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CME - inbound call proble

HELLO!
I have a router c2801-adventerprisek9_ivs-mz.151-4.M7.bin with cisco cme.

Asterisk (192.168.1.50) available via fa0/1. cme registered  on asterisk via sip ua:

sip-ua
 credentials username cisco password 7 110A1016141D realm 192.168.1.50
 authentication username cisco password 
 registrar ipv4:192.168.1.50 expires 3600
 sip-server ipv4:192.168.1.50
 connection-reuse
 host-registrar

 C#show sip-ua register status
Line                             peer       expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
cisco                            -1         2195         yes

Dial-PEER:
dial-peer voice 2 voip
 translation-profile incoming incoming
 translation-profile outgoing outgoing
 destination-pattern 9.T
 session protocol sipv2
 session target sip-server
 voice-class codec 1
 no voice-class sip outbound-proxy
 voice-class sip bind control source-interface FastEthernet0/1
 voice-class sip bind media source-interface FastEthernet0/1
 dtmf-relay rtp-nte
 no vad

and translation-profiles

voice translation-rule 1040
 rule 1 /^.$/ /61/
voice translation-profile incoming
 translate called 1040

 

When I try to call through Asterisk extensions cisco I see the following in the debug:

*Mar 16 00:07:46.464: //4782/4F5328EDAA8B/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
*Mar 16 00:07:46.464: //4782/4F5328EDAA8B/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
*Mar 16 00:07:46.464: //4782/4F5328EDAA8B/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
*Mar 16 00:07:46.464: //4782/4F5328EDAA8B/SIP/Error/sipSPI_ipip_update_call_entry:
failed to update call entry
*Mar 16 00:07:46.464: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 4782
SIP Call statistics tracing is enabled
DC#
*Mar 16 00:07:46.464: //4782/4F5328EDAA8B/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo: Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.
SIP: (4784) Attribute mid, level 1 instance 1 not found.
*Mar 16 00:08:06.308: //4784/5B271D43AA91/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
*Mar 16 00:08:06.308: //4784/5B271D43AA91/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
*Mar 16 00:08:06.312: //4784/5B271D43AA91/SIP/Error/sipSPI_ipip_update_call_entry:
failed to update call entry
*Mar 16 00:08:06.312: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 4784
*Mar 16 00:08:06.312: //4784/5B271D43AA91/SIP/Error/sip_iwf_sip_copy_sdp_to_channelInfo: Channel count is not set at this point. Not SIP-SIP or SET_MODE is not done.
*Mar 16 00:08:06.336: //4784/5B271D43AA91/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x6A4F36A0
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 791182681XX
Called Number            : 892539780XX
Source IP Address (Sig  ): 192.168.1.1
Destn SIP Req Addr:Port  : 192.168.1.50:5060
Destn SIP Resp Addr:Port : 192.168.1.50:5060
Destination Name         : 192.168.1.50

*Mar 16 00:08:06.336: //4784/5B271D43AA91/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 192.168.1.1
Source IP Port    (Media): 17940
Destn  IP Address (Media): 192.168.1.50
Destn  IP Port    (Media): 12112
Orig Destn IP Address:Port (Media): [ - ]:0

*Mar 16 00:08:06.336: //4784/5B271D43AA91/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 1
Disconnect Cause (SIP)   : 404

On logs in Asterisk:

Mar 15 22:14:52] WARNING[1406] chan_sip.c: Received response: "Forbidden" from '"791182681XX" <sip:79118268147@192.168.1.50>;tag=as45a98eba'
[Mar 15 22:19:03] WARNING[1406] chan_sip.c: Received response: "Forbidden" from '"791182681XX" <sip:79118268147@192.168.1.50>;tag=as63b73bad'

 

Please help!

 

 

 

3 REPLIES

Hi Aleksandr , Please attach

Hi Aleksandr ,

 

Please attach your CME router config with "debug ccsip message" and " debug isdn q931"

 

Thanks

Manish

New Member

Hello!Thank for  your reply.

Hello!

Thank for  your reply. I uploaded running-config

 

debug ccsip message

*Mar 16 22:13:13.082: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:89253978049@192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK5a0420e4
Max-Forwards: 70
From: "79117501907" <sip:79117501907@192.168.1.50>;tag=as379ce31a
To: <sip:89253978049@192.168.1.1:5060>
Contact: <sip:79117501907@192.168.1.50:5060>
Call-ID: 54b1abd83b114b1a69553bd471da6316@192.168.1.50:5060
CSeq: 102 INVITE
User-Agent: VoxStack Wireless Gateway
Date: Sun, 16 Mar 2014 17:55:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Best-Codec: ulaw
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 1549434951 1549434951 IN IP4 192.168.1.50
s=VoxStack Wireless Gateway
c=IN IP4 192.168.1.50
t=0 0
m=audio 12894 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

*Mar 16 22:13:13.106: //6881/78E3F681BE36/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK5a0420e4
From: "79117501907" <sip:79117501907@192.168.1.50>;tag=as379ce31a
To: <sip:89253978049@192.168.1.1:5060>
Date: Sun, 16 Mar 2014 18:13:13 GMT
Call-ID: 54b1abd83b114b1a69553bd471da6316@192.168.1.50:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0


*Mar 16 22:13:13.106: //6881/78E3F681BE36/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK5a0420e4
From: "79117501907" <sip:79117501907@192.168.1.50>;tag=as379ce31a
To: <sip:89253978049@192.168.1.1:5060>;tag=F1A1568-263A
Date: Sun, 16 Mar 2014 18:13:13 GMT
Call-ID: 54b1abd83b114b1a69553bd471da6316@192.168.1.50:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=1
Content-Length: 0


*Mar 16 22:13:13.110: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:89253978049@192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK5a0420e4
Max-Forwards: 70
From: "79117501907" <sip:79117501907@192.168.1.50>;tag=as379ce31a
To: <sip:89253978049@192.168.1.1:5060>;tag=F1A1568-263A
Contact: <sip:79117501907@192.168.1.50:5060>
Call-ID: 54b1abd83b114b1a69553bd471da6316@192.168.1.50:5060
CSeq: 102 ACK
User-Agent: VoxStack Wireless Gateway
Content-Length: 0

 

debug isdn q931 - I not founded this command on my router

VIP Super Bronze

The number dialled from the

The number dialled from the asterisk gateway diesnt exist on your ccme..You have a xlation rule to xlate any number to 61 but its not applied to any incoming dial-peer..so I suggest you do this

dial-peer voice 5 voip

incoming called number .

session protocol sip

translation-profile incoming incoming

 

 

 

 

89253978049

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