CME to SfB Integration-Issue with intersite calls

[afsal abdul gafoor]

Dear Team,

Kindly note that we are facing issues in inter site calls  , from SfB to CME.  Please note that all other calls , like internal,PSTN calls working well from SfB.Issue persits only for intersite calls,which is with the below call flow.

SfB --> CME --> ICT -->CUCM.

I will copy the relevent configuration here, and from the SfB side user will be dialing 4 as prefix for reaching CME. site access code from CME to CUCM is 63XXXX.

voice translation-rule 13
 rule 1 /^4\(.*\)/ /\1/
 rule 2 /\+971/ /9/
 rule 4 /^\+/ /900/

voice translation-rule 12
 rule 1 /^.*\(....\)/ /\1/

voice translation-profile fromSfB
 translate calling 12
 translate called 13

dial-peer voice 9 voip
 description fromSfB
 translation-profile incoming fromSfB
 preference 1
 destination-pattern .T
 b2bua
 session protocol sipv2
 session target ipv4:XXXX:5060
 session transport tcp
 incoming called-number .T
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad

I will attach the logs for a working call (SfB to CME ) and also non working one. (SfB to CUCM through CME).

Appreciate for any suggestions.

[afsal abdul gafoor]

Please find the attached files

[edwardforgacs]

Not sure about your debugs, but we have had direct CME to SfB integration working for quite some time. Our dial peers for SfB have a lot of other commands/hacks on them which I am happy to share, but I don't have on hand the reasoning for every single one of them.

Of particular note is that for SfB 2015 it had to be changed to go to port 5068

description Skype 2015
 translation-profile incoming Lync-Out
 translation-profile outgoing Lync-In
 destination-pattern 2..
 notify redirect ip2pots
 media flow-around
 media forking
 b2bua
 rtp payload-type comfort-noise 13
 session protocol sipv2
 session target ipv4:XXXX:5068
 session transport tcp
 incoming uri via LYNC
 no voice-class sip call-route url
 voice-class sip profiles 5
 voice-class sip block 183 sdp absent
 voice-class sip options-keepalive
 dtmf-relay rtp-nte
 codec g711alaw
 fax rate disable
 fax protocol none
 no vad
 supplementary-service h450.12
 no supplementary-service sip refer

voice class sip-profiles 5
 request ANY sdp-header Audio-Attribute modify "a=(.*)AES_CM_128_HMAC_SHA1_(.*) inline(.*)\|(.*):(.*)" "a=\1CRYPTO_UNKNOWN inline\3|\4:\5"
 response ANY sdp-header Audio-Attribute modify "a=(.*)AES_CM_128_HMAC_SHA1_(.*) inline(.*)\|(.*):(.*)" "a=\1CRYPTO_UNKNOWN inline\3|\4:\5"

[afsal abdul gafoor]

Thanks for your coments .. I will be going through the configuration you provided. Please note that for me also integration works normally. I can make calls from SfB end points to CME end points. the problem I am facing is specific to intersite calls from SfB through CME

[R0g22]

CME and CUCM are doing sip or h323 ?
Are there any phones registered on cme router ?

[afsal abdul gafoor]

CME to CUCM is through ICT -h323 non gatekeeper controlled trunk. Yes phones registered in CME , and I can make calls from SfB to CME and also vise versa through SNR.

[R0g22]

Share a "show run" from the CME please. Your debugs are not complete. Take a log for the following debugs -

debug ccsip message
debug ccsip error
debug voice ccapi inout
debug h225 asn1
debug h245 asn1

Share the calling/called numbers.

[afsal abdul gafoor]

Attached is the configuration, please note I cannot take live test now and infact I am getting the same 404 error when we were testing the internal calls.

[R0g22]

Logs for the requested debugs ?

[afsal abdul gafoor]

Please find the attached log.

calling party : +97148049122 (from SfB)

called party: 638742 (from CUCM)

[R0g22]

Checked the logs. So the IOS does the select a outbound DP to IOS and disconnects with CV=3 which is "No Router to Destination".

Feb 15 16:10:26.850: //17503/91CDB709A07C/CCAPI/ccCallDisconnect:
Cause Value=3, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
Feb 15 16:10:26.850: //17503/91CDB709A07C/CCAPI/ccCallDisconnect:
Cause Value=3, Call Entry(Responsed=TRUE, Cause Value=3)

First, I don't see a dial-peer to accomodate the destination i.e. "dest=4638742". The reason for CV=3 is that you do not have "allow-connections sip to h323" under "voice service voip".

You need to work on both of these and your calls should work if there aren't any additional issues.

[afsal abdul gafoor]

Hello

Please note there is dial peer for 63.... and the digit 4 will be striped
of in the dial peer. Same way I could call 49... sucessfully

[R0g22]

There isn't a dial-peer for 63..... There is this one though -

dial-peer voice 1000 voip
corlist outgoing CALL-INTERNAL
translation-profile outgoing OUTGOING_CALLINGPARTY_INTERSITE
destination-pattern 6.....
session target ipv4:10.25.XXX
incoming called-number .T
dtmf-relay rtp-nte
codec g711ulaw
no vad

Similarly there is nothing for a 49..... the best is

dial-peer voice 4 voip
corlist outgoing CALL-INTERNAL
destination-pattern 64....
session target ipv4:10.25.250.1
dtmf-relay rtp-nte
codec g711ulaw
no vad

None of these dial-peers will accommodate a call for "4638742". Please review your config.

[afsal abdul gafoor]

Nipun,

Thanks for your comments ,but please note there is dial peer for 6...., which should match 638742 (as it is the dial peer towards CUCM).

In the same incoming dial-peer, there is a voice translation profile to stripe off , the prefix 4. and what i meant is if I dial , 49... from  SfB the same dial-peer the call is accepting and 4 is striped off and its reaching to CME extension.

[R0g22]

Ok. First you need to add the allow-connection command if these calls need to go through the cme to cucm. Add that and then test. In case it still fails, enable "debug voip translation" in addition to the previous debugs and attach them in a text file.