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...

Similar Messages

Configuring pickup groups

We have a site that has a PagePac unit connected to a FXS port. CM is
4.1.3sr2.
When the receptionist steps away / end of day, they forward the main
number to the overhead night bell.
The Night Bell is in a pickup group, as are all the phones, so that
after hours anyone can pickup incoming calls.
The problem is that because all phones are in the same pickup group, any
user can pickup anyone else's ringing phone, which they don't want. The
other issue is that once a user hears the overhead, they press Pickup
and the incoming call starts ringing at that phone, they get CallerID,
and then can't hit iDivert to send it back to the main number's VM box
if they don't want the call. Instead, they have to wait for RNA in
order for the call to end up in the main number VM.
I investigated the other pickup options:
Group Pickup might work, but means that it's now a two-step process to
pickup a call, but it will allow me to put the phones in one pickup
group and have the overhead in another pickup group. I would remove
"Pickup" from the softkey template for the phones.
I tried to configure OPickup, thinking that I could put the phones in a
pickup group where the pickup group number was in a a partition that wasn't in their calling search
space, and have the overhead in an associated pickup group that _was_ in
their CSS. Doesn't work; CallManager won't let you configure a DN with
a pickup group whose partition isn't in your CSS.
Any other ideas on how to implement this functionality?

Cisco AS recommends "longest idle hunt"mechanism among the members to be the most fair for the hot line staff. For overflow condition, Cisco AS suggest that we use vm dn. However, if you put the vm dn in the longest idle hunt group, you occasionally will have users rolling directly into the vm. To avoid this, create a hunt group with "first available hunt " and put the pilot of the "longest idle hunt" (in a nested fashion) first and then the vm DN (with always route option checked) second.

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
trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 5
station-id number XXXXXXXXXX
caller-id enable
voice-port 2/5
trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 6
station-id number XXXXXXXXXX
caller-id enable
voice-port 2/6
trunk-group HuntGroup1
timeouts interdigit 3
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
voice-port 2/8
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
voice-port 2/10
trunk-group HuntGroup1
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
voice-port 2/12
trunk-group HuntGroup1
timeouts interdigit 3
description HuntGroup 1 Line 13
station-id number XXXXXXXXXX
caller-id enable
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
Sent from Cisco Technical Support iPhone App

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.

Source address for FXS port

My confusion is about the source address that voice packets assume for a FXS port in a Ciso router.
I am pasting relevant configuration from 2 routers below.
For the 1st router I have the session targets in the dial peer config as the loopback addresses but the QoS is working using a access-list where the source address is the serial ip.
While in the other router I am getting no packet matches for either the loopback ip or the serial ip.
ROUTER 1
class-map shell_voip
match access-group 170
policy-map shell_voip
class shell_voip
priority 64
class class-default
fair-queue
random-detect
interface Loopback0
ip address 10.66.12.25 255.255.255.255
interface Multilink101
mtu 100
bandwidth 1544
ip address 10.66.50.14 255.255.255.252
no ip mroute-cache
load-interval 30
service-policy output shell_voip
no cdp enable
ppp multilink
ppp multilink fragment-delay 20
ppp multilink interleave
multilink-group 101
access-list 170 permit udp host 10.66.50.14 range 16000 35000 any range 16000 35000
access-list 170 permit tcp any eq 1720 any
access-list 170 permit tcp any any eq 1720
voice-port 2/0
cptone IN
voice-port 2/1
input gain -6
cptone IN
dial-peer voice 1 pots
destination-pattern 40
port 2/0
dial-peer voice 100 voip
destination-pattern 10
session target ipv4:10.129.67.105
dial-peer voice 2 pots
destination-pattern 99
port 2/1
dial-peer voice 102 voip
destination-pattern 11
session target ipv4:10.129.67.105
ROUTER 2
no voice hpi capture buffer
no voice hpi capture destination
class-map match-all Vsp_voice
match access-group 160
policy-map Vsp_voip
class Vsp_voice
priority 32
class class-default
fair-queue
random-detect
interface Loopback0
ip address 10.65.10.121 255.255.255.248
interface Multilink60
ip address 10.65.50.246 255.255.255.252
service-policy output Vsp_voip
load-interval 30
no cdp enable
ppp multilink
ppp multilink fragment delay 10
ppp multilink interleave
ppp multilink group 60
access-list 160 permit udp host 10.65.50.246 range 16000 35000 any range 16000 35000
access-list 160 permit tcp any eq 1720 any
access-list 160 permit tcp any any eq 1720
voice-port 2/0
cptone IN
voice-port 2/1
cptone IN
dial-peer cor custom
dial-peer voice 9 pots
destination-pattern 1101
port 2/0
dial-peer voice 10 pots
destination-pattern 1102
port 2/1
dial-peer voice 5 voip
destination-pattern 8901
session target ipv4:10.196.3.57
dial-peer voice 6 voip
destination-pattern 8902
session target ipv4:10.196.3.57

You may want to refer to the following link.
http://www.cisco.com/en/US/products/sw/iosswrel/ps1834/products_feature_guide09186a0080080115.html
Your dial peers are using H.323, your source will be what ever interface is used to exit the router as determined by the routing table.
You could also use a debug IP packet to have a look at your source and destination if you are unsure.
For this case you may want to just apply:
h323-gateway voip bind srcaddr 10.66.12.25 on Router 1 and h323-gateway voip bind srcaddr 10.65.10.121 to Router 2. Rememeber to put them under the loopback interface.

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

Is there a way to answer a PickUp group if you are on the phone?

I have a user with a standard 7942 IP phone with a single DN on a CUCM 8.1 server. She can place a call on hold and make or receive a new call.
However, she's also part of a call Pickup group. She's stating that there isn't a softbutton option to pickup a call from this pickup group if she's on her current line.
Is there something that can be set to allow her to answer the pickup group if she's currently on the phone?
I know that's very basic...if there's something else I need to include configuration wise, please let me know. I just don't want to clutter up the question.

Thanks Jaime,
Does that mean that there isn't an alternative keypress or similar that will allow her to pick up something in the call pickup group short of hanging up on the existing call?

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

Adding an extension to a Call Pickup Group

I am new to call manager 7.0 and just wanted to know if adding an extension to a Call Pickup group was as easy as just going to the directory number configuration page of the extension and selecting a call pickup group within the "Call Forward and Call Pickup Settings?" Would there be anything else I would need to do?

nope, that's it
Assigning a Call Pickup Group to Directory Numbers
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fscpickg.html#wp1056555
HTH
java
if this helps, please rate

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,
It sounds like these FXS ports are NOT registered to CUCM 8.5.1? If they aren't, then yes COR lists is the most pratical way to do this. You could do a "connection plar xxxx" to a receptionist extension and her her/him forward the calls to the correct extension.

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

7962 Can't use pickup group

Hey guys, maybe someone can help me out here. I have a pickup group and about 15 phones that are using it. One of the phones, a 7962, cannot pickup the calls. The CSS does have access to the pickup group partition. I've reset the phone several times and even rebuilt the phone completely. Stare and compare reveals no difference in the configs.
The user reports when they press the pickup softkey they get a key not available message. I have confirmed this is the behavior experienced when the line is not a member of a pickup group at all.
CUCM trace reveals nothing in the logs from that phone except for keepalives during a test call. It's like the phone never requests the pickup from CUCM. Another phone that is able to use pickup and answer the call has all kinds of messages from pickupmanager in the SDL.
I opened a TAC case but they are taking their time getting back to me. Any ideas?
Thanks!

Bug below references 7960 phones:
CSCsb74215
CallPickup is Broken !!!
1. I have Three 7960 Phones ---> Phone A ( 1400) in CallPick Group -->1384020005 , Phone B ( 1399 ) ---> in Callpick Group ---> 1384020005 .
Now Phone C ( 1403) Calls Phone A ( 1400 ) , Now don't answer the call on Phone A , Pick the Call from Phone B as both are in same Pickup Group , mesage displayed " No Call Available for Pickup ".
Known Affected Releases: (1)
5.0(1.51.35)
If you don't believe bug is related:
What is the call manager version?
phone firmware version?
Is the line on the 7962 a shared line? If it is a shared line, is part of the same pickup group?
Hope this helps,
Please rate helpful answers,
Thanks.

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?
Then change “Subscriber Information -> Contact list” to202. Then on the FXS 2 tab we should configure: User ID=202
Does this look correct?
What if we have two DID numbers associated with a single SIPaccount? Can we still direct one of the DID’s to the FXS 2 port?

FXS Port Query?

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.
Out of the 8 ports, two continually go off-hook, FXSLS_WAIT_OFFHOOK before finally going into a FXSLS_PARK state. I have moved the cabling and the error occurs on which ever port the two lines plugs into.
Has anyone ever see this before. To my mind it looks like a cabling issue?
Any ideas?
Thanks

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
Ensure that all other wires are left disconnected

EFTPOS No Response X0 Using FXS Port

Hi All,
Trying to get an EFTPOS machine working connected into an FXS Port using SCCP/STCAPP Protocol for call control.
The port is registered in CUCM, I can connect an analogue phone to the port and successfully call the Bank's 1800 Number etc, I can make other external calls no problem.
When I try the EFTPOS Machine, the call setup runs through normally and I see the call is then in a connected state. Approx about 10 or seconds later the call is cleared by the local device with an 0x8090 code Normal Clearing. The call is then disconnected and released as per any other call.
The EFTPOS Machine prints a error for "Declined No Response X0"
I have attached the following debugs.
debug isdn q931
debug voip vtsp all
debug vop ccapi inout
debug voip rtp session named-event
Calling Number is 0245604627
Called Number is 1800509183
General environment setup is as follows.
EFTPOS connect to FXS Port -> Registered to CUCM (9.1.2) -> SIP Trunk -> IOS 15.x -> ISDN Pri.
Any feedback i smuch appreciated.
Ben

Not so sure if the service code being utilize by the FXS is the same as what is mentioned in VSA.Then again, the manual says "dial the corresponding * code on the client station" and if you look at an SPA9xx phone, it has the following codes defined for call park and call unpark code.
*38
*39

FXS Ports & Pickup Groups

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