[cisco-voip] CCM 3.3 to SIP
Mark Koci
mkoci at ideacomtech.com
Fri Apr 7 17:47:01 EDT 2006
Has anyone tried this configuration. My company wants to move to a
SIP provider, but we're running CCM 3.3. I thought "no problem" we'll
just get an ip2ip gateway, have the CCM come in on a H323 trunk to the
gateway and convert it to SIP. Now I have a problem, I cannot get this
to work. The most I can get is a SIP softphone signaling a Cisco phone,
but no audio. The setup is 7960 <-> CCM 3.3 < H225 Trunk > 2651XM ip2ip
gateway < SIP Trunk > Asterisk <-> SJphone. This is the first time I've
worked with either SIP or Gatekeepers. The CCM has a H225 Trunk with
the IP address of the gateway as 172.16.1.17 and the gatekeeper of
192.168.1.100. I've tried using MTP and not using on the Trunk. Below
is the router configuration, error message from SIP phone to Cisco
phone, and error message from Cisco to SIP phone.
When I make a call from the SIP phone the Cisco phone rings and I can
see the Caller ID from the SIP phone, but when I go offhook on the Cisco
the SIP phone keeps ring. When I put the Cisco onhook the SIP phone
comes back with "all circuits are busy". I haven't been able to get
the Cisco phone to ring the SIP phone.
I can call out to the PSTN via the FXO port from both phones so "I
think" the SIP trunk and H225 trunk are configured correctly. Thanks in
advance.
Mark
This is the configuration on the 2651XM:
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
!
!
no aaa new-model
!
resource policy
!
memory-size iomem 10
clock timezone EDT -5
clock summer-time EDT recurring
no network-clock-participate slot 1
no network-clock-participate wic 0
ip subnet-zero
ip cef
!
!
!
!
no ip domain lookup
!
!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
sip
!!
!
!
!
!
!
!
!
!
!
!
!
!
!
interface FastEthernet0/0
ip address 172.16.1.17 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip id KXTNGK01 ipaddr 192.168.1.100 1719
h323-gateway voip tech-prefix 1#
h323-gateway voip bind srcaddr 172.16.1.17
!
interface FastEthernet0/1
ip address 192.168.1.100 255.255.255.0
duplex auto
speed auto
!
ip classless
!
ip http server
!
!
!
control-plane
!
!
!
voice-port 1/0/0
!
voice-port 1/0/1
!
voice-port 1/1/0
!
voice-port 1/1/1
!
!
!
!
!
dial-peer voice 1 pots
incoming called-number .
!
dial-peer voice 9 pots
destination-pattern 9
port 1/1/0
!
dial-peer voice 3149 voip
destination-pattern 615583....
session protocol sipv2
session target ras
session transport tcp
dtmf-relay rtp-nte
codec g711ulaw
no vad
! dial-peer voice 8350 voip
destination-pattern 865766....
session target ras
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
gateway
timer receive-rtp 1200
!
sip-ua
sip-server ipv4:192.168.1.212
!
!
gatekeeper
zone local KXTNGK01 ipsbs.net 192.168.1.100 outvia KXTNGK01
zone prefix KXTNGK01 615583*
zone prefix KXTNGK01 865766*
gw-type-prefix 1#* default-technology
no shutdown
!
!
line con 0
line aux 0
line vty 0 4
login
!
!
end
This is the error message I get when I call from the SIP phone to the
Cisco phone:
*Mar 7 22:22:10.517: //50/A786B061801A/H323/cch323_get_caps_chn_info:
SIP->H323 xcoding, try to get xtream resources
*Mar 7 22:22:10.517: //50/A786B061801A/H323/cch323_get_peer_info: Entry
*Mar 7 22:22:10.517: //50/A786B061801A/H323/cch323_get_peer_info: Have
peer
*Mar 7 22:22:10.517: //50/A786B061801A/H323/cch323_set_pref_codec_list:
Peer channel present: dp pref mask=1
*Mar 7 22:22:10.517: //50/A786B061801A/H323/cch323_set_pref_codec_list:
First preferred codec(bytes)=5(160)
*Mar 7 22:22:10.517: //50/A786B061801A/H323/cch323_get_peer_info: Flow
Mode set to FLOW_THROUGH
*Mar 7
22:22:10.517: //50/A786B061801A/H323/cch323_set_h245_state_mc_mode_outgoing: call_spi_mode = 3
*Mar 7
22:22:10.517: //50/A786B061801A/H323/cch323_set_h245_state_mc_mode_outgoing: h245 state m/c mode=0x4F0, h323_ctl=0x2F
*Mar 7
22:22:10.521: //50/A786B061801A/H323/cch323_set_h323_control_options_outgoing: h245 sm mode = 1264
*Mar 7
22:22:10.521: //50/A786B061801A/H323/cch323_set_h323_control_options_outgoing: h323_ctl=0x20
*Mar 7 22:22:10.537: //50/A786B061801A/H323/cch323_rotary_validate: No
peer_ccb available
*Mar 7
22:22:10.541: //50/A786B061801A/H323/cch323_build_local_encoded_fastStartOLCs: state_mc_mode=0x4F0 on outbound leg
*Mar 7
22:22:10.541: //50/A786B061801A/H323/cch323_build_local_encoded_fastStartOLCs: srcAddress = 0xAC100111, h245_lport = 0, flow mode = 1, minimum_qos=0
*Mar 7
22:22:10.541: //50/A786B061801A/H323/cch323_generic_open_logical_channel: current codec = 5:160:160
*Mar 7 22:22:16.863: //50/A786B061801A/H323/h245_address_ind: ev=4
*Mar 7 22:22:16.863: //-1/xxxxxxxxxxxx/H323/h245_address_ind: Sending
event
*Mar 7 22:22:16.863: //50/A786B061801A/H323/cch323_h245_addr_notify:
Sending event CC_EV_H245_ADDR, ev 4
*Mar 7 22:22:16.863: //50/A786B061801A/H323/cch323_h245_addr_notify:
Unable to send event CC_EV_H245_ADDR
This is the error message when I call from the Cisco phone to the SIP
phone:
*Mar 7 22:27:03.497: //51/00606C290F00/H323/setup_ind: Receive bearer
cap infoXRate 16, rateMult 0
*Mar 7
22:27:03.505: //51/00606C290F00/H323/cch323_set_h245_state_mc_mode_incoming: h245 state m/c mode=0x10F, h323_ctl=0x0
*Mar 7 22:27:03.521: //-1/xxxxxxxxxxxx/H323/cch245_event_handler:
callID=51
*Mar 7 22:27:03.521: //-1/xxxxxxxxxxxx/H323/cch245_event_handler: Event
CC_EV_H245_SET_MODE: data ptr=0x83D5A388
*Mar 7 22:27:03.521: //-1/xxxxxxxxxxxx/H323/cch323_set_mode: callID=51,
flow Mode=1 spi_mode=0x3
*Mar 7 22:27:03.525: //51/00606C290F00/H323/cch323_do_call_proceeding:
set_mode NOT called yet...saved deferred CALL_PROC
*Mar 7 22:27:03.525: //51/00606C290F00/H323/cch323_process_set_mode:
Setting inbound leg mode flags to 0x4F0, flow-mode to FLOW_THROUGH
*Mar 7 22:27:03.525: //51/00606C290F00/H323/cch323_process_set_mode:
Sending deferred CALL_PROC
*Mar 7 22:27:03.525: //51/00606C290F00/H323/cch323_do_call_proceeding:
set_mode called so we can proceed with CALLPROC
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