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
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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

Advice about FXS Port configuration for connecting to modems

Hello all,
We have not confirm the problem, but we are having reports that analog devices like USR V.92 Modems and Power meters are not connecting or dropping the connection. We have an ISR 2911 with the 4-port FXS-DID WIC (VIC3-4FXS/DID), configure to an internet SIP truck provider. The Router is running IOS 15.3(3)M2. If we connect an analog phone to the FXS port and call, voice is clear. So the question is what configuration settings should we have for FXS ports and/or dialpeers for connections to modems? Attached is the current configuration of the router.
Do we have to be worried about Codec? If so which Codec should we configure?
Any Physical FXS settings should be set?
Are my DSP setup correctly?
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Hi Amit,
No problem on the delay. Your help is very much appreciated.
You're losing me a bit. Because I don't know if your giving me dial-peer 1 as an example. 1 being just a random number. Or your telling me to change my current dial-peer 1 configuration. Which is currently a pots dial-peer:
dial-peer voice 1 pots
description POTS port 0/1/0
destination-pattern 111111111
port 0/1/0
dial-peer 1000 voip is our outbound call leg. Sorry call legs are not exactly clear to me yet. In fact they haven't been in years. :) So change my dial-peer to dial-peer voice 1 voip?
My other question still unknown to me is should we use:
fax protocol pass-through g711ulaw
Even though we are trying to use a modem and not a fax? Also this command is a dial-peer command and not a global command. So what global command should i be using?
Again thank you for your time and help,
Nick

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?
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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...

SPA9000 How do you direct SIP to FXS port?

How do you point a SIP account to the #1 FXS port on the SPA9000? I have a separate SIP account for my fax machine and need to point it to the analog FXS port.
Thanks

When you say “Line Tabs” do you mean where you configureLine 1-4? So if Line 1 is set for ITSP and Line 2 is the SPA400, I should useLine 3 for Fax? Then under ‘Line 3’ I should configure the User ID, Password,and Proxy like we did on the ITSP page except for the Fax SIP account?
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Hi,
We are running VoIPovFR to a remote location, which has 2 x 4FXS/DID module installed into a C2621XM chassis. We are using IOS c2600-is-mz.122-15.
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Try wiring all FXS ports like this
FXS PORT 0 RJ11 3-----------------------A ANALOGUE
FXS PORT 0 RJ11 4-----------------------B PHONE
FXS PORT 1 RJ11 3-----------------------A ANALOGUE
FXS PORT 1 RJ11 4-----------------------B PHONE
FXS PORT 2 RJ11 3-----------------------A ANALOGUE
FXS PORT 2 RJ11 4-----------------------B PHONE
FXS PORT 3 RJ11 3-----------------------A ANALOGUE
FXS PORT 3 RJ11 4-----------------------B PHONE
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Phone in FXS Port in Key mode

I have a system in Key Mode, and cannot seem to figure out how to get an analog phone that's plugged into the FXS Port to make an outgoing call.

Hi Russ,
In pure key system mode the FXS port (and SPA502 phone) can't be configured as a member of a Shared FXO Line. There is no way to directly reach these phones on an inbound call. They can however, be used to make out bound calls if the system is configured in blended mode and the option 'Include in Outbound calls' is checked on the Shared FXO Lines page. In this scenario the FXS phone would dial 9 + address and the system will let the FXS use an idle FXO that has that capability enabled.
Hope this helps.
Chris

UC 540 Assigning an FXS port to a DID

Hello,
I have a UC 540 and an trying to redirect a DID number to voicemial, so the users can dial directly into voicemail
voice-port 0/0/0
station-id number 401
caller-id enable
voice-port 0/0/1
station-id number 402
caller-id enable
voice-port 0/0/2
station-id number 403
caller-id enable
voice-port 0/0/3
station-id number 404
caller-id enable
Via CCA telephony\ports and trunks\fxs ports I am unable to change 401 402 403 or 404
Via CCA telephony dial plan incoming, when trying to set this up it does not give me a trunk option
Any help is much appreciated

Hello,
Try and change the FXS extensions under:
Configure -> Telephony -> Users and Extensions -> User and Phones -> In the window that opens there is a tab for analog extensions
What are you using for PSTN connectivity? FXO, PRI, or SIP? Only PRI and SIP support DIDs.
Under Configure -> Telephony -> Dial Plan -> Incoming -> In the window that opens there is a tab for direct dials. You would click the new button to add a DID translation to an internal extension.
Thanks,
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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.
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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
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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
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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
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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

Cisco 2432-24FXS Calls won't ring FXS ports when placed in Huntgroup

Hi,
I am having an issue getting calls to ring through on the FXS ports when using the "trunkgroup" command under the POTS dial-peers. Calls will ring through fine if I use the "port" command under the pots dial-peer. What I am trying to accomplish is getting a single number to hunt through 13 FXS ports in a round-robin fashion. With the below setup I am getting a SIP 404 not found message in the SIP debug info. I am not sure if I am missing something here or what the deal is. If the below information is not enough I would be happy to provide additional info. Any idea's why this configuration doesn't seam to work?
trunk group HuntGroup1
hunt-scheme round-robin both up
dial-peer voice 1 voip
translation-profile outgoing 7to11
destination-pattern .T
progress_ind setup enable 3
no modem passthrough
voice-class codec 1
session protocol sipv2
session target dns:XXXXXXXX
dtmf-relay rtp-nte
fax rate 14400
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
ip qos dscp cs5 media
no vad
dial-peer voice 300 pots
trunkgroup HuntGroup1
description TEST NUMBER
destination-pattern XXXXXXXXXX
voice-port 2/0
trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 1
station-id number XXXXXXXXXX
caller-id enable
voice-port 2/1
trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 2
station-id number XXXXXXXXXX
caller-id enable
voice-port 2/2
trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 3
station-id number XXXXXXXXXX
caller-id enable
voice-port 2/3
trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 4
station-id number XXXXXXXXXX
caller-id enable
voice-port 2/4
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timeouts interdigit 3
description HuntGroup 1 Line 5
station-id number XXXXXXXXXX
caller-id enable
voice-port 2/5
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timeouts interdigit 3
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station-id number XXXXXXXXXX
caller-id enable
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trunk-group HuntGroup1
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description HuntGroup 1 Line 7
station-id number XXXXXXXXXX
caller-id enable
voice-port 2/7
trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 8
station-id number XXXXXXXXXX
caller-id enable
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trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 9
station-id number XXXXXXXXXX
caller-id enable
voice-port 2/9
trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 10
station-id number XXXXXXXXXX
caller-id enable
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timeouts interdigit 3
description HuntGroup 1 Line 11
station-id number XXXXXXXXXX
caller-id enable
voice-port 2/11
trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 12
station-id number XXXXXXXXXX
caller-id enable
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trunk-group HuntGroup1
timeouts interdigit 3
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station-id number XXXXXXXXXX
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Message was edited by: Thomas Schmidt
Added CCSIP Debug and Router Config.

Hi.
Can you please attach the complete config and a debug ccsip message?
Thanks
Regards
Carlo
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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.

Restricting FXS ports to internal calls only

Hi,
I have recently intalled Call Manager 8.5.1 with a H323 Gateway. On the gateway, I have a number of FXS ports for lobby phones. Is there a way I can restrict these phones to allow only internal calls only? Do I need to use COR lists like in Call Manager express or is there a more practical way?
Thanks,
Derek

Hi Derek,
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Installing an analog polycom soundstation 2 on FXS port in CUCME

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