Analog door phone to dial phones configuration

I have a remote office that I'm trying to setup CME. Everything works for now except one thing. There is a analog phone installed on outside of front door for visitors, deliveries, etc. Trying to setup this phone to ring automatically reception desk and also enable for others to answer the line from their phone when there is no one available in the reception area. I was thinking of connecting this door phone to one of the FXS ports on the router and configure PLAR to ring reception desk phone. Any suggestion? how do I setup other phones to be able to answer/pick up when no one at the reception desk? Pickup group? Thanks.

Thanks for your reply kkoeper12. I have configured the following. Did I get this right? I really can't test it at the moment. Just want to check first. Thanks.
voice hunt-group 1 sequential
final 2212
list 2212,2211,2213,2214,2202,2203,2204,2205
timeout 30
pilot 2227
voice-port 0/2/0 ---- this is where I have a analog door phone connected.
connection plar 2227
ephone-dn 26 dual-line
number 2227

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Error on UC560 configuring Door phone

Hi All.
I've come up with some error and can't find an answer what is causing that.
There is UC560 and 3rd party Door phone configured over SIP protocol. When i assign ip address of voice vlan to that phone it's not reachable, but when it's on data vlan then it's reachable but i'm getting this kind of error:
SIP/2.0 400 Request-URI MUST NOT have user
There is some config and debugging:
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol pass-through g711alaw
no fax-relay sg3-to-g3
sip
registrar server expires max 3600 min 3600
outbound-proxy ipv4:88.11.22.33:5060
no update-callerid
sip-profiles 1000
voice register global
mode cme
source-address 10.1.1.1 port 5060
max-dn 5
max-pool 5
load 9971 sip9971.9-2-2
load 9951 sip9951.9-2-2
load 8961 sip8961.9-2-2
time-format 24
create profile sync 0002444005002498
voice register dn 1
number 441
name doorbox
no-reg
voice register pool 1
id mac 0002.D619.0001
number 1 dn 1
dtmf-relay rtp-nte
username 441 password 12345
codec g711ulaw
no vad
And debugging:
Received:
REGISTER sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.175:5060;rport;branch=z9hG4bKPjzBkGsf92m-hRYybMttoDDAFgZ72Rv.SU
Max-Forwards: 70
From: <sip:[email protected]>;tag=9PhVyBvfTMgz0nYukYXX7ubo.VZplwtt
To: <sip:[email protected]>
Call-ID: gOBtNc3mNu2nnsQoiawp4vmqjk.rknhS
CSeq: 24450 REGISTER
Contact: <sip:[email protected]>
Expires: 1800
Allow: INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, OPTIONS, INFO
Content-Length: 0
028232: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 192.168.1.175,Port 5060, Transport 1, SentBy Port 5060
028233: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 192.168.1.175,Port 5060, Transport 1, SentBy Port 5060
028234: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Subsq Transaction Address 192.168.1.175,Port 5060, Transport 1, SentBy Port 5060
028235: //-1/9FE1CF99B825/SIP/Transport/sipSPISendResponse: Sending INFO Response to the transport layer
028236: //-1/9FE1CF99B825/SIP/Transport/sipSPITransportSendMessage: msg=0x8C9F77C4, addr=192.168.1.175, port=5060, sentBy_port=5060, local_addr=, is_req=0, transport=1, switch=0, callBack=0x0
028237: //-1/9FE1CF99B825/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
028238: //-1/9FE1CF99B825/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
028239: //-1/9FE1CF99B825/SIP/Transport/sipTransportLogicSendMsg: Trying to send resp=0x8C9F77C4 to default port=5060
028240: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:192.168.1.175, rport:5060 with laddr:
028241: //-1/9FE1CF99B825/SIP/Transport/sipTransportLogicSendMsg: Connection obtained...sending msg=0x8C9F77C4
028242: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x8C9F77C4, addr=192.168.1.175, port=5060, local_addr=, connId=3 for UDP
028243: //-1/9FE1CF99B825/SIP/Transport/sipSPISendResponse: Sending INFO Response to the transport layer
028244: //-1/9FE1CF99B825/SIP/Transport/sipSPITransportSendMessage: msg=0x8C9D225C, addr=192.168.1.175, port=5060, sentBy_port=5060, local_addr=, is_req=0, transport=1, switch=0, callBack=0x814ACF94
028245: //-1/9FE1CF99B825/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
028246: //-1/9FE1CF99B825/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
028247: //-1/9FE1CF99B825/SIP/Transport/sipTransportLogicSendMsg: Trying to send resp=0x8C9D225C to default port=5060
028248: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:192.168.1.175, rport:5060 with laddr:
028249: //-1/9FE1CF99B825/SIP/Transport/sipTransportLogicSendMsg: Connection obtained...sending msg=0x8C9D225C
028250: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x8C9D225C, addr=192.168.1.175, port=5060, local_addr=, connId=3 for UDP
028251: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.175:5060;rport;branch=z9hG4bKPjzBkGsf92m-hRYybMttoDDAFgZ72Rv.SU
From: <sip:[email protected]>;tag=9PhVyBvfTMgz0nYukYXX7ubo.VZplwtt
To: <sip:[email protected]>
Date: Thu, 20 Feb 2014 16:05:17 GMT
Call-ID: gOBtNc3mNu2nnsQoiawp4vmqjk.rknhS
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 24450 REGISTER
Content-Length: 0
028252: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
IPT-UC560#SIP/2.0 400 Request-URI MUST NOT have user
Via: SIP/2.0/UDP 192.168.1.175:5060;rport;branch=z9hG4bKPjzBkGsf92m-hRYybMttoDDAFgZ72Rv.SU
From: <sip:[email protected]>;tag=9PhVyBvfTMgz0nYukYXX7ubo.VZplwtt
To: <sip:[email protected]>;tag=2500D288-1EB9
Date: Thu, 20 Feb 2014 16:05:17 GMT
Call-ID: gOBtNc3mNu2nnsQoiawp4vmqjk.rknhS
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 24450 REGISTER
Content-Length: 0
If anybody would have a clue of is that I'll be highly appretiated.
Thanks

Hi Alex.
Thanks for your reply.
I'll try to implement those commands when i get to a client site next time.
I tried to put that phone on voice vlan and connected to a switch port which is used for normal ip phone but it was no connectivity after that i couldn't ping CME. Once i assigned data vlan ip address the connectivity was restored, and the door phone was trying to connect. I think it should work on both vlans.
Cheers,

2 uc500 tasks? building door locks and door phone?

Hello,
Every install seems to present unique requsts from each client.
Client has a uc540. Currently they have a keyfob door lock system at the front and back doors of the building. They have a phone/speaker at the front door. Appears to be avaya - but so not sure. This is all tied into an antiquated phone system the uc540 will replace.
When a person arrives at the front door, they hit the intercom. This rings 3 different extensions. Then the user at the phone can bunch a sequence or hard button on the phone to unlock the front door.
Anyone provided a solution like this for the uc540? I've seen some remote door locks that operate thru SIP a year or so ago I believe. I know how to configure a "lobby" phone that when picked up, dials a dedicated extension and immediately goes 2-way intercom. I imagine there would be something along the same lines where a picked up handset could auto-dial a parallel hunt group?
Thanks

Hello,
I would reference this document: https://supportforums.cisco.com/docs/DOC-9496
That doc discusses a door phone configuration with the UC. This requires CLI to configure the door phone as a SIP endpoint.
Thanks,
-john

Door phone, coverage path

Hi, I am configuring an analog push button door phone that I want to follow a certain call path until someone answers. I come from a traditional PBX environment where I had this implemented succesfully with time-of-day coverage paths (Avaya terms).
I have the phone hotdialing a number when the button is pushed, I just need a point in the right direction so that I can get the DN to follow a certain path until a user amswers. This path will change based on the time of day also.
Example: Push button - extension 100 rings, no answer - ring extension 200, etc. After 5 pm? then ring extension 300, then 400 so on.
Thanks for your help.
Dave

Assuming u have CM 4.1 or later.
TOD routing is possible with partitions, time periods and time schedules.
1. Configure extension 100 with call forward no answer set to 200. so if 100 doesnt answer, 200 will ring.
2. Define a time period in Callmanager, say its called as BusinessHours (M-F, 8am-5pm). 3. Add this time period to a time schedule. 4. Apply this time schedule to a partition that you define, say its called DoorPhonePn. So partition DoorPhonePn is active only during M-F, 8am-5pm.
5. Create a second partition say DoorPhoneBackupPn. You may or may not define a time schedule for this pattern.
6. Set Call Forward No Answer on extn 300 to ring extn 400. So if 300 doesnt answer, 400 rings.
7. When you setup a Plar in Callmanager, you have to define a null translation pattern. For TOD to work with Plar, you will need to define two null translation patterns.
8. Put Null translation pattern 1 in DoorPhonePn partition
9. Set Null translation pattern 1 to translate to extn 100.
10. Put Null translation pattern 2 in DoorPhoneBackupPn
11. Set Null translation pattern 2 to translate to extn 300.
12. Make sure the translation patterns have a CSS that has access to the partitions in which extn, 100 and 300 are.
13. Create a new CSS, called DoorPhoneCSS and list the partitions in this order.
DoorPhonePn
DoorPhoneBackupPn
During business hours, DoorPhonePn is active
So it will plar to 100.
After hours, DoorPhoneBackupPn is active, So it will plar to 300.
Here is a link on how to setup basic plar.
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a0080232b9f.shtml
HTH
Sankar
PS: please remember to rate posts.

Integration CME 8.6 with 2N IP Helios SIP Door Phone

Good Day,
I have a Cisco CME install on router 2911 with CME version 8.6 and i need to have any boday do the integration between CME 8.6 and SiP door intercome
my SIP door Intercome is 2N IP Helios Door Phone With 6 button , i already did integration before but with CME it was with cisco call manager business edition by create end user and integrate the user with the system and add that user in hunt group for particular phone how should ring...... any way if any boday have help in the same issue please give me ASAP.
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Hi
did you manage to get this working
i have video and audio working but cannot get the door to release from IPPhone.
Did you manage to get the door release working from 9951 to 2N Helios Door system.
can you share some configuration.
Regards
shameer

UC320W and Pantel Door Phone - Almost working !!!

I have an older single line analogue, single button Pantel doorphone that is working fine with a Cisco UC320W on its analogue port.
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I am not sure whether this is a Pantel issue or a Cisco issue so have also posted on Pantel customer support website!

Hi Oren,
we tried the configuration you've stated (we have a SPA9000 and a Pantel/Pancode IP Door Phone, but the door phone never manages to communicate with the extensions... And it's not also found from the SPA9000 ... Have you added the door phone as a client to the SPA9000 and if yes, how did you achieved that ?

Error accessing Phone Configuration page in Call Manager

I used BAT to delete auto-generated device profiles. Status and log file showed records were deleted successfully, however, I get the following error message when attempting to access the phone configuration page:
Error
The following error occurred while trying to retrieve the information for the requested phone.
selecting device: OK
Getting RIS for device: OK
Getting device properties: OK
authenticationStr [object Object]OK
keySizeList 2OK
certOperationList 1OK
upgradeFinishTime 2005:7:23:10OK
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packetCaptureMode 0OK
packetCaptureDuration 60 DONE selecting device: OK
Getting hotelling info:
The current item or one of its properties is not valid. It may have been deleted since the page was last refreshed. (2)
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Secretary IP Phone Configuration

Hi,
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I mean the best practices of configuring DNs, Speed Dials or something else
for such kind of employee role. Let's imaging he/she has got 7961 terminal
with 2 (two) 7914 expansion modules.
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Hi Tobivan,
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True Intercom can be setup;
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From this 6.x doc;
http://www.cisco.com/en/US/docs/voice_ip_comm/connection/2x/release/notes/cucmbe_relnotes/601ccmrn.html#wp218023
The Lines on the 7914's will likely change as well with the introduction of "built-in Presence"
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In Cisco Unified CallManager 6.x this is called - Speed Dial Presence
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States are:
Idle
Busy
Unknown
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_guide_chapter09186a008070bc02.html#wp1070507
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Hope this helps!
Rob

Analog conference phone on digital PBX?

It won't work on a DIGITAL line, but most Digital PBX had options for analog ports, and you can swap into that port just like you were hanging a fax line.

pretty much what it says up top, we've got a nortel norstar modular PBX that is (I think) digital, and I've got 2 polycom soundstation 2W conference phones that have apparently been collecting dust for 3 years. I could get one line peeled off and sent to our conference room just for the phones, but it looks like there are adapters that would allow at least SOME of our phone functions to work, but I'm having a bear of a time figuring out what I need. Can I get this phone working with this system, or should I stop fighting and just give it it's own line?
The phone line does come in via RJ11 to each office
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This topic first appeared in the Spiceworks Community

Nokia Phone Configuration Settings

This subject is not covered under any other headings so here goes
I have ordered standard and advanced settings for various Nokia phones from the the Nokia website
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display name
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Analog Emergency Phone Voltage Problem

We have a customer who has some emergency phones around their campus. These are made by Ramtel, and are simple devices on a pedestal, w/ two buttons, black and red. Typically, one dials security and the other dials 911, though we currently have both programmed to dial 911. You program the device by dialing into it. The device is programmed properly.
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I'm having an almost identical problem. These boxes were working fine (on vg224) and then one day they all decided, nope. We've discovered a lot of things.
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#3 Even ATT's line does not meet ramtel's spec
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I've tested the VG's with and without the alt-battery command and an ATT line and the only difference is ATT is putting out 46V and the VG maxes at 43-44. Everything else is identical (26ma on hook, 6V 12ma off hook). I'm going to call ramtel and see what they believe is going on since the voltage is in spec and the current is the same as what ATT is providing.

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I opened a TAC case and they say they dont have it either. LOL
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I phone configuration utility

hi all
sorry if this is on the wrong catagory
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winxp pro sp3
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iphone configuration utility errors on start up with. failed to locate apple application support.
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Its for use at home.
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DX650 cannot show caller ID name after speed dial is configured

Hi all,
Once I replaced my IP phone from CP-7975 with DX650, I found the incoming caller name display is missing for those caller phone numbers are configured as "one-button speed dial" on the DX650. For other callers calling to DX650, the caller name can be displayed correctly.
Anyone encounters this issue before and how to solve it?? Right now, DX650 is not mapped with the google account and other directory setting.
thanks,
samuel

Hello,
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and make a test , your probleme we will resolved.
Regards

Analog door phone to dial phones configuration

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