IP2IP Gateway-Asterisk Integration

Unanswered Question
Oct 15th, 2009

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

I have this problem too.
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jojatopi27 Thu, 10/15/2009 - 14:33

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:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.137.138:5060;branch=z9hG4bK561f6972;rport

From: "Unknown" ;tag=as2e4a1809

To:

Contact:

Call-ID: [email protected]

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" ;tag=as2e4a1809

To:

Date: Wed, 14 Oct 2009 19:39:30 GMT

Call-ID: [email protected]

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" ;tag=as2e4a1809

To: ;tag=AEFD5910-B45

Date: Wed, 14 Oct 2009 19:39:30 GMT

Call-ID: [email protected]

CSeq: 102 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: ;party=called;screen=no;privacy=off

Contact:

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" ;tag=as2e4a1809

To: ;tag=AEFD5910-B45

Date: Wed, 14 Oct 2009 19:39:30 GMT

Call-ID: [email protected]

CSeq: 102 INVITE

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Remote-Party-ID: ;party=called;screen=no;privacy=off

Contact:

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" ;tag=as2e4a1809

To: ;tag=AEFD5910-B45

Date: Wed, 14 Oct 2009 19:39:30 GMT

Call-ID: [email protected]

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:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.137.138:5060;branch=z9hG4bK561f6972;rport

From: "Unknown" ;tag=as2e4a1809

To: ;tag=AEFD5910-B45

Contact:

Call-ID: [email protected]

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0

Nicholas Matthews Thu, 10/15/2009 - 15:38

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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