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In coming Call from SIP trunk through IPTSP is not matching inbound voip dialpeer.

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

3 Replies 3

Jonathan Schulenberg
Hall of Fame
Hall of Fame

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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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

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