Pantel VoIP door phone with UC520

pmountcastle's picture
Submitted by pmountcastle on Mon, 07/07/2008 - 20:03

I am trying to connect a Pantel VoIP door phone (one-button model) to my UC520 that is currently in production. I have followed some other posts here to modify config to allow for SIP to SIP communication. However, 2 things still bother me;

1. Door phone has an IP on the data network - If I try to statically assign an IP from the voice network, I lose connectivity. Stupid quesiton, but door phone needs to have an IP on my voice network, correct?

2. Phone registers with UC520. Tested yields a busy signal when the button is pressed.

Below is the config (modified for SIP use) that is currently in place. Can anyone shed some light on this issue?

Thanks,

Paul-

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
fax protocol none
modem passthrough nse codec g711ulaw
sip
registrar server expires max 1200 min 300
no update-callerid
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
!
!
!
!
!
!
!
!
!
voice register global
mode cme
source-address 10.1.1.1 port 5060
max-dn 1
max-pool 1
authenticate register
time-format 24
date-format D/M/Y
voicemail 90
file text
create profile sync 0246360689430548
!
voice register dn 1
number 210
allow watch
name Pantel SIP
label 210
!
voice register pool 1
id mac 0050.C26F.13DF
number 1 dn 1
dtmf-relay sip-notify
username 210 password 123456
codec g711ulaw

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Comments

wizit's picture

Pante\Pancode IP Vs. Cisco Call Manager

by wizit - 03/14/2010 - 08:35

Found a solution in a undocumented setting on the Cisco. Apparently there is a way to FORCE RFC2833 DTMF on a SIP TRUNK only. Which means the phone cannot register to the system. Here are the applicable cisco commands.

  •  This forces ulaw for the phone 

voice class codec X ---- Replace X with an unused codec number codec preference X g711ulaw ---- Replace X with an unused codec number - This forces ulaw for the phone voice class codec X ---- Replace X with an unused codec number codec preference X g711ulaw ---- Replace X with an unused codec number
 

  •  This sets up your SIP trunk

dial-peer voice X voip ---- Replace X with unused dial-peer number description Pancode Door Phone translation-profile outgoing PSTN_Outgoing destination-pattern XXXX ---- Replace XXXX with the extension number voice-class codec 1 voice-class sip dtmf-relay force rtp-nte ---- This forces rtp-nte regardless of failed negotiation session protocol sipv2 session target ipv4:XXX.XXX.XXX.XXX ---- IP address of the phone dtmf-relay rtp-nte no vad On the pancode, set the user the whatever destination pattern you set above. Leave the password blank. Set the domain and proxy server settings to your CME system IP.
 

  • This seems to do the job without issue. 

 
Great Product - recommended for all IP PBX's + they have a solution for VIDEO over IP with this product.

cjsnet1's picture

No Joy

by cjsnet1 - 11/07/2008 - 17:32

Hiya,

Ended up speaking with the Cisco TAC re the DTMF issue and confirmed it must be a bug on the Pancode. Everything seemed to work ok with the Pancode except the door release relay. i.e. we could call from it, call to it, answer, etc...

Anyway we bought a Dallas Delta SIP door phone and happy days this worked fine.

Cheerios

cjsnet1's picture

Status on this?

by cjsnet1 - 10/11/2008 - 21:08

I have one of these Pantel door phones and also need help trying to get the door relay option to work correctly.

Issue:
- DTMF from Cisco SCCP handets not opening the Pantel door phone relay.

Notes:
- The Pantel is registered correctly within the CME (sh voice regi pool 1) so I can call it from another extension and also have the Pantel call extensions \ hunt groups without any problems.
- The Pantel DTMF option is set to ‘Transparent’
- I Can call the Pantel from a CME registered SIP Linksys SPA942 handset which opens the door relay, but not from a SCCP Cisco handset.

Questions:
- Anyone have this working in production? If so can you please pass on the Cisco router config snippets and Pantel config details?

Cheers

pmountcastle's picture

UC520LA01#sh sip status SIP

by pmountcastle - 07/22/2008 - 19:36

UC520LA01#sh sip status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED

SIP User Agent for TLS over TCP : ENABLED
SIP User Agent bind status(signaling): DISABLED
SIP User Agent bind status(media): DISABLED
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards : 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
SIP support for ISDN SUSPEND/RESUME: ENABLED
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED

SDP application configuration:
Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported: audio image
Network types supported: IN
Address types supported: IP4
Transport types supported: RTP/AVP udptl

Paul-

pmountcastle's picture

Update 2

by pmountcastle - 07/22/2008 - 19:21

Please see below. Do I need to enabled signaling?

UC520LA01#sh sip status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED

SIP User Agent for TLS over TCP : ENABLED
SIP User Agent bind status(signaling): DISABLED
SIP User Agent bind status(media): DISABLED
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards : 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
SIP support for ISDN SUSPEND/RESUME: ENABLED
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED

SDP application configuration:
Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported: audio image
Network types supported: IN
Address types supported: IP4
Transport types supported: RTP/AVP udptl

Paul-

pmountcastle's picture

Update

by pmountcastle - 07/22/2008 - 19:17

I did more shooting in the dark and added the following:

UC520LA01#config t
UC520LA01(config)#sip
UC520LA01(config-sip-ua)#credentials username pancode password 654321 realm 10.1.1.1

Customer now able to place calls from door phone to internal extension; however, when fully tested with door lock, it appears that the tone for the key pressed (8 in our case) does not get back to the door phone to release the lock. I've modified the DTMF signaling to transparent, but have not tested yet.

Has anyone configured signaling from a SCCP phone back to a SIP phone successfully?

Thanks,

Paul-

marchern's picture

Check this app note for ideas

by marchern - 10/11/2008 - 21:26

http://www.uc500.com/door-phoneintercom-integration-uc500

Try also changing the CME side to 2833 DTMF relay.

Good luck!

Marcos

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