03-15-2014 12:50 PM - edited 03-16-2019 10:07 PM
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!
03-15-2014 10:48 PM
Hi Aleksandr ,
Please attach your CME router config with "debug ccsip message" and " debug isdn q931"
Thanks
Manish
03-16-2014 11:00 AM
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
03-16-2014 11:41 AM
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
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide