06-04-2014 09:40 AM - edited 03-16-2019 10:59 PM
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.
06-04-2014 11:46 AM
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
06-04-2014 12:56 PM
Hello,
Yes, I do have transcoding configured with all the required codec as per your description above.
What else could be the problem?
JK.
06-04-2014 05:41 PM
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.
06-04-2014 07:56 PM
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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