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leandro.brito
Beginner

SIP Trunk Between 2 CMEs (error: No Inbound Container Created !!!)

Dear all,

I have the following scenario:

CME-A ----- SIP Trunk ----- CME-B

CME-A Extensions 30..

CME-B Extensions 64..

CME-B can call CME-A - working fine

When a user from CME-A tries to reach an extension on CME-B I always get the busy tone and the following message on the phone: "Unknown Number".

Both dial-peers are configured the same way, changing only the destination address.

On debug I got the following error:

039196: Jun 26 20:41:50.663: //122612/A6F93FE59ADB/SIP/Error/act_sentinvite_wait_100: Out of retries

039197: Jun 26 20:41:50.663: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQSIG: No Inbound Container Created !!!

039198: Jun 26 20:41:50.663: //-1/xxxxxxxxxxxx/SIP/Error/sipSPIGetContentQ931: No Inbound Container Created !!!

039199: Jun 26 20:41:50.663: //122612/A6F93FE59ADB/SIP/Media/sipSPIDestroyRtpSession: stream:30ED0D70

Any ideas on what can be done to help me find this issue?

Regards,

Leandro Brito

8 REPLIES 8

Please send us your shrun from both ccme and the debug ccsip messages and debug voip ccapi inout from both. Attach it to the post here

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Follow the configs and some more information:

CME-A

- All incoming/outcoming calls to PSTN are made by a SIP Trunk (working fine)

- The problem is only with calls to dial-peer 102 (172.30.0.1)

- I tried to get only the relevant information

voice service voip

ip address trusted list

  ipv4 172.40.0.0 255.255.255.0

  ipv4 192.168.254.0 255.255.255.0

  ipv4 172.20.0.0 255.255.255.0

  ipv4 192.168.254.1

  ipv4 172.30.0.0 255.255.255.0

  ipv4 192.168.151.0 255.255.255.0

  ipv4 201.33.210.0 255.255.255.0

  ipv4 201.33.209.0 255.255.255.0

  ipv4 10.10.51.0 255.255.255.0

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

redirect ip2ip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

h323

  h225 h245-address on-connect

  call preserve

sip

  rel1xx disable

  min-se 86400 session-expires 86400

  header-passing

  early-offer forced

  midcall-signaling passthru

  g729 annexb-all

!

voice class codec 1

codec preference 1 g729br8

codec preference 2 g729r8

codec preference 3 g711ulaw

voice translation-rule 1

rule 1 /^.*/ /xxxxxxx/

!

voice translation-rule 2

rule 1 /0/ /011/

!

voice translation-rule 3

rule 1 /00/ /0/

!

!

voice translation-profile PSTN_Outgoing

translate calling 1

translate called 2

!

voice translation-profile SIP_DDD

translate calling 1

translate called 3

!

!

interface GigabitEthernet0/0.172

description ## Voice Lan ##

encapsulation dot1Q 172

ip address 172.40.0.1 255.255.255.0

dial-peer voice 1 voip

description ## Incoming Call from SIP Trunk ##

session protocol sipv2

session target sip-server

incoming called-number .

voice-class sip dtmf-relay force rtp-nte

dtmf-relay rtp-nte

no vad

dial-peer voice 102 voip

description $$ CME-B ###

destination-pattern 64..

session protocol sipv2

session target ipv4:172.30.0.1

dtmf-relay rtp-nte

no vad

sip-ua

credentials username xxxxxx password bbbbbbbb realm sip.provider.net

authentication username xxxxxx password bbbbbbb

no remote-party-id

disable-early-media 180

retry invite 2

retry register 10

timers connect 100

registrar dns:sip.provider.net expires 300

sip-server dns:sip.provider.net

no transport tcp

!

telephony-service

no auto-reg-ephone

max-ephones 15

max-dn 15

ip source-address 172.40.0.1 port 2000

auto assign 1 to 30

time-zone 17

time-format 24

date-format dd-mm-yy

max-conferences 8 gain -6

call-forward pattern 20..

call-forward pattern 64..

transfer-system full-consult

transfer-pattern 64..

transfer-pattern 20..

secondary-dialtone 0

create cnf-files version-stamp 7960 Jun 27 2013 09:40:49

!

!

ephone-dn  7

number 3007

label 3007

name Teste

!

ephone  7

mac-address 588D.0972.6274

max-calls-per-button 2

type 6961

button  1:7

CME-B

- Connected to PSTN by an E1 card (working fine incoming/outcoming calls)

- Calls to dial-peer 130 (CME-A) goes fine;

CME-B#show run

card type e1 0 3

network-clock-participate wic 3

voice-card 0

dspfarm

dsp services dspfarm

voice service voip

ip address trusted list

  ipv4 172.20.0.0 255.255.255.0

  ipv4 172.30.0.0 255.255.255.0

  ipv4 192.168.252.0 255.255.255.0

  ipv4 10.10.51.0 255.255.255.0

  ipv4 192.168.252.1

  ipv4 172.40.0.0 255.255.255.0

  ipv4 192.168.254.0 255.255.255.0

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

redirect ip2ip

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

h323

sip

  bind control source-interface GigabitEthernet0/0.172

  bind media source-interface GigabitEthernet0/0.172

  registrar server expires max 1200 min 300

dial-peer voice 1 pots

description # Inbound PSTN E1 #

incoming called-number .

direct-inward-dial

dial-peer voice 2 voip

description ## Inbound SIP Trunk ##

incoming called-number .

direct-inward-dial

dial-peer voice 130 voip

description $$ CME-A ###

destination-pattern 30..

session protocol sipv2

session target ipv4:172.40.0.1 (CME-A)

dtmf-relay rtp-nte

no vad

telephony-service

no auto-reg-ephone

max-ephones 20

max-dn 15

ip source-address 172.30.0.1 port 2000

auto assign 1 to 30

url services http://172.30.0.2/voiceview/common/login.do

url authentication http://172.30.0.1/CCMCIP/authenticate.asp 

cnf-file location flash:

cnf-file perphone

time-zone 17

time-format 24

date-format dd-mm-yy

dialplan-pattern 1 11xxxxxx.. extension-length 4 extension-pattern 64..

voicemail 8200

max-conferences 8 gain -6

call-park system application

call-forward pattern 89..

call-forward pattern 20..

call-forward pattern 30..

moh "music-on-hold.au"

dn-webedit

time-webedit

transfer-system full-consult

transfer-pattern 64..

transfer-pattern 82..

transfer-pattern 89..

transfer-pattern 20..

transfer-pattern 0T

transfer-pattern 30..

secondary-dialtone 0

create cnf-files version-stamp Jan 01 2002 00:00:00

ephone-dn  7

number 6420

name TESTE

ephone  4

device-security-mode none

mac-address 0026.C763.CBE6

username "user4" password 1234

type CIPC

button  1:7

I'll get the debugs and will share with you in a while.

It already has it.

Hi

Yon can try to modify this on CME-B

dial-peer voice 2 voip

description ## Inbound SIP Trunk ##

incoming called-number 64..

session protocol sipv2

session target ipv4:172.30.0.1

dtmf-relay rtp-nte

no vad

Unfortunately I got the same error after that.

cmaiolani
Beginner

Hi,

have you insert an incoming dial peer for trunk in CME-A?

Sent from Cisco Technical Support iPhone App

leandro.brito
Beginner

Solved

On CME-B I've added as trusted address the IP of the interface used by CME-A to send SIP signaling/audio and it works.

The problem was that I couldn't found this "toll fraud" situation by debug.

UP and RUNNING - thanks guys for your help!!

If you had sent the debug--we will have seen this in good time. In the debug you will see a cause code of 21..That is the toll fraud error code

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

Please rate all useful posts
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