Route outgoing PSTN call to the remote gateway

[O.Zang]

Hi All,

Please Helps solve this case.

I am deploying a multiple site collaboration solution.

I have created route group for all sites. 

for Biling purposes I have to route all outgoing PSTN call to the remote gateway on the HQ.

Ip phones using the HQ gateway and it's Device pool are able to call out to the pstn.

But ip phones on other site can't call PSTN numbers.  

I have configured a dialpeer to route PSTN from others site to The HQ gateway but still can't place a call.

Below are the Dialpeer that  created:

dial-peer voice 30 voip

description "outgoing call from HQ Premary gateway "
preference 1
destination-pattern .T
session target ipv4:10.163.0.40
voice-class codec 1
dtmf-relay h245-alphanumeric
no vad
!
dial-peer voice 1 pots
trunkgroup Hunt_Outgoing
description "outgoing call in srst mode "
preference 3
destination-pattern .T
progress_ind setup enable 3
progress_ind progress enable 8
progress_ind connect enable 8
forward-digits all
!
dial-peer voice 31 voip

description "outgoing call from HQ secondary gateway "

preference 2
destination-pattern .T
session target ipv4:10.163.0.41
voice-class codec 1
dtmf-relay h245-alphanumeric
no vad

Thanks in advance for your help. 

[Jaime Valencia]

what's the detailed call flow you want?

what do the debug say when you try with your current config?

[Dennis Mink]

on the gateway do a debug voip dial-peer to see what dial peer gets hits. also add debug ccsip messages  and state the calling and called numbers

[O.Zang]

Hi Java and Denis

Thank for replying,

Here is the call Flow that I want:

branch site: Trechville-av8

Device Pool: DP_treichville-av8

Route group: LRG_treichville ( I have configure SLRG)

IP Phone DN: 800712

got h323 Gateway  with FXO port ( FXO to be use only in case of faillure.

Main Site: ( outgoing calls  to PSTN are working fine)

DP: DP_SIEGE

route group: LRG-BACKUP-SIEGE

got a H323 gateway with E1 card.

 calls from Ip phone on branch site ( 800712) to PSTN number 58487437 need to be handled by the gateway on the main site.

Below are the Dialpeer that  created:

dial-peer voice 30 voip

description "outgoing call from HQ Premary gateway "
preference 1
destination-pattern .T
session target ipv4:10.163.0.40
voice-class codec 1
dtmf-relay h245-alphanumeric
no vad
!
dial-peer voice 1 pots
trunkgroup Hunt_Outgoing
description "outgoing call in srst mode "
preference 3
destination-pattern .T
progress_ind setup enable 3
progress_ind progress enable 8
progress_ind connect enable 8
forward-digits all
!
dial-peer voice 31 voip

description "outgoing call from HQ secondary gateway "

preference 2
destination-pattern .T
session target ipv4:10.163.0.41
voice-class codec 1
dtmf-relay h245-alphanumeric
no vad

Below are the debugs output:

calling Number: 800712, called number 58487437

Debug voip dialpeer: 

GW_TREICHVILLE-AV8#debug voip dialpeer
voip dialpeer default debugging is on
GW_TREICHVILLE-AV8#
GW_TREICHVILLE-AV8#
GW_TREICHVILLE-AV8#
GW_TREICHVILLE-AV8#
*Jun 19 12:01:21.147: //-1/50B3052D81A4/DPM/dpAssociateIncomingPeerCore:
Calling Number=800712, Called Number=800712, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Jun 19 12:01:21.147: //-1/50B3052D81A4/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=30
*Jun 19 12:01:21.147: //-1/50B3052D81A4/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0
*Jun 19 12:01:34.742: //-1/80695E610300/DPM/dpAssociateIncomingPeerCore:
Calling Number=800712, Called Number=58487437, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Jun 19 12:01:34.742: //-1/80695E610300/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=30
*Jun 19 12:01:34.742: //-1/80695E610300/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0
*Jun 19 12:01:34.742: //-1/80695E610300/DPM/dpAssociateIncomingPeerCore:
Calling Number=800712, Called Number=58487437, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*Jun 19 12:01:34.742: //-1/80695E610300/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=30
*Jun 19 12:01:34.742: //-1/80695E610300/DPM/dpMatchSafModulePlugin:
dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0

Debug ccsip messages

*Jun 19 11:57:20.313: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
REGISTER sip:10.163.7.161 SIP/2.0
Via: SIP/2.0/UDP 10.163.7.164:5060;branch=z9hG4bK2a824c12
From: <sip:800712@10.163.7.161>;tag=0076861bfd520037375f67a9-3e670c82
To: <sip:800712@10.163.7.161>
Call-ID: 0076861b-fd52000c-47a94125-11ac892f@10.163.7.164
Max-Forwards: 70
Session-ID: b06c60db00105000a0000076861bfd52;remote=00000000000000000000000000000000
Date: Tue, 19 Jun 2018 11:56:24 GMT
CSeq: 107 REGISTER
User-Agent: Cisco-CP7821/11.7.1
Contact: <sip:a1600c03-9ea1-908d-64b1-eda78300102e@10.163.7.164:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0076861bfd52>";+u.sip!devicename.ccm.cisco.com="SEP0076861BFD52";+u.sip!model.ccm.cisco.com="621"
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Content-Length: 0
Expires: 0

*Jun 19 11:57:20.315: //3326/C126F8CF81A0/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.163.7.164:5060;branch=z9hG4bK2a824c12
From: <sip:800712@10.163.7.161>;tag=0076861bfd520037375f67a9-3e670c82
To: <sip:800712@10.163.7.161>;tag=1F3F53E0-1383
Date: Tue, 19 Jun 2018 11:57:20 GMT
Call-ID: 0076861b-fd52000c-47a94125-11ac892f@10.163.7.164
Server: Cisco-SIPGateway/IOS-15.5.3.S4b
CSeq: 107 REGISTER
Content-Length: 0

[R0g22]

You don't need SIP debugs if the gateway is H.323. Enable the following and collect the logs -

debug h225 asn1
debug h245 asn1
debug voice ccapi inout
debug isdn q931
debug voip translation
debug voip dialpeer

[O.Zang · Accepted solution]

Hi Nipun Singh Raghav,

I want let you know that I have solved the issue.

The customer's requirement is to route call outgoing to pstn throught the gateway on the main site.

So i just use the local route group of the main site in the  device pool configuration of the branch site.

thank you for your Help Nipun, Java and Dennis