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Beginner

CME Incoming Calls Busy Tone Constantly

All,

I've setup a new uc560 system and have been experiencing issues with incoming calls going directly to busy. Outgoing calls seem to be working fine but incoming always busy. Ideally when the call comes in I expect it to hit the auto attendant at 3098 and allowing customers to make a selection. Executed a "debug ccsip error" session and see the following errors;

SIP: (3204) Attribute mid, level 1 instance 1 not found.
002206: Nov 29 09:19:01.146: //3204/FD96AA03A5E5/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
002207: Nov 29 09:19:01.146: //3204/FD96AA03A5E5/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
002208: Nov 29 09:19:01.146: //3204/FD96AA03A5E5/SIP/Error/sipSPI_ipip_update_call_entry:
failed to update call entry
002209: Nov 29 09:19:01.146: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 3204
002210: Nov 29 09:19:01.146: //3204/FD96AA03A5E5/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: (3205) Attribute mid, level 1 instance 1 not found.
002211: Nov 29 09:19:01.486: //3205/FDCA8B07A5EA/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
002212: Nov 29 09:19:01.486: //3205/FDCA8B07A5EA/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
002213: Nov 29 09:19:01.486: //3205/FDCA8B07A5EA/SIP/Error/sipSPI_ipip_update_call_entry:
failed to update call entry
002214: Nov 29 09:19:01.486: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 3205
002215: Nov 29 09:19:01.490: //3205/FDCA8B07A5EA/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: (3206) Attribute mid, level 1 instance 1 not found.
002216: Nov 29 09:19:02.578: //3206/FE712B81A5EF/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
002217: Nov 29 09:19:02.578: //3206/FE712B81A5EF/SIP/Error/sipSPI_ipip_update_codec_params_in_channelInfo:
failed to update call entry
002218: Nov 29 09:19:02.578: //3206/FE712B81A5EF/SIP/Error/sipSPI_ipip_update_call_entry:
failed to update call entry
002219: Nov 29 09:19:02.578: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_ipip_set_channel_count: Unable to set CHANNEL_COUNT for callid 3206
002220: Nov 29 09:19:02.578: //3206/FE712B81A5EF/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.
002222: Nov 29 09:19:37.131: //3208/000000000000/SIP/Error/sipSPIAddPrivacyHeader: Orig Container is NULL...should have value
002223: Nov 29 09:19:37.131: //3208/000000000000/SIP/Error/sipSPIAddAssertedIDHeader: Orig Container is NULL...should have value
002224: Nov 29 09:19:37.131: //3208/000000000000/SIP/Error/sipSPIAddPreferredIDHeader: Orig Container is NULL...should have value
002227: Nov 29 09:21:17.276: //3212/000000000000/SIP/Error/sipSPIAddPrivacyHeader: Orig Container is NULL...should have value
002228: Nov 29 09:21:17.276: //3212/000000000000/SIP/Error/sipSPIAddAssertedIDHeader: Orig Container is NULL...should have value
002229: Nov 29 09:21:17.276: //3212/000000000000/SIP/Error/sipSPIAddPreferredIDHeader: Orig Container is NULL...should have value
002231: Nov 29 09:22:09.281: //-1/xxxxxxxxxxxx/SIP/Error/get_content_length: Could not get Content-length
002232: Nov 29 09:22:09.281: //3215/000000000000/SIP/Error/sipSPIRegPthruProcessResponse: Error NO RPCB

002233: Nov 29 09:22:57.382: //3217/000000000000/SIP/Error/sipSPIAddPrivacyHeader: Orig Container is NULL...should have value
002234: Nov 29 09:22:57.382: //3217/000000000000/SIP/Error/sipSPIAddAssertedIDHeader: Orig Container is NULL...should have value
002235: Nov 29 09:22:57.382: //3217/000000000000/SIP/Error/sipSPIAddPreferredIDHeader: Orig Container is NULL...should have value
002238: Nov 29 09:24:37.480: //3222/000000000000/SIP/Error/sipSPIAddPrivacyHeader: Orig Container is NULL...should have value
002239: Nov 29 09:24:37.480: //3222/000000000000/SIP/Error/sipSPIAddAssertedIDHeader: Orig Container is NULL...should have value

Any assistance would be greatly appreciated!

1 ACCEPTED SOLUTION

Accepted Solutions
Rising star

Have you created voice

Have you created voice translation rule to convert incoming called number (2105551234) to 3098 ? I am seeing disconnect cause 1 in logs and that means number is unassigned.

Suresh

8 REPLIES 8

Hi,

Hi,

please check the following in your configuration:

voice service voip
sip
bind control source-interface <your interface>
bind media source-interface <your interface>


Hope this helps.

Beginner

Here is what I have:

Here is what I have:

voice service voip
 ip address trusted list
  ipv4 0.0.0.0 0.0.0.0
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 supplementary-service h450.12
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 fax protocol none
 no fax-relay sg3-to-g3
 sip
  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0
  registrar server expires max 3600 min 1800
  localhost dns:xxxxxxxx.com
  outbound-proxy dns:pbx.singlepipe.net
  no update-callerid
  sip-profiles 1000

G0/0 is the WAN on a public IP

Cisco Employee

can  you please share the

can  you please share the output of " show license status application from CUE.

make one incoming call enable below debug

debug voice ccapi inout

debug ccsip messages

Br, Nadeem Please rate all useful post.
Beginner

UC_560#service-module

UC_560#service-module integrated-Service-Engine 0/0 session
Trying 10.1.10.2, 2002 ... Open

se-10-1-10-1# show license status application
voicemail enabled: 12 ports, 12 sessions, 125 mailboxes
ivr disabled, no unexpired installed ivr session license available

I've attached the debug as requested in a text file.

On a side note I noticed this ip address ( 172.16.6.141 ) in the log a few times and it's non-existent on my UC560 system.

UC_560#sh ip ro 172.16.6.141
% Network not in table

Rising star

Have you created voice

Have you created voice translation rule to convert incoming called number (2105551234) to 3098 ? I am seeing disconnect cause 1 in logs and that means number is unassigned.

Suresh

Beginner

Here are my translation rules

Here are my translation rules;

voice translation-rule 410
rule 1 /^9\(.*\)/ /\1/
rule 15 /^....$/ /2105551234/
!
voice translation-rule 411
rule 1 /^9\(.*\)/ /ABCD9\1/
!
voice translation-rule 412
rule 1 /^ABCD\(.*\)/ /\1/
!
voice translation-rule 422
rule 1 /^ABCD91900......./ //
rule 2 /^ABCD91976......./ //
rule 15 /^ABCD\(.*\)/ /\1/
!
voice translation-rule 1000
rule 1 /.*/ //
!
voice translation-rule 1111
rule 15 /^.*$/ /2105551234/
!
voice translation-rule 1112
rule 1 /^9/ //
!
voice translation-rule 2001
!
voice translation-rule 2002
rule 1 /^6/ //
rule 2 /^A/ //
!
voice translation-rule 2222
rule 1 /^91900......./ //
rule 2 /^91976......./ //
!
!
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
!
voice translation-profile CallBlocking
translate called 2222
!
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
!
voice translation-profile PSTN_CallForwarding
translate redirect-target 410
translate redirect-called 410
!
voice translation-profile PSTN_Outgoing
translate calling 1111
translate called 1112
translate redirect-target 410
translate redirect-called 410
!
voice translation-profile SIP_Incoming
translate called 411
!
voice translation-profile SIP_Passthrough
translate called 412
!
voice translation-profile SIP_Passthrough_CallBlocking
translate called 422
!
voice translation-profile XFER_TO_VM_PROFILE
translate redirect-called 2002
!
voice translation-profile nondialable
translate called 1000

Highlighted
Cisco Employee

Can you please share show run

Can you please share show run from Router.

As correctly pointed by Suresh, There is no route for DN 2105551212/ dial-peer for outgoing leg match, hence we are sending 404 not found. you need to correct dial-peer or translation pattern which need to match or convert to AA number something

Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 209.209.172.231:5060;branch=z9hG4bKtfqf5o201g2h8ucjh2i0.1
From: <sip:2105551212@172.16.6.141;user=phone>;tag=807785917-1480477104226-
To: "Jack Smith"<sip:2105551234@voice.comms.com>;tag=D77E1C-2C1
Date: Wed, 30 Nov 2016 03:38:24 GMT
Call-ID: BW223824226291116690284813@172.16.6.141
CSeq: 430435378 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=1
Content-Length: 0

Br, Nadeem Please rate all useful post.
Beginner

I corrected the issue with

I corrected the issue with your advice and added the voice translation which resolved the issues and calls are coming into AA properly.

Thanks for the assistance!!

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