Auto-Attendant Voice mail issue

johnroche_2 · ‎05-06-2011

I have a UC520 running an Auto attendant.

DDI is provided by skype.

When you call in on the DDI the AA plays the message, if you select option 1 it will ring an extension. All this works fine.

If the extension ring no answers to voicemail you hear silence?

any ideas

Darren DeCroock · ‎05-06-2011

John,

My first thought was that the voicemail message was just blank. So, is there an actual message recorded, and can you access and hear it internally?

Since this is coming in over a SIP trunk, are you using G711ulaw as your codec, or is that the one being negociated? Do you have transcoding setup? I would try forcing the unit to use G711ulaw as a test, since this is the codec that CUE (AA and VM) uses.

Thank you,

Darren

johnroche_2 · ‎05-08-2011

hi Darren,

the messages are definetly note silence.

I configured the skype sip trunk with cca. I believe by default cca set the codec t 711u.

The system is a uc520, I havent configured transcoding.

How do I force the codec?

David Trad · ‎05-08-2011

Hi John,

The system is a uc520, I havent configured transcoding.

I would strongly encourage you to configure transcoding on the system, and it could potentially solve this problem for you imediatly.

The only way I know how to do this in CCA is to enable conferencing capabilities, even if it is not used you still get the benefit of it and also the transcoding capabilities.

Is this a system completely configured via CCA or is this a Hybrid of CLI and CCA?

Cheers,

David.

Cheers, David Trad. **When you rate a persons post, you are indicating a thank you or that it helped, but at the same time you are also helping to maintain the community spirit - You don't have to rate posts and you wont be looked down upon :) *

johnroche_2 · ‎05-09-2011

Hi David,

I configured the box via CCA.

I tried to add conferencing but CCA prevented me due to lack of DSP resources.

Transcoding was configured (I guess by CA when the sip trunk was configured) I added a few more codec and made a test call again, Same issue, called AA option 1 which calls an extension, if the extension ring no answers, silence. Dont hear the greeting and no message.

I also set an option to call voicemail, when I do a test call to that silence also.

dspfarm profile 2 transcode

description CCA transcoding for SIP Trunk Skype for SIP

codec g711ulaw

codec g729abr8

codec g729ar8

codec g711alaw

codec g729r8

codec g729br8

maximum sessions 6

associate application SCCP

David Trad · ‎05-09-2011

Hi John,

Two things for you to look at if you don't mind...

Run the following command for me and post it up here: ROUTER# sh dsp all
Remove all the Codecs and just have G.711 uLaw enabled only, if Skype still works then there is no need for Transcoding and would like to see if the mode of operation stays the same or it operates exactly how you want it too

I am curious to see if all the DSP resources has been allocated to the SIP trunk, scary if that is the case because it means the UC runs out of resources too quickly and would require a larger PVDM or more of them.

You may need to consider contacting support and logging a case so they can do proper debugging, and just make sure that a PVDM hasnt dropped out into the ether, but the command above should show that anyway.

Cheers,

David.

Cheers, David Trad. **When you rate a persons post, you are indicating a thank you or that it helped, but at the same time you are also helping to maintain the community spirit - You don't have to rate posts and you wont be looked down upon :) *

johnroche_2 · ‎05-10-2011

Here is the show DSP farm all

Dspfarm Profile Configuration

Profile ID = 2, Service = TRANSCODING, Resource ID = 1

Profile Description : CCA transcoding for SIP Trunk Skype for SIP

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 : 6

Number of Resource Available : 6

Codec Configuration

Codec : g711ulaw, Maximum Packetization Period : 30

Codec : g729abr8, Maximum Packetization Period : 60

Codec : g729ar8, Maximum Packetization Period : 60

Codec : g711alaw, Maximum Packetization Period : 30

Codec : g729r8, Maximum Packetization Period : 60

Codec : g729br8, Maximum Packetization Period : 60

SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

0 2 26.8.1 UP N/A FREE xcode 1 - - -

Total number of DSPFARM DSP channel(s) 6

The transcode profile has 6 sessions configured.

I tried removing the codecs, Skype needs g729r8

I included a debug ccsip calls.

The inbound call negotiates G729r8 when it diverts to CUE it negotiates g711u. But caller gets silence.

I also tried reducing the number of transcode sessions, CCA still prevents setting up conferencing. I tried to set it up via CLI but max session option can only be 0.

015966: May 10 12:36:57.893 GMT: //527/895FECB38150/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x8815B868

State of The Call : STATE_DEAD

TCP Sockets Used : NO

Calling Number : Anonymous

Called Number : 99051000000666

Source IP Address (Sig ): 93.107.101.99

Destn SIP Req Addr:Port : 193.120.218.68:5060

Destn SIP Resp Addr:Port : 193.120.218.68:5060

Destination Name : 193.120.218.68

015967: May 10 12:36:57.893 GMT: //527/895FECB38150/SIP/Call/sipSPIMediaCallInfo

:

Number of Media Streams: 1

Media Stream : 1

Negotiated Codec : g729r8

Negotiated Codec Bytes : 20

Nego. Codec payload : 18 (tx), 18 (rx)

Negotiated Dtmf-relay : 6

Dtmf-relay Payload : 101 (tx), 101 (rx)

Source IP Address (Media): 93.107.101.99

Source IP Port (Media): 17746

Destn IP Address (Media): 193.120.218.68

Destn IP Port (Media): 26994

Orig Destn IP Address:Port (Media): [ - ]:0

015968: May 10 12:36:57.893 GMT: //527/895FECB38150/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC) : 16

Disconnect Cause (SIP) : 200

015969: May 10 12:36:57.905 GMT: //536/9985D594815B/SIP/Call/sipSPICallInfo:

The Call Setup Information is:

Call Control Block (CCB) : 0x88145BC8

State of The Call : STATE_DEAD

TCP Sockets Used : NO

Calling Number : Anonymous

Called Number : 399

Source IP Address (Sig ): 10.1.10.2

Destn SIP Req Addr:Port : 10.1.10.1:5060

Destn SIP Resp Addr:Port : 10.1.10.1:5060

Destination Name : 10.1.10.1

015970: May 10 12:36:57.905 GMT: //536/9985D594815B/SIP/Call/sipSPIMediaCallInfo

:

Number of Media Streams: 1

Media Stream : 1

Negotiated Codec : g711ulaw

Negotiated Codec Bytes : 160

Nego. Codec payload : 0 (tx), 0 (rx)

Negotiated Dtmf-relay : 8

Dtmf-relay Payload : 0 (tx), 0 (rx)

Source IP Address (Media): 10.1.10.2

Source IP Port (Media): 17876

Destn IP Address (Media): 10.1.10.1

Destn IP Port (Media): 20806

Orig Destn IP Address:Port (Media): [ - ]:0

015971: May 10 12:36:57.905 GMT: //536/9985D594815B/SIP/Call/sipSPICallInfo:

Disconnect Cause (CC) : 16

Disconnect Cause (SIP) : 200

David Trad · ‎05-10-2011

Hi John,

It would seem that CCA has configured all available DSP resources to the SIP trunk... Whilst bizarre and a little odd I guess that is the way it is instructed to do it.

Assuming you have an SBS contract on this system, I would suggest having them investigate this issue and resolve it for you quickly, I can see what the problem is but any instructions provided would most likely take you out of support scope as it will require some CLI modification for testing purposes and may be too hard to explain on the forums.

Log a case, spend maybe a couple of hours with them and get it resolved it will be quicker than trying to work your way through it here.. Unless you do not have an SBS contract at which point we can talk about remote access to the system and do some further tests on it.

Cheers,

David.

Cheers, David Trad. **When you rate a persons post, you are indicating a thank you or that it helped, but at the same time you are also helping to maintain the community spirit - You don't have to rate posts and you wont be looked down upon :) *