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One way audio with FXS and STCAPP
evanr
Member Posts: 5 ■□□□□□□□□□
Hello everyone, I hope maybe someone can help me out.
I have a cordless phone connected to my 3725 on FXS port 1/1/1.
System image file is "slot0:c3725-adventerprisek9-mz.124-12.bin"
PROBLEM: I call my cordless from my cell. Cellphone can hear voice, but callee on the cordless phone can not hear anything.
What could be the problem here?
stcapp ccm-group 1
stcapp
!
!
!
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
h323
emptycapability
ip circuit max-calls 10
session transport udp
sip
registrar server expires max 3600 min 600
!
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g723r63
codec preference 5 clear-channel
!
voice-port 1/1/1
timeouts interdigit 12
timing hookflash-in 50 0
description LEFT FXS PORT
station-id name HOMEPHONE
caller-id enable
!
ccm-manager config server 172.16.1.1
ccm-manager config
!
sccp local FastEthernet0/0
sccp ccm 172.16.1.1 identifier 1 priority 1
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
!
dial-peer voice 6 voip
description ## INBOUND DID ##
translation-profile incoming INC
voice-class codec 1
session protocol sipv2
session target dns:callcentric.com
session transport udp
incoming called-number <snip>
dtmf-relay cisco-rtp rtp-nte digit-drop h245-alphanumeric
no vad
!
dial-peer voice 100 pots
service stcapp
port 1/1/1
forward-digits 0
!
dial-peer voice 2 voip
description ### LD USA ###
translation-profile outgoing OUT
destination-pattern 1[2-8]T
voice-class codec 1
session protocol sipv2
session target dns:callcentric.com
session transport udp
dtmf-relay cisco-rtp rtp-nte digit-drop h245-alphanumeric
!
dial-peer voice 3 voip
description ## ASTERISK ##
destination-pattern 2T
session protocol sipv2
session target ipv4:192.168.200.2
session transport udp
dtmf-relay cisco-rtp rtp-nte digit-drop h245-alphanumeric
codec g711ulaw
!
sip-ua
authentication username <snip>
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 5
timers register 250
registrar dns:callcentric.com expires 3600
sip-server dns:callcentric.com
telephony-service
max-ephones 2
max-dn 2
ip source-address 172.16.1.1 port 2000
no caller-id name-only
calling-number initiator
time-zone 8
create cnf-files version-stamp 7960 Oct 24 2013 20:19:21
voicemail 202
max-conferences 4 gain -6
transfer-system full-blind
transfer-pattern .T
secondary-dialtone 9
!
!
ephone-dn 1
number 100 secondary <sip#> no-reg primary
label Home Telephone
call-forward busy 202
call-forward noan 202 timeout 30
!
!
ephone 1
mac-address D295.021E.0281
type anl
button 1:1
I have a cordless phone connected to my 3725 on FXS port 1/1/1.
System image file is "slot0:c3725-adventerprisek9-mz.124-12.bin"
PROBLEM: I call my cordless from my cell. Cellphone can hear voice, but callee on the cordless phone can not hear anything.
What could be the problem here?
stcapp ccm-group 1
stcapp
!
!
!
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
h323
emptycapability
ip circuit max-calls 10
session transport udp
sip
registrar server expires max 3600 min 600
!
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g723r63
codec preference 5 clear-channel
!
voice-port 1/1/1
timeouts interdigit 12
timing hookflash-in 50 0
description LEFT FXS PORT
station-id name HOMEPHONE
caller-id enable
!
ccm-manager config server 172.16.1.1
ccm-manager config
!
sccp local FastEthernet0/0
sccp ccm 172.16.1.1 identifier 1 priority 1
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
!
dial-peer voice 6 voip
description ## INBOUND DID ##
translation-profile incoming INC
voice-class codec 1
session protocol sipv2
session target dns:callcentric.com
session transport udp
incoming called-number <snip>
dtmf-relay cisco-rtp rtp-nte digit-drop h245-alphanumeric
no vad
!
dial-peer voice 100 pots
service stcapp
port 1/1/1
forward-digits 0
!
dial-peer voice 2 voip
description ### LD USA ###
translation-profile outgoing OUT
destination-pattern 1[2-8]T
voice-class codec 1
session protocol sipv2
session target dns:callcentric.com
session transport udp
dtmf-relay cisco-rtp rtp-nte digit-drop h245-alphanumeric
!
dial-peer voice 3 voip
description ## ASTERISK ##
destination-pattern 2T
session protocol sipv2
session target ipv4:192.168.200.2
session transport udp
dtmf-relay cisco-rtp rtp-nte digit-drop h245-alphanumeric
codec g711ulaw
!
sip-ua
authentication username <snip>
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 5
timers register 250
registrar dns:callcentric.com expires 3600
sip-server dns:callcentric.com
telephony-service
max-ephones 2
max-dn 2
ip source-address 172.16.1.1 port 2000
no caller-id name-only
calling-number initiator
time-zone 8
create cnf-files version-stamp 7960 Oct 24 2013 20:19:21
voicemail 202
max-conferences 4 gain -6
transfer-system full-blind
transfer-pattern .T
secondary-dialtone 9
!
!
ephone-dn 1
number 100 secondary <sip#> no-reg primary
label Home Telephone
call-forward busy 202
call-forward noan 202 timeout 30
!
!
ephone 1
mac-address D295.021E.0281
type anl
button 1:1
Comments
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Optionsevanr
Member Posts: 5 ■□□□□□□□□□
On 2811 seems to work fine. Waiting for additional memory for 3725 to see if an IOS upgrade solves these problems.
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mistabrumley89
Member Posts: 356 ■■■□□□□□□□
Just looking at this makes my head hurt and makes me afraid to even start CCNA Voice lolGoals: WGU BS: IT-Sec (DONE) | CCIE Written: In Progress
LinkedIn: www.linkedin.com/in/charlesbrumley -
Optionsevanr
Member Posts: 5 ■□□□□□□□□□
Heh. I know. I welcome the challenge, and honestly Voice a lot of fun because you get to directly see [hear] the product of your work! Ringing phones! Also keep in mind I have a lot of stuff in this config that is probably NOT on the CCNA voice, mostly SIP trunk stuff. I'm just starting CCNA Voice but so far I have heard no mention of SIP trunking.