 999222 pots
service mgcpapp
port 2/22
dial-peer voice 999888 pots
service mgcpapp
port 2/23
The CUCM 6 is registered with the voice gateway.

Is your campaign using CPA? If so, what's the behavior if CPA is not enabled?
I think the best thing to do is to run a trace...
Call Manager > Cisco Unified Serviceability > Trace > Configurations
Select a CUCM server - any subscriber would work.
Service Group - CM Services
Cisco CallManager (Inactive)
Enable SIP Stack Trace and apply to all nodes. Download and install RTMT
Make a bunch of outbound tests and then open RTMT > Trace & Log Central > Collect Files > Check "All Servers" for Cisco CallManager > Next > Next > Relative Range if you made the test calls within the last X minutes, otherwise you can set a From and To datetime. Click Finish and go through your SDL logs and see what errors you find and post them here.
Also, make sure your phone is in the correct CSS in Call Manager

What are the ports required for the Audio, Video and A/V conferencing when the following end points are enabled for QoS in Lync 2013 server?

Hi All,
What are the ports required for the Audio, Video and A/V conferencing when the following clients are enabled for QoS in Lync 2013 server?
Client Type
Port range and Protocol required for Audio
Port range and Protocol required for
Video
Port range and Protocol required for
A/Vconferencing
Windows Desktop Client
Windows mobile App
Iphone
Ipad
Andriod phone
Andriod Tablet
MAC desktop client
Please advise. Many Thanks.

Out of the box, 1024-65535 for all of the client ports. :)
https://technet.microsoft.com/en-us/library/gg398833.aspx
You'll want to tune your client ports a bit
https://technet.microsoft.com/en-us/library/jj204760.aspx as seen here, and then the client ports would use those ranges which is easier to set QoS markings. I'm not sure the mobile clients respect that setting.
Elan's got the best writeup for Windows clients here:
http://www.shudnow.net/2013/02/16/enabling-qos-for-lync-server-2013-and-various-clients-part-1/
However, the marking of the packets is the tricky part. Windows can do it via Group Policy, but for the other clients you'll need to have the network specifically prioritize ports regardless of DSCP markings. You have to do it based on ports
as the traffic could be peer to peer.
Please remember, if you see a post that helped you please click "Vote As Helpful" and if it answered your question please click "Mark As Answer".
SWC Unified Communications
This forum post is based upon my personal experience and does not necessarily reflect the opinion or view of Microsoft, its employees, or other MVPs.

MGCP FXS ports requires a license in CUCM9

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