10-15-2009 02:29 PM - edited 03-15-2019 08:05 PM
Hi everybody
I'm trying to integrate a cisco IP-IP gateway with an asterisk. The Gateway is a Cisco 2851 running the image c2800nm-ipvoice_ivs-mz.124-24.T1.bin, and it acts as an H323 gateway for a Call Manager server 6.1.
Between the IP-IP GW and the asterisk the protocol is SIP.
The asterisk is connected to a PBX as a SIP trunk.
If a call is placed from an IP phone registred in the call manager to a phone in the PBX everything works fine. But if a call is placed from PBX to an IP phone, the IP phone rings even if somebody answers the phone, and finally the call is dropped.
I captured the messages in the router by using the command "debug ccsip messages" and the Gateway doesn't send the OK message to the Asterisk when it recevices the call.
If the gateway receives call, I understand that a SIP flow call must have an INVITE, then the the gateway must send TRYING, RINGING and then OK to the Asterisk after the RTP traffic, but the OK is never sent.
Can enyone help me with this problem. I send the relevant configuration of the Gateway, a network graphic and the debug results.
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
fax protocol cisco
h323
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
voice class codec 100
codec preference 1 g729br8
codec preference 2 g729r8
codec preference 3 g711ulaw
codec preference 4 g711alaw
codec preference 5 g723ar53
codec preference 6 g723ar63
codec preference 7 g723r53
codec preference 8 g723r63
translation-rule 100
Rule 1 ^1001 1
Rule 2 ^1002 2
Rule 3 ^1003 3
Rule 4 ^1004 4
Rule 5 ^1004 5
Rule 6 ^1004 6
Rule 7 ^1004 7
Rule 8 ^1004 8
Rule 9 ^1004 9
interface GigabitEthernet0/0
ip address x.x.x.x 255.255.255.248
duplex auto
speed auto
!
interface GigabitEthernet0/1
ip address x.x.x.x 255.255.255.128
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr x.x.x.x
dial-peer voice 7000 voip
description SCN1
destination-pattern 100.....
translate-outgoing called 100
voice-class codec 100
voice-class h323 1
session target ipv4:x.x.x.x
sip-ua
credentials username bcos password 7 091D1C5A4D50 realm default
registrar ipv4:x.x.x.x expires 3600
10-15-2009 02:33 PM
This is the result of the debug command:
Log Buffer (160000 bytes):
*Oct 14 19:39:30.497: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:10011439@10.120.0.22 SIP/2.0
Via: SIP/2.0/UDP 192.168.137.138:5060;branch=z9hG4bK561f6972;rport
From: "Unknown"
To: <10011439>10011439>
Contact:
Call-ID: 3d7c75622f4cdbda25161908750e37b2@192.168.137.138
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 15 Oct 2009 05:41:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 338
v=0
o=root 2516 2516 IN IP4 192.168.137.138
s=session
c=IN IP4 192.168.137.138
t=0 0
m=audio 16246 RTP/AVP 0 8 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
*Oct 14 19:39:30.509: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.137.138:5060;branch=z9hG4bK561f6972;rport
From: "Unknown"
To: <10011439>10011439>
Date: Wed, 14 Oct 2009 19:39:30 GMT
Call-ID: 3d7c75622f4cdbda25161908750e37b2@192.168.137.138
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Oct 14 19:39:30.597: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.137.138:5060;branch=z9hG4bK561f6972;rport
From: "Unknown"
To: <10011439>;tag=AEFD5910-B4510011439>
Date: Wed, 14 Oct 2009 19:39:30 GMT
Call-ID: 3d7c75622f4cdbda25161908750e37b2@192.168.137.138
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <11439>;party=called;screen=no;privacy=off11439>
Contact: <10011439>10011439>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Oct 14 19:39:38.345: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.137.138:5060;branch=z9hG4bK561f6972;rport
From: "Unknown"
To: <10011439>;tag=AEFD5910-B4510011439>
Date: Wed, 14 Oct 2009 19:39:30 GMT
Call-ID: 3d7c75622f4cdbda25161908750e37b2@192.168.137.138
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <11439>;party=called;screen=no;privacy=off11439>
Contact: <10011439>10011439>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Oct 14 19:39:42.153: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 192.168.137.138:5060;branch=z9hG4bK561f6972;rport
From: "Unknown"
To: <10011439>;tag=AEFD5910-B4510011439>
Date: Wed, 14 Oct 2009 19:39:30 GMT
Call-ID: 3d7c75622f4cdbda25161908750e37b2@192.168.137.138
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=16
Content-Length: 0
*Oct 14 19:39:42.197: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:10011439@10.120.0.22 SIP/2.0
Via: SIP/2.0/UDP 192.168.137.138:5060;branch=z9hG4bK561f6972;rport
From: "Unknown"
To: <10011439>;tag=AEFD5910-B4510011439>
Contact:
Call-ID: 3d7c75622f4cdbda25161908750e37b2@192.168.137.138
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
10-15-2009 03:38 PM
You would need to look at the h323 side as well:
debug h225 asn1
debug h245 asn1
debug h225 q931
-nick
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