CME 4.1 SIP Trunk No Speech Path

[DogBiscuits]

Hello,

I have just configured a sip Trunk to callcentric.com. I am able to place calls inbound and out but there is no audio.

I think this is part of the problem.

"CME2600#show voip rtp connection
VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 718 719 19472 63918 192.168.205.251 204.11.192.38
Found 1 active RTP connections"

When I trace the RTP connection it shows the IP address of my CME instead of the IP address of my telephone.

Maybe that is normal and I am barking up the wrong tree.

If anyone has any ideas why I am getting no speech path please let me know.

Here is my running-config:

sh run
Building configuration...

Current configuration : 3685 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname CME2600
!
boot-start-marker
boot-end-marker
!
enable secret 5 $1$xyMV$8raMACzO7KYdPZwUk/ini0
enable password 7 12090404011C03162E
!
no aaa new-model
clock timezone est -5
clock summer-time est recurring
clock save interval 8
no network-clock-participate slot 1
no network-clock-participate wic 0
no ip routing
!
!
!
!
ip name-server 192.168.205.1
!
multilink bundle-name authenticated
!
!
!
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
signaling forward unconditional
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
!
!
!
!
!
!
!
!
!
!
!
voice register global
max-dn 72
max-pool 24
!
!
voice translation-rule 1
rule 1 /^9/ //
!
voice translation-rule 2
rule 1 /228/ /17771234567/
!
voice translation-profile OUT
translate calling 2
translate called 1
!
!
!
!
!
!
archive
log config
hidekeys
!
!
!
!
!
!
!
!
interface FastEthernet0/0
ip address 192.168.205.251 255.255.255.0
no ip route-cache
speed auto
full-duplex
no mop enabled
!
ip default-gateway 192.168.205.1
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.205.1
!
!
ip http server
no ip http secure-server
!
!
!
!
!
tftp-server flash00405000700.bin
tftp-server flash00405000700.sbn
!
control-plane
!
!

!
!
!
!
dial-peer voice 901 voip
translation-profile outgoing OUT
destination-pattern 9.T
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target dns:callcentric.com
dtmf-relay sip-notify rtp-nte
codec g711ulaw
no vad
!
!
sip-ua
credentials username 17771234567 password 11080A011143595F50 realm callcentric.com
authentication username 17771234567 password 7 06071C254A1F5B4A51 realm callcentric.com
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 10
timers register 250
registrar dns:callcentric.com expires 3600
sip-server dns:callcentric.com
!
!
!
telephony-service
load 7910 P00405000700
max-ephones 10
max-dn 10
ip source-address 192.168.205.251 port 2000
auto assign 1 to 10
time-zone 12
max-conferences 4 gain -6
call-forward pattern .T
transfer-system full-consult
secondary-dialtone 9
server-security-mode non-secure
create cnf-files version-stamp Jan 01 2002 00:00:00

!
ephone-dn 1 dual-line
number 228 secondary 17771234567
label CIPC
!
!
ephone-dn 2 dual-line
number 229
!
!
ephone-dn 3 dual-line
number 230
!
!
ephone-dn 4 dual-line
number 231
!
!
ephone-dn 5 dual-line
number 232
!
!
ephone-dn 6 dual-line
number 233
!
!
ephone-dn 7 dual-line
number 234
!
!
ephone-dn 8 dual-line
number 235
!
!
ephone-dn 9 dual-line
number 236
!
!
ephone-dn 10 dual-line
number 237
!
!
ephone 1
no multicast-moh
device-security-mode none
mac-address 68A3.C44D.1D2A
type CIPC
button 1:1
!
!
!
ephone 2
no multicast-moh
device-security-mode none
!
!
!
ephone 3
no multicast-moh
device-security-mode none
!
!
!
ephone 4
no multicast-moh
device-security-mode none
!
!

ephone 5
no multicast-moh
device-security-mode none
!
!
!
ephone 6
no multicast-moh
device-security-mode none
!
!
!
ephone 7
no multicast-moh
device-security-mode none
!
!
!
ephone 8
no multicast-moh
device-security-mode none
!

!
ephone 9
no multicast-moh
device-security-mode none
!
!
!
ephone 10
no multicast-moh
device-security-mode none
!
!
!
line con 0
line aux 0
line vty 0 4
password 7 08314D5D1A0E0A0516
login
!
!
end

Thanks!

 

[MrNick0483]

To use SIP trunks you will need a CUBE or an Asterisk server for the trunks to terminate to. If you use a CUBE it will need to support SIP inspection to correct your internal to external ip problem. If you use Asterisk you will need to make sure the NAT config is correct.

9 times out of 10 no audio on a SIP trunk call is a NAT issue, as SIP is not a NAT friendly protocol.

 

[DogBiscuits]

I setup Asterisk now with callcentric and registered my CME to the Asterisk system.

I can place calls fine. two-way audio & DTMF work great. However, I am getting 1-way speech on incoming calls.

Here is my current CME config:

sh run
Building configuration...

Current configuration : 2337 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
enable secret 5 $1$8mLW$SmlE0kIdZrAFQ7nTfs6n.1
!
no aaa new-model
no network-clock-participate slot 1
no network-clock-participate wic 0
ip cef
!
!
!
!
!
multilink bundle-name authenticated
!
!
!
voice rtp send-recv
!
!
!
--More-- ######### #########!
!
!
!
!
!
!
!
!
!
voice translation-rule 1
rule 1 /^9/ //
!
voice translation-rule 2
rule 1 /228/ /2000/
!
!
voice translation-profile OUT
translate calling 2
translate called 1
!
!
!
!
!
!
archive
log config
hidekeys
!
!
!
--More-- ######### #########!
!
!
!
!
interface FastEthernet0/0
ip address 192.168.205.251 255.255.255.0
duplex auto
speed auto
!
ip default-gateway 192.168.205.1
ip forward-protocol nd
!
!
ip http server
no ip http secure-server
!
!
!
!
!
!
control-plane
!
!
!
!
!
!
!
dial-peer voice 901 voip
translation-profile outgoing OUT
--More-- ######### ######### destination-pattern 9.T
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:192.168.205.100
dtmf-relay sip-notify rtp-nte
codec g711ulaw
no vad
!
!
sip-ua
authentication username 2000 password 7 050A150B271D1C5A4D
mwi-server ipv4:192.168.205.100 expires 86400 port 5060 transport tcp unsolicited
registrar ipv4:192.168.205.100 expires 3600
sip-server ipv4:192.168.205.100
!
!
!
telephony-service
max-ephones 5
max-dn 5
ip source-address 192.168.205.251 port 2000
auto assign 1 to 5
voicemail 2010
max-conferences 4 gain -6
dn-webedit
time-webedit
transfer-system full-consult
secondary-dialtone 9
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
--More-- ######### #########ephone-dn 1 dual-line
number 228 secondary 2000
!
!
ephone-dn 2 dual-line
number 2001
!
!
ephone-dn 3 dual-line
number 2002
!
!
ephone-dn 4 dual-line
number 2003
!
!
ephone-dn 5 dual-line
number 2004
!
!
ephone 1
no multicast-moh
device-security-mode none
mac-address 0004.27E8.C8C1
type 7910
button 1:1
!
!
!
ephone 2
no multicast-moh
device-security-mode none
--More-- ######### #########!
!
!
ephone 3
no multicast-moh
device-security-mode none
!
!
!
ephone 4
no multicast-moh
device-security-mode none
!
!
!
ephone 5
no multicast-moh
device-security-mode none
!
!
!
line con 0
line aux 0
line vty 0 4
password
login
!
!
end

Router#exit

 

[DogBiscuits]

I found the problem. I needed to disallow all and allow ulaw on the asterisk box for calls coming to CME. I purchased the G.729 codec from Digium and used it to setup my connection to callcentric.com. So I guess it was passing G.729 to CME when I received calls. Once I forced G.711 it seemed to work fine.

 

[DogBiscuits]

Scratch that. It's still not working inbound. callers can hear me but I can't here them.

I registered a sip client to the asterisk box and called the CME and it worked fine. I can call the asterisk extension too and everything works.

I can call outbound from the CME to my cell and everything works fine.

If i call in from my cell to the CME I get 1 way speech. If I call in from my cell to a sip phone registered to asterisk it works fine.

Any ideas?

 

[MrNick0483]

Post a scrubed config of the outbound SIP trunk from the Asterisk server.

 

[DogBiscuits]

Peer Details:
context=from-pstn
fromdomain=callcentric.com
fromuser=1777XXXXXXX
host=callcentric.com
insecure=port,invite
secret=XXXXXX
type=peer
defaultuser=1777XXXXXXX
disallow=all
allow=g729

Register String:
1777XXXXXXX:XXXXXX@callcentric.com/1777XXXXXXX

 

[MrNick0483]

The way i have it sent up with outbound calling i have one trunk tied to an outbond route with the following info:
Outgoing settings

host=XXXXXXXXXXXXXXXX
type=friend
dtmfmode=rtp-nte
allow=ulaw
canreinvite=no
nat=no
insecure=very
qualify=no
reinvite=no
disallow=all
externip=XXXXXXXXXXXXXXX
localhost=0.0.0.0/255.255.255.0

Incoming settings:

User context from-internal

type=peer
context=from-internal
host=CUCM IP ADDRESS
disallow=all
allow=ulaw&alaw
nat=no
canreinvite=no
qualify=yes

My outbound calling would not work without the incoming setting being added to the outbound trunk. The incoming settings are that of the CUCM.