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
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
I'm not able to access my old voice mail messages all of a sudden. The recording says something like 'the message is currently not available'. This has never happened before in all the years I have been using this system. I have t...