[cisco-voip] H323 GW configuration

Kin Wai kinwai at singnet.com.sg
Fri Feb 19 04:28:54 EST 2010


It's hitting dial peer 20 but not going anywhere. It's my first time doing
GW/GK configuration, therefore I'm not sure whether it's normal behavior or
not. 

So far, I got it working by either 

1)      using "num-exp 651.. 1.." 

2)      in ephone-dn , number 117 secondary 65117

 

Is this the correct way? Or there should be a better way to do it? I tried
to apply the translation-profile in dial-peer 20, to strip away the tech
prefix "65" but it's not working somehow. 

 

Below is the output when it's not working. 

005967: *Feb 19 17:14:25.894 SGP:
//-1/00040F00C420/DPM/dpAssociateIncomingPeerCore:

   Calling Number=, Called Number=65117, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

005968: *Feb 19 17:14:25.898 SGP:
//-1/00040F00C420/DPM/dpAssociateIncomingPeerCore:

   Result=NO_MATCH(-1) After All Match Rules Attempt

005969: *Feb 19 17:14:25.898 SGP:
//-1/00040F00C420/DPM/dpAssociateIncomingPeerCore:

   Calling Number=, Called Number=65117, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

005970: *Feb 19 17:14:25.898 SGP:
//-1/00040F00C420/DPM/dpAssociateIncomingPeerCore:

   Result=NO_MATCH(-1) After All Match Rules Attempt

005971: *Feb 19 17:14:25.910 SGP:
//-1/00040F00C420/DPM/dpAssociateIncomingPeerCore:

   Calling Number=, Called Number=65117, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

005972: *Feb 19 17:14:25.910 SGP:
//-1/00040F00C420/DPM/dpAssociateIncomingPeerCore:

   Result=NO_MATCH(-1) After All Match Rules Attempt

005973: *Feb 19 17:14:25.914 SGP: //-1/00040F00C420/DPM/dpMatchPeersCore:

   Calling Number=, Called Number=65117, Peer Info Type=DIALPEER_INFO_SPEECH

005974: *Feb 19 17:14:25.914 SGP: //-1/00040F00C420/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=65117

005975: *Feb 19 17:14:25.914 SGP: //-1/00040F00C420/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

005976: *Feb 19 17:14:25.914 SGP: //-1/00040F00C420/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0) 

   List of Matched Outgoing Dial-peer(s): 

     1: Dial-peer Tag=20

005977: *Feb 19 17:14:25.914 SGP: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number=65117, Called Number=65117, Peer Info
Type=DIALPEER_INFO_SPEECH

005978: *Feb 19 17:14:25.914 SGP: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=65117

005979: *Feb 19 17:14:25.914 SGP: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

005980: *Feb 19 17:14:25.914 SGP: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0) 

   List of Matched Outgoing Dial-peer(s): 

     1: Dial-peer Tag=20

005981: *Feb 19 17:14:25.914 SGP: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number=65117, Called Number=65117, Peer Info
Type=DIALPEER_INFO_SPEECH

005982: *Feb 19 17:14:25.914 SGP: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=65117

005983: *Feb 19 17:14:25.914 SGP: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

005984: *Feb 19 17:14:25.914 SGP: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0) 

   List of Matched Outgoing Dial-peer(s): 

     1: Dial-peer Tag=20

005985: *Feb 19 17:14:25.914 SGP:
//-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Calling Number=65117, Called Number=, Voice-Interface=0x0,

   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search
Type=PEER_TYPE_VOICE,

   Peer Info Type=DIALPEER_INFO_SPEECH

005986: *Feb 19 17:14:25.914 SGP:
//-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:

   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=20

005987: *Feb 19 17:14:25.918 SGP: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Calling Number=, Called Number=65117, Peer Info Type=DIALPEER_INFO_SPEECH

005988: *Feb 19 17:14:25.918 SGP: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=65117

005989: *Feb 19 17:14:25.918 SGP: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

005990: *Feb 19 17:14:25.918 SGP: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0) 

   List of Matched Outgoing Dial-peer(s): 

     1: Dial-peer Tag=20

005991: *Feb 19 17:14:25.918 SGP: //-1/00040F00C420/DPM/dpMatchPeersCore:

   Calling Number=, Called Number=65117, Peer Info Type=DIALPEER_INFO_SPEECH

005992: *Feb 19 17:14:25.918 SGP: //-1/00040F00C420/DPM/dpMatchPeersCore:

   Match Rule=DP_MATCH_DEST; Called Number=65117

005993: *Feb 19 17:14:25.918 SGP: //-1/00040F00C420/DPM/dpMatchPeersCore:

   Result=Success(0) after DP_MATCH_DEST

005994: *Feb 19 17:14:25.918 SGP: //-1/00040F00C420/DPM/dpMatchPeersMoreArg:

   Result=SUCCESS(0) 

   List of Matched Outgoing Dial-peer(s): 

     1: Dial-peer Tag=20

 

 

When it's working, using either of the method stated above, I encountered a
new problem, no call progress tone. 

Debug dial-peer shows that it's going into dial-peer 200XX (those ephones),
is there any way to get call progress tones?

 

005798: *Feb 19 17:07:21.842 SGP:
//96/00040F00C41D/H323/cch323_process_set_mode: Setting inbound leg mode
flags to 0x10F, flow-mode to FLOW_THROUGH

005799: *Feb 19 17:07:21.842 SGP:
//96/00040F00C41D/H323/cch323_process_set_mode: Sending deferred CALL_PROC

005800: *Feb 19 17:07:21.842 SGP:
//96/00040F00C41D/H323/cch323_do_call_proceeding: gw_id=1

005801: *Feb 19 17:07:21.842 SGP:
//96/00040F00C41D/H323/cch323_do_call_proceeding: set_mode called so we can
proceed with CALLPROC

005802: *Feb 19 17:07:21.842 SGP: //96/00040F00C41D/H323/run_h225_sm:
Received event H225_EV_CALLPROC while at state H225_REQ_FS_SETUP

005803: *Feb 19 17:07:21.842 SGP:
//96/00040F00C41D/H323/cch323_h225_set_new_state: Changing from
H225_REQ_FS_SETUP state to H225_ACC_FS_CALLPROC state

005804: *Feb 19 17:07:21.842 SGP:
//96/00040F00C41D/H323/generic_send_callproc: ====== PI = 0

.

..

005834: *Feb 19 17:07:21.854 SGP:
//-1/xxxxxxxxxxxx/H323/cch323_iev_queue_service: Dispatch 0x1 internal event
to H225 SM

005835: *Feb 19 17:07:21.854 SGP: //96/00040F00C41D/H323/run_h225_sm:
Received event H225_EV_ALERT while at state H225_ACC_FS_CALLPROC

005836: *Feb 19 17:07:21.854 SGP:
//96/00040F00C41D/H323/cch323_h225_set_new_state: Changing from
H225_ACC_FS_CALLPROC state to H225_ACC_FS_ALERT state

005837: *Feb 19 17:07:21.854 SGP: //96/00040F00C41D/H323/generic_send_alert:
====== PI = 0

005838: *Feb 19 17:07:21.854 SGP:
//96/00040F00C41D/H323/cch323_get_embedded_obj_from_ccb: ccb=0x47A61E74,
tag=17, size=83

005839: *Feb 19 17:07:21.854 SGP:
//96/00040F00C41D/H323/cch323_get_embedded_obj_from_ccb: Extraction PASSED
from 0x4B852B50

 

Regards, 

Kin Wai

 

 

From: Kevin Thorngren [mailto:kthorngr at cisco.com] 
Sent: Friday, February 19, 2010 6:01 AM
To: Kin Wai
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] H323 GW configuration

 

Maybe try "incoming called-number 65T" under dial peer 20 with your
translation-rule.  May the incoming call is hitting a different dial peer.

 

Try looking at "debug voip dialpeer" to see what is happening with the call.

 

Kevin

On Feb 18, 2010, at 3:38 PM, Kin Wai wrote:





Hi,

I'm have a gateway registered to Gatekeeper.

 

Currently,

Local call to remote sites through the GK --- OK

Remote sites to PSTN through local gateway ---- OK (using the tech-prefix)

Remote sites to Local Phone ----- NOT OK (using the same tech-prefix)




I have the following configuration :

dial-peer voice 20 voip

 destination-pattern 6.T

 session target ras

!

dial-peer voice 21 pots

 destination-pattern 65.T

 port 0/1/0:15

 

*I have excluded the h323-gateway commands*

The tech prefix for this site is 65

 

What do I need to do in order for the remote site to call the local phones?
The gatekeeper is a 3rd party gatekeeper.

I suppose there's no issue with the communications as remote sites user can
make outgoing call to PSTN.  

 

Do I need to enable "allow connections h323 to h323" under voice service
voip ? Or I need to create an additional dial-peer to cater for the local
phones?

 

I tried to apply translation-rule to strip away the tech-prefix or even make
it the full DID number under dial-peer 20, but it still won't work.

 

Anyone have any clue about this problem?

 

Thanks in advance!

 

Regards,

Kin Wai

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