I've got next schema:
PSTN - ISP - (SIP trunk) - AS5350XM - (H.323) - CUCM8.0 - (SCCP) - IP Phone
AS
There is a problem with incoming call from PSTN.
This is how the progress of the call looks like:
Call comes to GW
GW sends it to CUCM. CUCM translate number to internal.
IP Phone rings.
I hook up the handset and hear nothing. Packet capture shows that some udp traffic goes between the phone and the GW.
134 1.174445000 10.20.8.46 10.20.8.10 UDP 214 Source port: 31926 Destination port: 17774
At the same time on the calling side I continue to hear ring-back.
After few seconds call drops.
H.245 Event Debug after hooking up shows next:
hq_as5350xm_gw1#changing from CONNECTING state to CONNECTED state
Sep 25 17:08:16.145: h323chan_chn_process_read_socket: fd=3 of type CONNECTED has data
Sep 25 17:08:16.145: h323chan_recvdata: No Data on fd=3
PROCESS_READ: FAILED/NOT COMPLETE,rc 10, fd=3
After some timeout and droping:
Sep 25 17:11:50.478: h323chan_chn_process_read_socket: fd=2 of type CONNECTED has data
Sep 25 17:11:50.482: h323chan_close: TCP connection from fd=2 closed
Sep 25 17:11:50.482: h323chan_close: TCP connection from fd=3 closed
And ccsip messages debug respectively:
Sent:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 193.***.***.145:5060;branch=z9hG4bKwqpxz8y1cxyvyutt7v7pmq7vx
From: <sip:4436****1@62.***.***.26;user=phone>;tag=sbc0407q77put8w-CC-22
To: <sip:322*****4@62.***.***.266;user=phone>;tag=8C2BA830-20B0
Date: Thu, 25 Sep 2014 17:11:38 GMT
Call-ID: isbcv1pz8by8ym8pwyuqxpmbqhmqhxph88px@SoftX3000
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=47
Content-Length: 0
Sep 25 17:11:50.490: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
hq_as5350xm_gw1#ACK sip:322*****4@62.***.***.26:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 193.***.***.145:5060;branch=z9hG4bKwqpxz8y1cxyvyutt7v7pmq7vx
Call-ID: isbcv1pz8by8ym8pwyuqxpmbqhmqhxph88px@SoftX3000
From: <sip:4436****1@62.***.***.26;user=phone>;tag=sbc0407q77put8w-CC-22
To: <sip:322*****4@62.***.***.26;user=phone>;tag=8C2BA830-20B0
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
Incoming dial-peer:
dial-peer voice 43 voip
translation-profile outgoing LV_KS-SIP_out
session protocol sipv2
session target ipv4:193.***.***.145
incoming called-number 322*****4
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1.202
voice-class sip bind media source-interface GigabitEthernet0/1.202
Outgoing Dial-peer
dial-peer voice 103 voip
destination-pattern 322*****4
session target ipv4:10.20.8.6
voice-class codec 1
Interface to ISP
interface GigabitEthernet0/1.202
encapsulation dot1Q 202
ip address 62.***.***.26 255.255.255.248
Interface to CUCM
interface GigabitEthernet0/0
ip address 10.20.8.10 255.255.255.0
duplex auto
speed auto
negotiation auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 10.20.8.10
voice service voip
ip address trusted list
ipv4 10.0.0.3
ipv4 10.0.0.4
ipv4 193.***.***.145
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol pass-through g711ulaw
sip
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8
As I understand I missed something in theory about SIP and H.323 interoperability. Please, help me to find my gap.
