10-31-2016 07:56 AM - edited 03-17-2019 08:32 AM
Hi All,
I want to configure the Polycom sound-station 2w conference phone in Cisco Call manager Express. It is a analog sip phone. I have done the SIP configuration for the 3rd party sip phone. But still the device is not up. .
CME version: Version 10.5
Phone Model: SoundStation 2W
Configuration :
1. Allow Calls among SIP endpoints, binding the source interface for SIP Traffic and Defines SIP Register server
voice service voip
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/0/1.40
bind media source-interface GigabitEthernet0/0/1.40
registrar server
2. SIP CME configuration
voice register global
mode cme
source-address <IP of GigabitEthernet0/0/1.40> port 5060
timeouts interdigit 5
max-dn 100
max-pool 10
authenticate register
authenticate realm local
timezone 42
date-format D/M/Y
voicemail 1199
tftp-path flash:
create profile sync 0230410430385285
conference hardware
camera
video
3. Configuring a Directory Number for a SIP Phone
voice register dn 2
number 3205
call-forward b2bua busy 1199
call-forward b2bua mailbox 1199
call-forward b2bua noan 1199 timeout 20
name Polycom_Conf x3205
huntstop channel 1
mwi
4. Configuring IP Phone using the following commands
voice register pool 2
busy-trigger-per-button 1
id mac 2074.9054.5121
number 1 dn 2
template 1
dtmf-relay rtp-nte sip-kpml
voice-class codec 1
username conf password conf1
no vad
5. FXS port configuration
voice-port 0/2/0
timeouts ringing infinity
description Polycom_Conf
caller-id enable
6. Configuration of POTS & VOIP dial peer
dial-peer voice 8 pots
translation-profile outgoing OUTDID
destination-pattern 9911
port 0/2/0
!
dial-peer voice 9 voip
destination-pattern 3025
session protocol sipv2
session target ipv4:10.29.203.1:5060
dtmf-relay rtp-nte sip-kpml
codec g711ulaw
no vad
So please help me on this.
Regards,
Rajeshpat
Solved! Go to Solution.
10-31-2016 10:01 AM
Hi Rajeshpat,
I believe the Polycom Soundstation 2w is a purely "analog" device and as such would need to connect to an FXS port (not a SIP config);
http://www.polycom.com/content/dam/polycom/common/documents/data-sheets/soundstation2w-ds-enus.pdf?utm_source=google&utm_medium=cpc&utm_term=polycom%20soundstation%202w&utm_campaign=CAN-B-Voice-Q416&utm_content=Conference&gclid=CjwKEAjw19vABRCY2YmkpO2OzTsSJAAzEt8sZ90w1KeLPPqCTtPS1jfvZS8VjqJBSbv5joAJXV1QXBoCCvLw_wcB
There are some nice config notes in this discussion;
https://supportforums.cisco.com/discussion/12039951/installing-analog-polycom-soundstation-2-fxs-port-cucme
Cheers!
Rob
10-31-2016 10:01 AM
Hi Rajeshpat,
I believe the Polycom Soundstation 2w is a purely "analog" device and as such would need to connect to an FXS port (not a SIP config);
http://www.polycom.com/content/dam/polycom/common/documents/data-sheets/soundstation2w-ds-enus.pdf?utm_source=google&utm_medium=cpc&utm_term=polycom%20soundstation%202w&utm_campaign=CAN-B-Voice-Q416&utm_content=Conference&gclid=CjwKEAjw19vABRCY2YmkpO2OzTsSJAAzEt8sZ90w1KeLPPqCTtPS1jfvZS8VjqJBSbv5joAJXV1QXBoCCvLw_wcB
There are some nice config notes in this discussion;
https://supportforums.cisco.com/discussion/12039951/installing-analog-polycom-soundstation-2-fxs-port-cucme
Cheers!
Rob
10-31-2016 10:55 AM
Hi Rob,
Thanks for your valuable guidance. Yes, your right. I have configured the FXS with STCAPP using SCCP protocol and created the POTS dial peer with stcapp service pointing to the FXS port. Finally configured ephone-dn and ephone. Now the phone is working. Please find the detail configuration below
Configuration:
sccp local <interface of CME>
sccp ccm <IP Address of CME> identifier 1 priority 1 version 7.0
sccp
!
sccp ccm group 1
bind interface <interface of CME>
associate ccm 1 priority 1
associate profile 1 register SC-CFB
--------------------------------------------------------------
stcapp ccm-group 1
stcapp
-------------------------------------------------------------
voice-port <fxs_port>
timeouts ringing infinity
description Polycom_Conf
caller-id enable
-----------------------------------------------
dial-peer voice 8 pots
service stcapp
port <fxs_port>
---------------------------------------
ephone-dn 29 octo-line
number 3204
description Conference Ad-hoc
conference ad-hoc
preference 1
no huntstop
ephone-dn 28
number 3206
description Polycom Conference
call-forward busy 1199
call-forward noan 1199 timeout 10
mwi sip
---------------------------------------------------------
ephone 37
description Polycom Conference
mac-address <XXXX.XXXX.XXXX> //to find the mac use the cmd: sh stcapp device summary //
max-calls-per-button 2
username "conf" password conf1
type anl
button 1:28
Regards,
Rajeshpat
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