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Cisco Unity Express - Codec Issues

Hello Community Members,

I am currently having an issue with my Cisco Unity Express setup. I am running CME on a Cisco 2951 (IOS 15) and Cisco Unity Express 8.6 from an ISM.  The CME and CUE are integrated via SIP and I do have a SIP Dial Peer from CME to CUE.

Everything is working well internally. When I dial from an IP Phone registered on the CME to an Auto Attendant number set up on CUE, I easy get connected and I am able to dial by name or number and the call is correctly routed to the right extension.

The CME connect onto my ITSP via a SIP Trunk and the ITSP has forced all incoming calls into the CME to G.729r8 only. But they have allowed me to choose any codec I want for outbound calls such as G.711 and G.729.

When I dial from the PSTN into the CUE, the Auto Attendant answer the call and prompt me if I would like to dial by extension or name. When I select the option to dial by extension and enter the extension followed by #, the Auto Attendant says, calling extension "5001" and then silence on the line until the call get disconnected after sometimes.

While my external call is connected onto the Auto Attendant, I can see that there is an active transcoding session going on converting the call coming from the ITSP from G.729 to G.711. Once I have entered the extension that I am trying to dial followed by such as "5001" then #, the conference resources that was initially allocated disappear and I hear nothing but silence.

Is there anything that I could do in order to fix this?

This setup used to work properly before because my ITSP was allowing both G.711 and G.729 inbound. Since they have locked that down to G.729, I am stuck.

When my external call is connected onto the Auto Attendant, I can see the following output on CME:

CME#sh sccp connection

sess_id    conn_id      stype mode     codec   sport rport ripaddr conn_id_tx

 

1769481    36           xcode sendrecv g711u   28196 2000  192.168.0.1           

1769481    40           xcode sendrecv g729    19970 2000  192.168.0.1           

 

Total number of active session(s) 1, and connection(s) 2

 

CME#sh sdspfarm sessions summary 

 

 max-mtps:5, max-streams:16, alloc-streams:16, act-streams:2

  ID   MTP  State      CallID confID   Usage                         Codec/Duration   Type

==== ===== ====== =========== ======== ============================= ==============   ===========

1    1     IDLE   -1          0x0                                     G711Ulaw64k /20ms Audio      

2    1     IDLE   -1          0x0                                     G711Ulaw64k /20ms Audio      

3    1     IDLE   -1          0x0                                     G711Ulaw64k /20ms Audio      

4    1     IDLE   -1          0x0                                     G711Ulaw64k /20ms Audio      

5    1     IDLE   -1          0x0                         1     IDLE   -1          0x0                                     G711Ulaw64k /20ms Audio      

7    1     START  9838    /2 ms Audio      

7    1     START  9838        0x1B0007   Ip-Ip                        G711Ulaw64k /20ms Audio      

8    1     START  9837        0x1B0007   Ip-Ip                        G729 /20ms Audio      

9    1     IDLE   -1          0x0                                     G711Ulaw64k /20ms Audio      

10   1     IDLE   -1          0x0                                     G711Ulaw64k /20ms Audio      

Any idea on how to go about this? I have transcoding resources configured locally and registered on the CME.

Warm regards,

Johnny Kabundi.

4 REPLIES

Do you have codec g728r8 in

Do you have codec g729r8 in you transcoder profile?

 

dspfarm profile 2 transcode  
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 codec g729r8
 maximum sessions 3
 associate application SCCP

Voice CCIE #37771
Community Member

Hello,Yes, I do have

Hello,

Yes, I do have transcoding configured with all the required codec as per your description above.

What else could be the problem?

JK.

Community Member

Hello,I was able to re-create

Hello,

I was able to re-create my client scenario by doing the following in my lab:

R1 (CME) connect to R2 (CME) over the WAN. IP Routing has been configured between the two sites and working.

NOTE: R2 is CME that also runs Cisco Unity Express 7.0 on AIM-CUE.

I have configured the 2 x CME sites passes calls over the WAN using G.729r8. 

R1 - Number Range is 2XXX

R2 - Number Range is 3XXX and 40XX on CUE.

The config on R2 (CME - CUE) look like this:

R2#sh run | s voice service voip
voice service voip
 ip address trusted list
  ipv4 10.10.0.0 255.255.0.0
  ipv4 192.168.1.0 255.255.255.0
 allow-connections h323 to h323
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily
BR2#sh run | s sccp
sccp local FastEthernet0/0.600
sccp ccm 10.10.160.1 identifier 1 version 7.0 
sccp
sccp ccm group 1
 bind interface FastEthernet0/0.600
 associate ccm 1 priority 1
 associate profile 1 register MTP123456789
BR2#sh run | s dspfarm
 dsp services dspfarm
dspfarm profile 1 transcode  
 codec g729r8
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 maximum sessions 5
 associate application SCCP
 sdspfarm units 1
 sdspfarm transcode sessions 5
 sdspfarm tag 1 MTP123456789
R2#sh run | s telephony
telephony-service
 sdspfarm units 1
 sdspfarm transcode sessions 5
 sdspfarm tag 1 MTP123456789
 no auto-reg-ephone
 max-ephones 10
 max-dn 20
 ip source-address 10.10.160.1 port 2000
 system message HOME LAB CUCME
 time-zone 29
 time-format 24
 date-format dd-mm-yy
 voicemail 4000
 mwi relay
 max-conferences 4 gain -6
 transfer-system full-consult
 create cnf-files version-stamp 7960 May 11 2014 19:01:37
R2#sh run | s dial-peer voice
dial-peer voice 201 voip
 destination-pattern 40..$
 session protocol sipv2
 session target ipv4:10.10.160.5
 dtmf-relay sip-notify
 codec g711ulaw
 no vad
dial-peer voice 202 voip
 destination-pattern 2...$
 session protocol sipv2
 session target ipv4:192.168.1.1
 dtmf-relay rtp-nte
 no vad
dial-peer voice 203 voip
 incoming called-number .
 voice-class codec 1  
 dtmf-relay rtp-nte

 

and the configuration on R1 (CME) only look like this:

R1#sh run | s voice service
voice service voip
 ip address trusted list
  ipv4 10.10.0.0 255.255.0.0
 allow-connections h323 to h323
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily
 h323
R1#sh run | s dial-peer voice
dial-peer voice 100 voip
 destination-pattern 3...$
 session protocol sipv2
 session target ipv4:10.10.160.1
 dtmf-relay rtp-nte
 no vad
dial-peer voice 101 voip
 destination-pattern 4...$
 session protocol sipv2
 session target ipv4:10.10.160.1
 dtmf-relay rtp-nte
 no vad

Warm regards,

Johnny Kabundi.

 

 

 

Cisco Employee

Hello,Is the phone that you

Hello,

Is the phone that you transferred the call to a SIP Phone or a SCCP Phone?

We also need to revisit what is the transfer mechanism used in CUE . Collect the output of "show ccn subsystem sip" from CUE.

 

Regards,

Harshdeep

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