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New Member

Unity Express - Incoming calls wont get voice mail

CUE works fine with telephones on my local network. Incoming and outgoing calls work fine.

However when I get an incoming call via SIP trunk the call will not get forwarded to unity express after 10 seconds. The line goes dead.

I searched for another post which suggested the following commands:

telephony-service

call-forward pattern .T

voice service voip

allow connections from h323 to sip

I've double checked them and there's still something wrong.

Here's my current configuration:

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

h323

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

!

telephony-service

load 7910 P00403020214

load 7960-7940 P00305000301

max-ephones 24

max-dn 24

ip source-address 192.168.20.1 port 2000

auto assign 1 to 24

system message Comtek

voicemail 3000

max-conferences 8 gain -6

call-forward pattern .T

moh music-on-hold.au

time-webedit

transfer-system full-consult

transfer-pattern 2...

transfer-pattern 3...

directory last-name-first

directory entry 2 2001 name Phone Two 7912

directory entry 3 2000 name Phone One 7970

!

ephone-dn 1 dual-line

number 2000 secondary 441833000000

call-forward busy 3000

call-forward noan 3000 timeout 10

no huntstop

!

ephone 1

no multicast-moh

device-security-mode none

mac-address 0017.0EF0.3642

type 7970

button 1:1

!

So pros, any suggestions?

Thanks

16 REPLIES
New Member

Re: Unity Express - Incoming calls wont get voice mail

What's your dial-peers for CUE look like?

Dan

New Member

Re: Unity Express - Incoming calls wont get voice mail

Sorry, that was a stupid thing to leave out. But as I say, I get forwarded through to voicemail with IP Phones on my network. My IP Phones have extension numbers of 2xxx and all the CUE extensions are 3xxx, with voicemail specifically being 3000

dial-peer voice 2000 voip

description Outgoing Call via SIP

translation-profile outgoing SIPout

destination-pattern 9T

voice-class codec 1

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

!

dial-peer voice 3000 voip

destination-pattern 3...

session protocol sipv2

session target ipv4:192.168.10.11

dtmf-relay sip-notify

codec g711ulaw

no vad

!

New Member

Re: Unity Express - Incoming calls wont get voice mail

Do you have a route to the CUE module?

What is the Extension you are pointing CUE at?

Re: Unity Express - Incoming calls wont get voice mail

Which dial-peer is your incoming dial-peer when the call comes into CME via SIP. Note that your incoming call leg if not one specified (using incoming-called number or answer-address) may match the default dial-peer 0 and that uses codec of g729r8. When the call comes in , do a "sh call active voice brief" to see which dial-peer is getting matched. If 2000 is not getting matched as the incoming call leg, then there lies the problem. CUE supports only g711 (your dial peer voice 3000 looks good). There is no transcoding support between two SIP call legs which use different codecs (as of today). I think this may come in the future, but as long as you use g711 on both legs, this call should work.

HTH

Sankar

PS: please remember to rate posts!

New Member

Re: Unity Express - Incoming calls wont get voice mail

I made a new dial-peer to handle incoming calls as follows.

dial-peer voice 1000 voip

description Incoming SIP

translation-profile incoming SIPin

voice-class codec 1

session protocol sipv2

incoming called-number .T

dtmf-relay rtp-nte

no vad

The translation-profile puts the call through to my 2000 extension.

This is my "show call active voice brief" when an external incoming call is ringing through to my 2000 ephone-dn.

To me this seems to show the dial-peer "1000" matching and using the g711ulaw codec

Telephony call-legs: 1

SIP call-legs: 1

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 2

1715 : 552 596706500ms.1 +-1 pid:1000 Answer +441833696807 connecting

dur 00:00:00 tx:0/0 rx:0/0

IP 87.127.240.98:16188 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off

media inactive detected:n media contrl rcvd:n/a timestamp:n/a

long duration call detected:n long duration call duration:n/a timestamp:n/a

1715 : 553 596706510ms.1 +-1 pid:20001 Originate 2000 connecting

dur 00:00:00 tx:0/0 rx:0/0

Tele 50/0/1 (553) [50/0/1.0] tx:0/0/0ms None noise:0 acom:0 i/0:0/0 dBm

Telephony call-legs: 1

SIP call-legs: 1

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 2

This is the "show call active voice brief" for an external incoming call when the call is established.

Telephony call-legs: 1

SIP call-legs: 1

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 2

1731 : 569 597220040ms.1 +3730 pid:1000 Answer +441833696807 active

dur 00:00:02 tx:105/16800 rx:104/16640

IP 87.127.240.98:15162 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off

media inactive detected:n media contrl rcvd:n/a timestamp:n/a

long duration call detected:n long duration call duration:n/a timestamp:n/a

1731 : 570 597220060ms.1 +3700 pid:20001 Originate 2000 active

dur 00:00:02 tx:0/0 rx:105/16800

Tele 50/0/1 (570) [50/0/1.0] tx:16180/16180/0ms g711ulaw noise:0 acom:0 i/0:0/0 dBm

Telephony call-legs: 1

SIP call-legs: 1

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 2

Not too sure where to go from here.

New Member

Re: Unity Express - Incoming calls wont get voice mail

Hi,

I'd try a 'debug voip dialpeer' to make sure when the phone transfers it is in fact hitting the correct dialpeer.

You can try calling the voicemail on number 3000 direct from a phone to rule out any transfers and so on.

The other reason may be that you don't have a voice mail box setup on the unity express for the user,

thanks,

Mark

New Member

Re: Unity Express - Incoming calls wont get voice mail

Thanks for your help Mark.

I can dial voicemail "3000" fine from any phone on my network. I can leave and retrieve voice messages without any problems.

I have attached the "debug voip dialpeer" when making an external incoming call, attempting to go through to voicemail (Dial-PeerExt.txt).

A couple of things concern me with that output:

1.Matching two dialpeers initially 20001, and 609996. I made 609996 myself, 20001 was made by CME.

Sep 4 10:36:05.353: //-1/00B1A9CB85B0/DPM/dpMatchPeersMoreArg:

Result=SUCCESS(0)

List of Matched Outgoing Dial-peer(s):

1: Dial-peer Tag=20001

2: Dial-peer Tag=600026

Here are the dial-peers in question.

dial-peer voice 600026 voip

destination-pattern 2000

voice-class codec 1

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

no vad

dial-peer voice 20001 pots

destination-pattern 2000$

call-forward busy 3000

call-forward noan 3000

progress_ind setup enable 3

port 50/0/1

As I didnt make the 20001 dial-peer i had to "show telephony-service dial-peer". Why does CME create these? is there any way i can get rid of it? and will it effect me adversly if i did?

2. The calling number disappears - "Calling Number=,"

For comparison i'll attach the "debug voice dialpeer" output when making an internal call to voicemail which works (Dial-PeerInt.txt).

Thanks

New Member

Re: Unity Express - Incoming calls wont get voice mail

Hi,

According to the debug, when a call comes in from outside, the number it's looking for is:

441833609996

This is the telco adding digits. You have a choice to digit strip on the way in to make it look like your dial peer (using a translation rule), or change the destination pattern in your dial-peer to be the number above.

I'd be confident of the second option working.

Give it a try, do another debug and let me know,

ta,

Mark

New Member

Re: Unity Express - Incoming calls wont get voice mail

This is what I have for my incoming dial-peer.

dial-peer voice 1000 voip

description Incoming SIP

translation-profile incoming SIPin

voice-class codec 1

session protocol sipv2

incoming called-number .T

dtmf-relay rtp-nte

no vad

With the following translation rules:

voice translation-profile SIPin

translate called 3

translate calling 4

voice translation-rule 3

rule 1 /^441833609996/ /2000/

voice translation-rule 4

rule 1 /44/ /0/

This translation-rule directs the call to my ephone-dn. If i changed that 2000 to the voicemail 3000 it goes straight through to voicemail. That is confusing me more, i've no idea why it can work like that but not when the call gets no answer or is busy.

I'm also not quite sure what you want me to do with the destination-pattern or in which dial-peer i should enter that.

Thanks again for your help Mark.

New Member

Re: Unity Express - Incoming calls wont get voice mail

So, in summary:

You call from outside to the number: 441833609996

You have a dial peer 1000, which is matching everything and applying a translation rule.

It is translating the number called to be 2000.

The number 2000 is then matched by dial peer 20001.

When the incoming call hits dial peer 20001, it is then diverted to 3000.

Number 3000 is matched by dial-peer 3000. The called number then becomes +01833699997

List of Matched Outgoing Dial-peer(s):

1: Dial-peer Tag=3000

Sep 4 10:36:15.365: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Calling Number=, Called Number=+01833699997, Peer Info Type=DIALPEER_INFO_SPEECH

Sep 4 10:36:15.365: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=+01833699997

For internal calls:

You are calling from number 2000. It is matching dial peer 20001, as you have dialled the number 3000 directly.

The called number remains 3000.

Can you list dial-peer 3000 and we can take a look at that? Basically it looks like the call works ok until it diverts.

When it matches the voice mail dial peer, it then changes the calling number to be the original number.

I'm not sure why the called number reverts to be the original calling number at that point.

I might be running out of ideas here, but list the dial-peer 3000 and see what it looks like,

thanks,

Mark

New Member

Re: Unity Express - Incoming calls wont get voice mail

Thanks again Mark,

here is my dial-peer 3000 as requested.

dial-peer voice 3000 voip

destination-pattern 3...

session protocol sipv2

session target ipv4:192.168.10.11

dtmf-relay sip-notify

codec g711ulaw

no vad

New Member

Re: Unity Express - Incoming calls wont get voice mail

Hi,

I have also done a debug.. and got the same result you got:

Sep 5 09:10:20.855: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=1699

Sep 5 09:10:20.855: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Result=Success(0) after DP_MATCH_DEST

Sep 5 09:10:20.855: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

Result=SUCCESS(0)

List of Matched Outgoing Dial-peer(s):

1: Dial-peer Tag=200

Sep 5 09:10:20.855: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Calling Number=, Called Number=47221699, Peer Info Type=DIALPEER_INFO_SPEECH

Sep 5 09:10:20.855: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=47221699

Sep 5 09:10:20.855: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Result=Success(0) after DP_MATCH_DEST

Sep 5 09:10:20.855: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:

Result=SUCCESS(0)

List of Matched Outgoing Dial-peer(s):

1: Dial-peer Tag=201

Sep 5 09:10:20.859: //-1/30C13A21800F/DPM/dpMatchPeersCore:

Calling Number=, Called Number=47221699, Peer Info Type=DIALPEER_INFO_SPEECH

Sep 5 09:10:20.859: //-1/30C13A21800F/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=47221699

Sep 5 09:10:20.859: //-1/30C13A21800F/DPM/dpMatchPeersCore:

Result=Success(0) after DP_MATCH_DEST

Sep 5 09:10:20.859: //-1/30C13A21800F/DPM/dpMatchPeersMoreArg:

Result=SUCCESS(0)

List of Matched Outgoing Dial-peer(s):

1: Dial-peer Tag=201

I have the 'dialplan pattern' command on my setup, which does in the expansion.

In addition to the internal dial-peer I also have an external dial peer to match the expanded number. I think in your case this would be something like:

dial-peer voice 3001 voip

destination-pattern 01833699997

session protocol sipv2

session target ipv4:192.168.10.11

dtmf-relay sip-notify

codec g711ulaw

no vad

Try adding this, then do another debug voip dialpeer to see it recognises the number. I notice it has a leading + sign on the called number in your debug - I'm not sure if you will also have to add that,

Let me know how you go

Mark

New Member

Re: Unity Express - Incoming calls wont get voice mail

I've added that dial-peer and I had to use this destination-pattern +441833699997, to get it to recognise. I've attached the debug dialpeer output.

The line still goes dead when it tries to transfer to voicemail and it doesnt quite look like yours. Mine says it is an incoming dial-peer, while yours is outgoing.

Thanks again Mark, i really do appreciate your help.

New Member

Re: Unity Express - Incoming calls wont get voice mail

Hello,

It's still not matching the dial peer is the problem. I've included the output below - its changing the called number to +01833699997, and repeatedly trying (and failing) to match a dial peer to that. I was hoping the dial peer 3001 would match, but I guess the destination-pattern in that isn't a match.

The link below details the options:

http://www.cisco.com/en/US/partner/products/sw/iosswrel/ps5207/products_command_reference_chapter09186a00801a7f23.html#wp1459870

Try the destination pattern as:

destination-pattern +01833699997

or

destination-pattern ^99997

I take your point about the dial peers you have - they do seem to be different from those I have, so maybe this is a little more fundamental.

See how you go with the above couple of patterns. Bear in mind that on the unity express you will need to configure the mail box for the e.164 number and the extension.

Happy to help,

Mark

List of Matched Outgoing Dial-peer(s):

1: Dial-peer Tag=3000

Sep 5 11:54:00.586: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Calling Number=, Called Number=+01833699997, Peer Info Type=DIALPEER_INFO_SPEECH

Sep 5 11:54:00.586: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

Match Rule=DP_MATCH_DEST; Called Number=+01833699997

Sep 5 11:54:00.586: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:

No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)

Sep 5 11:54:00.586: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:

Result=NO_MATCH(-1)

New Member

Re: Unity Express - Incoming calls wont get voice mail

Thanks again Mark.

I had been discussing this with a helpful cisco guy through a TAC request.

He got it working with a "hack" which makes it work unintentionally via, "service session" in my incoming dial-peer. I've no idea why this works but it does.

He said this failed because CME does not support local hairpin of such calls and sends:

Sent:

SIP/2.0 302 Moved Temporarily

This is not understood by the SIP trunk side and just hangs. He also said this will be supported later on next year.

So my problem was that SIP trunking does not support local hairpin for call forward and transfer.

What is a local hairpin? and is there a correct way to do this without the local hairpin?

Again, thanks for your help mark. I do appreciate it.

David

New Member

Re: Unity Express - Incoming calls wont get voice mail

HI,

Sorry I couldn't help. Hope the delay hasn't held you back.

My understanding is that the incoming non-voip call is terminated and then forwarded on as a voip call.

The only note I had on this is that a TCL script may be required to allow non-voip transfers to take place.

I'm not expert on this side of it, so would need to turn it over to the wider community for input - anyone else with a better explanation?

Best regards,

Mark

http://www.cisco.com/en/US/partner/products/sw/voicesw/ps4625/products_configuration_guide_chapter09186a00806a803d.html#wp1056015

"Hairpin call routing uses the VoIP-to-VoIP connection mechanisms that were introduced in Cisco CME 3.1 to transfer and forward calls that cannot use H.450 standards. When a call that originally terminated on a voice gateway is transferred or forwarded by a phone or other application attached to the gateway, the gateway reoriginates the call and routes the call as appropriate, making a VoIP-to-VoIP, or hairpin, connection. This approach avoids any protocol dependency on the far-end transferred-party endpoint or transfer-destination endpoint. Hairpin routing of transferred and forwarded calls also causes the generation of separate billing records for each call leg, so that the transferred or forwarded call leg is typically billed to the user who initiates the transfer or forward. "

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