Call from External to AA

ricardorojas123 · ‎12-31-2009

I have a trunk SIP connected to CCME, this trunk SIP is used to process incoming and outgoing calls. The incoming and outgoing calls works fine. The problem is the AutoAttendant and VoiceMail. Internally the extensions can hear the Auto Attendant and when a extension does not answer, the prompt of the voice mail is heard, but from the external call to AutoAttendant, the prompt is not heard, the same for the voice mail, the prompts are not heard.

Topology:

(Router_2800) --->SIP trunk ----> (ISP)

This is the configuration:

voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8

dial-peer voice 1900 voip
description *InternalExtensions*
destination-pattern 555....
voice-class codec 1
session protocol sipv2
session target ipv4:172.17.30.1
dtmf-relay sip-notify

dial-peer voice 1901 voip
description ***CUE_Voicemail***
destination-pattern 1901
session protocol sipv2
session target ipv4:172.17.30.2
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 1902 voip
description *AutoAttendant*
destination-pattern 1902
voice-class codec 1
session protocol sipv2
session target ipv4:172.17.30.2
dtmf-relay sip-notify
no vad

dial-peer voice 2000 voip
tone ringback alert-no-PI
description *External Calls*
destination-pattern 9143T
voice-class codec 1
session protocol sipv2
session target ipv4:172.19.25.2
dtmf-relay rtp-nte

Marwan ALshawi · ‎12-31-2009

try to add the follwoing dial peer and see if it helps or not

dial-peer voice 90 voip
incoming called-number .

codec g711ulaw

good luck

if helpful Rate

ricardorojas123 · ‎12-31-2009

I change the settings to dial-peer called dial-peer voice 1902 voip (dial peer to the AA), codec and incoming number, but dont work .

ricardorojas123 · ‎01-04-2010

Another suggestion?

ricardorojas123 · ‎01-04-2010

I noticed that the calls between the internal IP Cisco Phone to the Auto Attendant in the Cisco Unity Express are established using the g711ulaw codec, and the codec used between the external call to internal IP Cisco Phone is g711alaw . Is possible change the codec g711ulaw to g711alaw in he connection from the CUCM to CUE ?

Example, actual codec used:

ITSP --> g711alaw --> (CUCME) ---> g711ulaw ---> Auto Attendant (CUE)

Cisco IP Phone registered in the CUCME ---> g711ulaw ----> Auto Attendant (CUE)

Marwan ALshawi · ‎01-04-2010

you can by using incoming voip dialpeer same concept of the above one

lets say from cucm ip phnes dial cue with number 1000

in the voice gateway where cue reside:

dial-peer voice 99 voip

incoming called-number 100

codec g711ulaw

and you already have outgoing dial-peer with the command destination pattern to send the the call to the right destination

good luck

jon.aril.antonsen · ‎10-13-2010

Hello ricardorajas123

Did you manage to solve this problem?

I have exactly the same problem as you had, and I'm pulling my hair on this one...

I hope you have an answer.

regards,

Jon

dksingh · ‎10-13-2010

CUE does not support g711alaw codec.

Only g711ulaw is the supported codec.

So hardcode dialpeers pointed to CUE

(VM/AA etc.) with g711ulaw and no vad.

For the other leg (incoming) if the codec is

other than g711ulaw, you'd need to configure

transcoding.

http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetrnsc.html

DK

jon.aril.antonsen · ‎10-13-2010

Hi Dilip.

I configured the transcode a bit earlier today, and now the call is forwarded to VM/AA. But the problem now is that I do NOT hear the VM/AA prompt...just silence. With debug I can see that the call is active.

Jon

dksingh · ‎10-13-2010

Hmm..I think u mean, you do *NOT* hear CUE prompts. The first thing you want to check is the RTP legs and if

xcoding is getting invoked. Following will help:

sh voip rtp connect

sh sccp connection

FYI, for ulaw<->alaw conversion you'd need universal xcoders:

http://www.cisco.com/en/US/docs/ios/12_4t/12_4t15/it_unitr.html

Another thing to note is for xcoding to be invoked, you need to hardcode codecs on both inbound and outbound

dialpeers rather than using a voice class-codec command.

Are u comfortable with capturing debugs and sharing your config in case we need to troubleshoot it further?

DK

jon.aril.antonsen wrote:

Hi Dilip.

I configured the transcode a bit earlier today, and now the call is forwarded to VM/AA. But the problem now is that I do hear the VM/AA prompt...just silence. With debug I can see that the call is active.

Jon

jon.aril.antonsen · ‎10-13-2010

Hi.

I will look into this right away.

It will be no problem sharing debugs and configs. This is just in a lab enviroment...so no secrets here :-)

When you say both in an put dial peers I rekin you refer to the VM/AA dial-peers?

I will post the result shortly :-)

Jon.

jon.aril.antonsen · ‎10-13-2010

Hi again.

This is what i got from the show commands. The commands issued while having active call to the AA PSTN number.

UC520#sh voip rtp conn
VoIP RTP active connections :
No. CallId     dstCallId LocalRTP RmtRTP LocalIP            RemoteIP
1   1337       1338       19512    22430 MY_OUTSIDE_IP     SIP_GW_IP
2   1338       1337       16384    20844 10.1.10.2          10.1.10.1
3   1339       1340       19304    2000   10.1.10.2          10.1.1.1
4   1341       1340       17082    2000   10.1.10.2          10.1.1.1
Found 4 active RTP connections

UC520#sh sccp conn
sess_id conn_id stype mode codec sport rport ripaddr

7 7 xcode sendrecv g711u 17722 2000 10.1.1.1
7 8 xcode sendrecv g711a 17490 2000 10.1.1.1

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

I'm also quite sure the codecs are configured correctly. But still just silence.

Jon.

ADAM CRISP · ‎10-13-2010

Hi

As you need to get an incoming SIP call routed as a SIP call to the CUE, also check to see if you have allowed this

i.e

voice sevice voip

allow connections sip to sip

I would also recommend configuring the CUE dial-peer as a b2bua.

If you are switching on the sip-sip feature, please make sure you don't inadventantly make your router into an open SIP gateway.

Adam

jon.aril.antonsen · ‎10-13-2010

Hi Adam.

Cheers for the input.

All the settings you mentioned are set, but no luck :-(

Jon

dksingh · ‎10-13-2010

Yep, xcoder is getting invoked it seems but those IP addr.

don't ring a bell in the absence of config.

If u can, please go ahead and capture following debugs for

one call from SIP trunk/PSTN to CUE/AA:

deb ccsip mess

deb ccsip err

deb voip ccapi inout

Note: Please capture debugs in a buffer ; configure following

before enabling/capturing debugs:

conf t

service time deb date msec

service sequence

no logg con

logg mon warn

logg buffer 5000000 debug

no logging rate

Issue clear log just before making the test call and once done

issue term len 0 and capture sh log output in a text file.

Please include calling/called num, call flow and config (sh run)

Pl. capture sh voip rtp connect and sh sccp connect for the same

call as well.

DK