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.
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 -
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 -
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 advanceHi
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
regardsHi,
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 -
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. -
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 -
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 -
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,
DerekHi 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. -
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.
ThanksWhen 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? -
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?
ThanksTry 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.
BenNot 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 -
How to configure FXO and FXS port?
hi,
can anyone help me on how to configure FXO port on 2820 router and FXS port on 2810.
What are the things i need to configure to make a voip network that connect to PSTN network and PBX?hi
do refer this link which can provide you fair idea to get started with your configs.
http://www.cisco.com/en/US/partner/tech/tk1077/tech_configuration_examples_list.html
regds -
Door entry system on FXS port of UC320W
Hello,
I have inherited a door access system with no information on it. I have plugged it in to the FXS port on the UC320W and assigned it extension 199. I can dial 199 and press "8" and the door opens, so it is going well.
My problem is I don't know what number the door is auto dialling when you press the call button at the door end. I can hear the tones, so I know it is three digits, but I have no idea what they are. Is there any way to monitor the FXS port to see what number is being dialed when I press the call button?
Cheers,
TonyThis is the log, am I right in thinking it is dialing "5" or "325"?
Jul 5 15:38:42 UC320W user.debug voice: INVITE sip:[email protected]:6060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5080;branch=z9hG4bK-c3a41e2b
From: "Door" ;tag=15b8fd873df7b9cbo0
To:
Call-ID: xxxxxxxx
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Door"
Expires: 240
User-Agent: Cisco/UC320W-2.3.2(6)
Allow-Events: talk, hold, conference, x-spa-cti
P-Mailbox: 5199
Content-Length: 325
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 2244391 2244391 IN IP4 10.1.1.1
s=-
c=IN IP4 10.1.1.1
t=0 0
m=audio 16476 RTP/AVP 0 2 8 18 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
Jul 5 15:38:42 UC320W user.debug voice: SIP/2.0 404 Not Found
To: ;tag=1a27709e-0
From: "Door" ;tag=15b8fd873df7b9cbo0
Call-ID: xxxxxxxxx
CSeq: 101 INVITE
Via: SIP/2.0/UDP 10.1.1.1:5080;branch=z9hG4bK-c3a41e2b
Server: Cisco/UC320W-2.3.2(6)
Allow-Events: talk, hold, conference, x-spa-cti
Content-Length: 0
Jul 5 15:38:42 UC320W user.debug voice: ACK sip:[email protected]:6060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5080;branch=z9hG4bK-c3a41e2b
From: "Door" ;tag=15b8fd873df7b9cbo0
To: ;tag=1a27709e-0
Call-ID: xxxxxxxx
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Door"
User-Agent: Cisco/UC320W-2.3.2(6)
Allow-Events: talk, hold, conference, x-spa-cti
P-Mailbox: 5199
Content-Length: 0 -
Transmission Volume on a FXS Port
G'Day,
We have a fax machine connected to an FXS port in a 2851 (CME 4.1). Running tests on the machine reveal that the transmission volume is set too high, causing distortion on faxes and an inability to fax to some destinations.
Is there a way to turn the transmission volume down at the port-level? Apparently it cannot be done on the fax machine (? this sounds odd to me...but the technician for the machine is adament)
Thanks,
-marti-This depends if the faxing is being done via fax relay or fax passthrough.
If fax relay is being used, the fax traffic originates from the egress voice port connected to the PSTN, not from the FXS port that has the fax machine on it. The output level of the fax relay codec is fixed - at this stage it is not possible to alter the level of the outgoing audio on the 2800/3800 ISR's.
If you use fax passthrough, the audio is transmitted via a G711 codec. You can use the voice port 'input-gain' or 'output-attenuation' command to boost or reduce the levels in different directions.
If the level FROM the CME system towards the PSTN is too high (for example 3dB), you would use the command 'input-gain -3' on the FXS port to reduce the inwards audio level by 3dB, so the level SENT to the PSTN is correspondingly reduced by 3dB.
Most fax machines can set the outgoing audio levels - they may have a special config menu that is not normally directly accessible by users.
Maybe you are looking for
-
Like I said when I bought my mac in the apples store, iMovie was already included. When it comes time to upgrade it, it won't let me and it says to sign into the account I used to purchase it. But I didn't purchase it, it came along with the computer
-
When I go to "You" on my Flickr account, most of my photos do not show up. I cannot edit them or add titles because i cannot see them. Only the ones with photos can be brought up. When I open my groups many photos do not show up, only the the photogr
-
External HD not showing up on my new Macbook Air but works on iMAC
Hi, I am struggling to get my new Air book 11" to work with my external drive. it is visible on my iMac but does not show up at all on my Air Book. Can some one advice? I have done all possible options including resets, reformat, preference check etc
-
Applejack hung up on permissions
When I run Applejack on my PowerBook G4 (10.4.5), it works just fine until it gets to the Repair Permissions step, and there it hangs. It tells me that it takes a while, please wait, and then it begins generating its row of dots. An hour later it's u
-
Turning imac display off without putting it to sleep questions?
I like to leave my iMac on overnight but I don't want to put it to sleep COMPLETELY. Just to turn off the display. I heard you press control-shift-eject or do a hot spot on screensaver on system pref. But my question is, would it hurt the iMac in the