Calls from Sip Trunk to UC540 and then to CUE returned ** Service Unavailable**

[chrysostomos1980]

Hi to all

i have something strange here and i need your assistance

Call Flow:

Sip trunk-->UC540--> CUE

When calls coming to UC540 from outside and then going to cue then we send back service unavailable.I made a translation and i sent directly the incoming calls to CUE

The same behavior is also if i send the calls to dummy number and then from there set forward all to voice mail.

Incoming voicemail is working fine

Incoming calls to phones also ok

Uc540: 8.6

CUE: 8.6.5

A number: 99999999

B number: 22777777

Voice Mail Number:111

Attached is the trace

i see that we hit the correct dial peers .

I have enable only trancoder since MTP is not register ( don't know why , but i don't think also that is necessary..

voice service voip
 ip address trusted list
  ipv4 172.16.80.0 255.255.255.0
  ipv4 172.16.81.0 255.255.255.0
 allow-connections sip to sip
 supplementary-service h450.12
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 supplementary-service media-renegotiate
 sip
  no update-callerid

!

!
dial-peer voice 1000 voip
 description **SIP TRUNK**
 translation-profile incoming SIP-INCOMING
 translation-profile outgoing SIP-OUTGOING
 destination-pattern 9T
 modem passthrough nse codec g711alaw
 session protocol sipv2
 session target sip-server
 incoming called-number .T
 voice-class codec 2  
 voice-class sip dtmf-relay force rtp-nte
 dtmf-relay rtp-nte
 fax-relay ecm disable
 no fax-relay sg3-to-g3
 fax rate 9600
 fax protocol pass-through g711alaw
 no vad
!

!
dial-peer voice 2001 voip
 description ** cue voicemail pilot number **
 destination-pattern 111
 b2bua
 session protocol sipv2
 session target ipv4:10.1.10.1
 incoming called-number 111
 no voice-class sip outbound-proxy   
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!

Regards

chrysostomos

[Manish Prasad]

Hi Chrysostomos,

Please set the allow connection under voip service to allow the call to move from one sip leg to other.

Voice service voip

allow connections sip to sip

Second as you are using b2bua on dial-peer then its important to configure media source ip , so that it modify the media ip/port and act as media flow-through mode.

on global mode...

voice service voip

sip

bind media source interface ......

bind media control interface .......

or configure it on dial-peer (recommended)

dial-peer voice xxxx voip

voice-class sip bind control source-interface ........

voice-class sip bind media source-interface .......

Collect the debug if it fails again.

Thanks

Manish

[chrysostomos1980]

Hi

Problem solved

It was a BUG to the IOS of UC540

Changed the ios fixed the problem

[chrysostomos1980]

Hi

Interface                  IP-Address      OK? Method Status                Protocol
FastEthernet0/0            unassigned      YES NVRAM  up                    up
FastEthernet0/0.10         192.168.0.10    YES DHCP   up                    up   ----> For internet
FastEthernet0/0.20         10.151.5.130    YES NVRAM  up                    up  ------> For sip trunk
In0/0                      10.1.10.2       YES unset  up                    up    --------> default gw for cue
Vlan1                      unassigned      YES unset  up                    up
Vlan100                    unassigned      YES unset  up                    up
Vlan200                    unassigned      YES unset  up                    down
Vlan300                    unassigned      YES unset  up                    down
NVI0                       10.1.10.2       YES unset  up                    up
BVI1                       192.168.20.1    YES NVRAM  up                    up
BVI100                     10.1.1.1        YES NVRAM  up                    up   ---------> ip for cme
Loopback0                  10.1.10.2       YES NVRAM  up                    up   ------> default gw for cue

!

!
dial-peer voice 2001 voip
 description ** cue voicemail pilot number **
 destination-pattern 111
 b2bua
 session protocol sipv2
 session target ipv4:10.1.10.1
 incoming called-number 111
 no voice-class sip outbound-proxy
 voice-class sip bind control source-interface BVI100
 voice-class sip bind media source-interface BVI100
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
!

!
interface FastEthernet0/0.10
 description **FOR INTERNET**
 encapsulation dot1Q 10
 ip address dhcp
 ip access-group 105 in
 ip nat outside
 ip inspect SDM_LOW out
 ip virtual-reassembly in
!
interface FastEthernet0/0.20
 description **FOR SIP TRUNK WITH ISP**
 encapsulation dot1Q 20
 ip address 10.151.5.130 255.255.255.240
!

ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
!
ping 10.1.10.1 source bvi100
Type escape sequence to abort.
Sending 5, 100-byte ICMP Echos to 10.1.10.1, timeout is 2 seconds:
Packet sent with a source address of 10.1.1.1
!!!!!
Success rate is 100 percent (5/5), round-trip min/avg/max = 1/1/1 ms

I have bind the interface of cme ( 10.1.1.1) but the call fails again

Attached is the trace

Anything to advice?

[Manish Prasad]

Hi,

Which all codecs that voice-class codec 2 contains ?

The intial INVITE has little wired SDP parameters , it does not contains audio media attribute like this..

a=rtpmap:8 PCMA/8000

Nevertheless this attribute is not a mandatory parameters of SDP.

As only single media is offered from the initial INVITE i.e. G711alaw and you know CUE only supports ulaw , so you need transcoders to convert alaw to ulaw.

What is the trancoder status and how it is configured ?

Thanks

Manish

[chrysostomos1980]

Hi

Thank you for the response again

Yes i have a trancoder enable and is registered successfully

///////////////////////////////////

voice class codec 2  

codec preference 1 g711alaw

 codec preference 2 g711ulaw
 codec preference 3 g729br8
 codec preference 4 g729r8

!

sccp local bvi100
sccp ccm 10.1.1.1 identifier 1 version 7.0
sccp
!
sccp ccm group 1
 associate ccm 1 priority 1
 associate profile 1 register transcode

!
dspfarm profile 1 transcode  
 codec g729abr8
 codec g729ar8
 codec g711alaw
 codec g711ulaw
 codec g729r8
 codec g729br8
 maximum sessions 8
 associate application SCCP
!

telephony-service
 sdspfarm units 1
 sdspfarm tag 1 transcode

!

show dspfarm all
Dspfarm Profile Configuration

 Profile ID = 1, Service = TRANSCODING, Resource ID = 1
 Profile Description :
 Profile Service Mode : Non Secure
 Profile Admin State : UP
 Profile Operation State : ACTIVE
 Application : SCCP   Status : ASSOCIATED
 Resource Provider : FLEX_DSPRM   Status : UP
 Number of Resource Configured : 8
 Number of Resource Available : 8
 Codec Configuration: num_of_codecs:6
 Codec : g729br8, Maximum Packetization Period : 60
 Codec : g729r8, Maximum Packetization Period : 60
 Codec : g711ulaw, Maximum Packetization Period : 30
 Codec : g711alaw, Maximum Packetization Period : 30
 Codec : g729ar8, Maximum Packetization Period : 60
 Codec : g729abr8, Maximum Packetization Period : 60
Dspfarm Profile Configuration

SLOT DSP VERSION  STATUS CHNL USE   TYPE    RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

0    1   28.3.10  UP     N/A  FREE  xcode   1      -         -         -
0    2   28.3.10  UP     N/A  FREE  xcode   1      -         -         -
0    2   28.3.10  UP     N/A  FREE  xcode   1      -         -         -
0    2   28.3.10  UP     N/A  FREE  xcode   1      -         -         -
0    2   28.3.10  UP     N/A  FREE  xcode   1      -         -         -
0    2   28.3.10  UP     N/A  FREE  xcode   1      -         -         -
0    2   28.3.10  UP     N/A  FREE  xcode   1      -         -         -
0    3   28.3.10  UP     N/A  FREE  xcode   1      -         -         -

[Manish Prasad]

Hi,

Under telephony service you only have that much configuration ? I mean that you have to configure dummy DN and ephone under telephony service and also interface binding like this..

telephony service

max-dn 1

max-ephone 1

ip source-address 10.1.1.1 port 2000

ephone-dn  1

number xxxx

ephone 1

mac zzz.zzz.zzz

button 1:1

Then create cnf under telephony service..

And also add

sdpfarm transcode session 8 under telephony service.

Do a "no sccp" and then "sccp" and then test a call , if fails again send trace and "show sccp" logs.

Thanks

Manish

[chrysostomos1980]

Hi

My friend i have done thousands of UC implementations :) Yes its there all the appropriates configs.Just i upload here only the related with this issue

All the config is correct

Its seems that the trancoding is not working at all

May is a bug of the current software/IOS  dont know

Current IOS:

boot system flash:/uc500-advipservicesk9-mz.151-4.M7

[Manish Prasad]

If trancoders are showing active in "show sccp" and configuration are correct , then it might be an IOS bug and you need to look at bug toolkit.

I will give a last try if you give me "debug ccsip all" only for a failing call to look at sip stack level trace.

Thanks

Manish

[g.egregoriou]

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