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!

Similar Messages

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

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

Adding additional FX0 ports on uc540 via gateways

Hi,
Our sales rep has sold a UC540W-BRI-K9 to a customer but the customer doesn't have any BRI connections, they use PSTN lines.
The issue is that they have 7 PSTN lines. We can purchase a 4 port FXO interface card but it's not enough to connect all the PSTN lines because the UC540 has only one expansion port..
I guess the only option is to purchase a gateway device to allow the additional FXO.
My question is, which media Gateway is supported on the UC540 via CCA programming?
I know that SPA8800 is supported on the UC540 with the new software pack but it's only for the FXS ports.
Are there any other devices?
Regards
Raymond

Hi,
I am looking for possible solutions for adding an additional FXO line to a UC540 as well. Mine already has the additional VIC2-4FXO module and all FXO ports are used. From reading the Manuals, Release Notes, and Forums, it sounds like the 8800 will only support additional FXS ports. I have not found a solid answer as to why that is. It seems like it would be a hardware/platform overload issue for the UC540 to support more than 8 FXO lines. Is this assumption accurate or is there something we can do about it?

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.

H323-gateway voip interface h323-gateway voip bind srcaddr could i configure this on l2 port

h323-gateway voip interface
h323-gateway voip bind srcaddr
how Can i configure this to L2 Link ?

Hi,
Router#
Router# config t
Router(config)# int fas ((your L2 link name))
Router(config-if)# h323-gateway voip interface
Router(config-if)# h323-gateway voip bind srcaddr ((ip_address))
Router(config-if)#end
Router#wr
Regards

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.

Caller ID not working on H323 Gateway

Hi,
I have a CCM Version 4.1.3 set-up to cater for 200 phones, I have a VG gateway in one of the sites that uses an FXO card for local calls, this VG is set-up as an H323 gateway, however calls coming throguh the PSTN line onto the FXO card and then onto an IP phone are not displaying Caller ID.
Just wondering if anyone has seen this before, I know that on an MGCP gateway the FXO cards don't support Caller-ID but we have it set-up as H323.
Thanks
Paul

Funny, this is a question straight from one of the CCVP tests, although I won't say which one.
Try "caller-id enable" in global mode and "station-id name" on the voice-port.
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_configuration_example09186a00800a9a49.shtml

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

CUCM and H323 gateway-Cause i = 0x80A6 - Network out of order

Hi,
I cant get calls into the CUCM from a H323 gateway. Incoming external calls here out of service message or Number Unobtainable. I've attched logs if anyone can help?
Rich

Hi Alex,
Thanks for reply. I should've attached the full config as it shows the Translation Rules. See below.
Can anyone help?
voice translation-rule 2
rule 1 /^56/ /5\2/
rule 2 /^6/ /5/
rule 3 /2/ /52/
voice translation-rule 3
rule 1 /^1$.........$$/ /01\1/
rule 2 /^2$.........$$/ /02\1/
rule 3 /^7$.........$$/ /07\1/
rule 4 /^8$.........$$/ /08\1/
rule 5 /^4$......$$/ /01914\1/
rule 6 /^2$......$$/ /01912\1/
voice translation-rule 6
rule 1 /^[1-9]/ /90\0/ type international international
voice translation-rule 7
rule 1 /^1/ /901/
rule 2 /^2/ /902/
rule 3 /^3/ /903/
rule 4 /^4/ /904/
rule 5 /^5/ /905/
rule 6 /^6/ /906/
rule 7 /^7/ /907/
rule 8 /^8/ /908/
rule 9 /^9/ /909/
voice translation-profile INCOMING
translate calling 7
translate called 2
voice translation-profile OUTGOING
translate called 3
voice-card 0
dsp services dspfarm
interface FastEthernet0/0
ip address 192.168.178.66 255.255.255.192
ip pim dense-mode
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.178.66
interface FastEthernet0/1
no ip address
duplex auto
speed auto
no keepalive
interface BRI0/1/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
isdn incoming-voice voice
isdn static-tei 0
interface BRI0/1/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
isdn incoming-voice voice
isdn static-tei 0
control-plane
voice-port 0/1/0
translation-profile incoming INCOMING
compand-type a-law
cptone GB
description ISDN2 .......... lines 1+2
voice-port 0/1/1
translation-profile incoming INCOMING
compand-type a-law
cptone GB
description ISDN2 .......... lines 3+4
ccm-manager music-on-hold
mgcp fax t38 ecm
dial-peer cor custom
dial-peer voice 999 pots
destination-pattern 999
port 0/1/0
forward-digits all
dial-peer voice 9991 pots
destination-pattern 999
port 0/1/1
forward-digits all
dial-peer voice 112 pots
destination-pattern 9112
port 0/1/0
forward-digits 3
dial-peer voice 1121 pots
destination-pattern 9112
port 0/1/1
forward-digits 3
dial-peer voice 9999 pots
destination-pattern 9999
port 0/1/0
forward-digits 3
dial-peer voice 99991 pots
destination-pattern 9999
port 0/1/1
forward-digits 3
dial-peer voice 100 voip
preference 2
destination-pattern 52...
progress_ind setup enable 3
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.0.150
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 101 voip
preference 1
destination-pattern 52...
progress_ind setup enable 3
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.203.20
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 1 pots
translation-profile outgoing OUTGOING
preference 3
destination-pattern 0T
translate-outgoing called 10
incoming called-number .
fax rate disable
direct-inward-dial
port 0/1/0
forward-digits all
dial-peer voice 11 pots
translation-profile outgoing OUTGOING
preference 4
destination-pattern 0T
incoming called-number .
fax rate disable
direct-inward-dial
port 0/1/1
forward-digits all
dial-peer voice 9 pots
translation-profile outgoing OUTGOING
preference 1
destination-pattern 9T
incoming called-number .
fax rate disable
direct-inward-dial
port 0/1/0
forward-digits all
dial-peer voice 91 pots
translation-profile outgoing OUTGOING
preference 2
destination-pattern 9T
incoming called-number .
fax rate disable
direct-inward-dial
port 0/1/1
forward-digits all
dial-peer voice 2 pots
translation-profile outgoing OUTGOING
destination-pattern 2......
incoming called-number .
fax rate disable
direct-inward-dial
port 0/1/1
forward-digits all
dial-peer voice 21 pots
translation-profile outgoing OUTGOING
destination-pattern 2......
incoming called-number .
fax rate disable
direct-inward-dial
port 0/1/1
forward-digits all
dial-peer voice 52982 voip
preference 1
destination-pattern 562982
progress_ind setup enable 3
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.0.150
dtmf-relay h245-alphanumeric
no vad

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...

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.

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

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.

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.

Adding MGCP FXS Ports to H323 Gateways

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