02-15-2018 12:30 AM - edited 03-17-2019 12:12 PM
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.
02-15-2018 12:33 AM
02-15-2018 01:33 AM
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"
02-15-2018 01:47 AM
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
02-15-2018 01:28 AM
02-15-2018 01:49 AM
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.
02-15-2018 05:00 AM
02-15-2018 07:29 AM
02-15-2018 07:53 AM
02-15-2018 08:14 AM
02-15-2018 09:54 AM
02-15-2018 10:12 AM
02-15-2018 10:53 AM
02-15-2018 11:19 AM
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.
02-15-2018 11:37 AM
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