Howto:FXS FXO trunk with IP22 media gateway and Cisco 1700 router: Difference between revisions
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Cisco FXO gateway must forward incoming calls to IP22. | Cisco FXO gateway must forward incoming calls to IP22. | ||
This automatic mechanism is realized with PLAR OPX feature. | This automatic mechanism is realized with PLAR OPX feature. | ||
The private line automatic ringdown Off-Premises extension allows remote FXS devices to appear to a central PBX as a directly connected extension.When the FXO port detects a ring from PBX, the Cisco router sends a VoIP call setup to IP22 but it does not take the FXO port off-hook. As a result, the PBX only sees the call answer signal when IP22 FXS port is picked up. | The private line automatic ringdown Off-Premises extension allows remote FXS devices to appear to a central PBX as a directly connected extension. When the FXO port detects a ring from PBX, the Cisco router sends a VoIP call setup to IP22 but it does not take the FXO port off-hook. As a result, the PBX only sees the call answer signal when IP22 FXS port is picked up. | ||
In the same way, IP22 must forward outgoing calls to FXO gateway. | In the same way, IP22 must forward outgoing calls to FXO gateway. | ||
| Line 39: | Line 39: | ||
playout-delay mode fixed | playout-delay mode fixed | ||
cptone IT | cptone IT | ||
timeouts call-disconnect 1 | timeouts call-disconnect 1 | ||
timeouts wait-release 1 | timeouts wait-release 1 | ||
timing guard-out 300 | timing guard-out 300 | ||
connection plar opx 102 | connection plar opx 102 <--- incoming calls from PBX generate calls to phone number 102; Cisco SIP dial-peer number 2 is matched | ||
! | ! | ||
voice-port 2/1 | voice-port 2/1 | ||
| Line 53: | Line 53: | ||
playout-delay mode fixed | playout-delay mode fixed | ||
cptone IT | cptone IT | ||
timeouts call-disconnect 1 | timeouts call-disconnect 1 | ||
timeouts wait-release 1 | timeouts wait-release 1 | ||
timing guard-out 300 | timing guard-out 300 | ||
connection plar opx 103 | connection plar opx 103 <--- incoming calls from PBX generate calls to phone number 103; Cisco SIP dial-peer number 2 is matched | ||
Dial Peer, command line configuration: | Dial Peer, command line configuration: | ||
dial-peer voice 2 voip | dial-peer voice 2 voip | ||
description SIP_CALL_TO_IP22 | |||
destination-pattern 1.. | destination-pattern 1.. | ||
modem passthrough nse codec g711alaw | modem passthrough nse codec g711alaw | ||
| Line 75: | Line 76: | ||
! | ! | ||
dial-peer voice 15 pots | dial-peer voice 15 pots | ||
description | description FROM_IP22_TEL3 | ||
destination-pattern 203 | destination-pattern 203 | ||
port 2/1 | port 2/1 | ||
! | ! | ||
dial-peer voice 16 pots | dial-peer voice 16 pots | ||
description | description FROM_IP22_TEL2 | ||
destination-pattern 202 | destination-pattern 202 | ||
port 2/0 | port 2/0 | ||
Revision as of 20:09, 6 April 2009
This article describes how to realize an analog trunk using Innovaphone FXS media gateway and Cisco router.
Applies To
This information applies to
- IP 22, V7
Build 09-70300.14 and later.
- Cisco 1700 router
IOS C1700-IPVOICEK9-M, Version 12.4(19b)
More Information
Problem Details
In this case Innovaphone IP22 is used as FXS media gateway with a Cisco 1700 Router to realize an analog PBX extension over an IP network. The VoIP signaling protocol used is SIP.
Installation
Configuration
Connect devices as show in figure.
Cisco FXO gateway must forward incoming calls to IP22. This automatic mechanism is realized with PLAR OPX feature. The private line automatic ringdown Off-Premises extension allows remote FXS devices to appear to a central PBX as a directly connected extension. When the FXO port detects a ring from PBX, the Cisco router sends a VoIP call setup to IP22 but it does not take the FXO port off-hook. As a result, the PBX only sees the call answer signal when IP22 FXS port is picked up. In the same way, IP22 must forward outgoing calls to FXO gateway.
CISCO CONFIGURATION: (network configurations is omitted)
FXO Voice port, command line configuration:
voice-port 2/0 input gain -6 output attenuation 6 no non-linear no vad playout-delay nominal 100 playout-delay mode fixed cptone IT timeouts call-disconnect 1 timeouts wait-release 1 timing guard-out 300 connection plar opx 102 <--- incoming calls from PBX generate calls to phone number 102; Cisco SIP dial-peer number 2 is matched
! voice-port 2/1 input gain -6 output attenuation 6 no non-linear no vad playout-delay nominal 100 playout-delay mode fixed cptone IT timeouts call-disconnect 1 timeouts wait-release 1 timing guard-out 300 connection plar opx 103 <--- incoming calls from PBX generate calls to phone number 103; Cisco SIP dial-peer number 2 is matched
Dial Peer, command line configuration:
dial-peer voice 2 voip description SIP_CALL_TO_IP22 destination-pattern 1.. modem passthrough nse codec g711alaw session protocol sipv2 session target ipv4: __IP22__IP_ADDRESS__ session transport udp incoming called-number . dtmf-relay rtp-nte codec g711alaw fax rate disable fax protocol pass-through g711alaw no vad ! dial-peer voice 15 pots description FROM_IP22_TEL3 destination-pattern 203 port 2/1 ! dial-peer voice 16 pots description FROM_IP22_TEL2 destination-pattern 202 port 2/0
SIP UA configuration:
sip-ua disable-early-media 180 max-forwards 10 retry invite 3 timers expires 60000 no suspend-resume