05-01-2013 12:13 AM - edited 03-16-2019 05:05 PM
Hi all,
In coming Call from SIP trunk through IPTSP is not matching inbound voip dialpeer. so i am unable to play IVR prompt for outside caller.
Outbound calls are working fine.
Topology:
sccp H323 SIP trunk ( using SIP-UA)
IP Phone--------------CUCM--------------VoiceGateway-----------------------------------------SIP IPTSP
Please see the attached debug log file.
debug ccsip mess
debug voip dialpeer all
debug voip ccapi inout
Calling number # 01755521544 ( PSTN nuber)
Called Number# 09610997929 but IPTSP is sending 9610997929 (without 0)
Voice Gateway WAN IP#10.10.10.142
IPTSP Server IP#10.10.20.236
VoiceGateway IOS version: c2800nm-spservicesk9-mz.124-22.T.bin
Please help to solve this issue.
-Thanks
05-01-2013 04:04 AM
By an IVR prompt do you mean a TCL script on the router or an application external to the router? The call does match incoming dial-peer 2365 on the incoming called-number:
*May 1 06:32:29.275: //-1/BB0B78D782B1/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_INCOMING_DNIS; Called Number=9610997929
*May 1 06:32:29.275: //-1/BB0B78D782B1/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*May 1 06:32:29.275: //-1/BB0B78D782B1/DPM/dpMatchCore:
Dial String=9610997929, Expanded String=9610997929, Calling Number=
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*May 1 06:32:29.275: //-1/BB0B78D782B1/DPM/MatchNextPeer:
Result=Success(0); Incoming Dial-peer=2365 Is Matched
*May 1 06:32:29.275: //-1/BB0B78D782B1/DPM/dpMatchPeertype:exit@5600
*May 1 06:32:29.275: //-1/BB0B78D782B1/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=2365
In fact, it matches again on the answer-address command:
*May 1 06:32:29.283: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Calling Number=01755521544, Called Number=, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
*May 1 06:32:29.283: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Match Rule=DP_MATCH_ANSWER; Calling Number=01755521544
*May 1 06:32:29.283: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:
Is Incoming=TRUE, Number Expansion=FALSE
*May 1 06:32:29.283: //-1/xxxxxxxxxxxx/DPM/dpMatchCore:
Dial String=, Expanded String=, Calling Number=01755521544T
Timeout=TRUE, Is Incoming=TRUE, Peer Info Type=DIALPEER_INFO_SPEECH
*May 1 06:32:29.283: //-1/xxxxxxxxxxxx/DPM/MatchNextPeer:
Result=Success(0); Incoming Dial-peer=2365 Is Matched
*May 1 06:32:29.287: //-1/xxxxxxxxxxxx/DPM/dpMatchPeertype:exit@5600
*May 1 06:32:29.287: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_ANSWER; Incoming Dial-peer=2365
The call is being matched to dial-peer 2364 for the outbound leg which fails the call. Note that the calling/called number has been translated. I'm not sure whether this was intentional or not.
*May 1 06:32:32.895: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 480 Temporarily not available
Via: SIP/2.0/UDP 10.10.10.142:5060;branch=z9hG4bK7D31E6E
From: <>9610997929@10.10.20.236>;tag=E162D48-F4D>
To: <>9610997929@10.10.20.236>;tag=f78f28af4f0458e3fd8ae721>
Call-ID: BB119398-B15F11E2
VoiceGW#-82B7853B-A6F7589C@10.10.10.142
CSeq: 101 INVITE
User-agent: SysMaster VoIP Gateway v1.2.0
Content-Length: 0
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05-01-2013 10:13 PM
Thanks for your response.
It was worng in my voice translation rule i fixed this.
But now i am able to play IVR prompt from router local flash: but when users are dialing the enduser ext. through IVR calls are landing but no ringback tone & no RTP. Calls are disconnecting after few seconds.
Please see the attached new log file.
show voice call active brief
1629 : 2990 325947910ms.1 +3770 pid:30 Originate 434 active
dur 00:00:08 tx:0/0 rx:442/70720
IP 192.168.10.242:25212 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
No tx:0/0 value,
CUCM IP:192.168.10.242
Call flow is: SIP IPTSP------------Voice Gateway------IVR (router flash)------------CUCM--------IP Phone.
for more detail log pls see the attached log file.
-Thanks
05-02-2013 02:15 AM
Hi,
Please share the running configuration of the CUBE. Also, any chance that you can set the CCM traces to detailed and collect them for one call that faces the no audio issue?
Please do also mention the calling number and the End user extension dialed.
Regards,
Jagpreet
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