07-27-2015 02:53 AM - edited 03-17-2019 03:46 AM
Hello Team,,
We are facing the issue with Cisco CME with call manager .
If we are trying the SIP trunk calls from call manager not going to cme ,from cme to call manager sip call is connecting but there is no audio.
with the same setup H.323 calls going on both the way without any issue.
Configuration Done.
SIP trunk and route pattern configured and pointed to the CME ip addres..
Dial-peer has been configured in the cme router.
Any thing is missing and suggestion please...
08-01-2015 05:55 AM
Thanks for your support,
The same setup h.323 call is working normally with sccp endpoints,
We have already check with network team they are saying there is no blocking in between the cucm and cme.
Is there any option we can prove the network issue ?
Also please verify the sh run for the CME.
Thanks
08-01-2015 06:09 AM
SIP and H.323 uses different network ports and hence working one protocol through firewall doesn't mean that other will work too.
Anyway irrespective of your CME configuration, if call from CUCM is not hitting CME, there must be some issue in network. That is what I can think of.
One thing you can verify as you have bind the dial peer for inbound call leg to UDP (session transport udp), you can cross verify in CUCM, SIP trunk security profile has transport set to UDP for outgoing messages.
Last thing to check your network scenario... drop CUCM for a time being. Take any softphone like Xlite and check some messages like REGISTER, INVITE etc to see if those messages are reaching CME. Ensure that you set the UDP port to 5060 in Xlite as it used random source port by default.
08-02-2015 12:12 AM
Hello Vivek,
One thing you can verify as you have bind the dial peer for inbound call leg to UDP (session transport udp), you can cross verify in CUCM, SIP trunk security profile has transport set to UDP for outgoing messages.
Yes verified in sip profile UDP is using for sip out going messages.
Last thing to check your network scenario... drop CUCM for a time being. Take any softphone like Xlite and check some messages like REGISTER, INVITE etc to see if those messages are reaching CME. Ensure that you set the UDP port to 5060 in Xlite as it used random source port by default.
We have registered on branch(cme site) SIP Ip phone directly in the HO call manager,but the same issue is happening.(from Head office to branch office call is not going but from branch office to head office call is connecting but there is no audio and disconnecting)
Also will try with x lite soft phones.
Thanks
08-01-2015 05:56 AM
Thanks for your support,
The same setup h.323 call is working normally with sccp endpoints,
We have already check with network team they are saying there is no blocking in between the cucm and cme.
Is there any option we can prove the network issue ?
Also please verify the sh run for the CME.
Thanks
08-01-2015 07:41 AM
sh ver
sh inv
I think u need to configure dspfarm
08-02-2015 12:02 AM
08-02-2015 12:36 AM
configure dspfarm
voice-card 0
dspfarm
dsp services dspfarm
voice rtp send-recv
sccp local Loopback0
sccp ccm (Loopback0) identifier 1 priority 1 version 7.0
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register CME-TRANSC
!
dspfarm profile 1 transcode universal
codec g711alaw
codec g711ulaw
maximum sessions 5
associate application SCCP
!
08-02-2015 12:52 AM
08-02-2015 03:02 AM
interface Loopback0
description " Router ID"
ip address 172.18.100.3 255.255.255.255
sccp local Loopback0
no sccp ccm 172.16.5.100 identifier 1 priority 1 version 7.0
sccp ccm 172.18.100.3 identifier 1 priority 1 version 7.0
telephony-service
protocol mode ipv4
sdspfarm units 5
sdspfarm transcode sessions 4
sdspfarm tag 1 CME-TRANSC
exi
wr
sh gspf all
08-02-2015 03:24 AM
configured,still the same issue,call is not going from the call manager.
please find the below sh dspf all
MOE-Jalalabad-RTR#sh dspf all
Dspfarm Profile Configuration
Profile ID = 1, Service =Universal TRANSCODING, Resource ID = 1
Profile Description :
Profile Service Mode : Non Secure
Profile Admin State : DOWN
Profile Operation State : DOWN
Application : SCCP Status : NOT ASSOCIATED
Resource Provider : FLEX_DSPRM Status : NONE
Number of Resource Configured : 5
Number of Resources Out of Service : 5
Codec Configuration: num_of_codecs:4
Codec : g711ulaw, Maximum Packetization Period : 30
Codec : g711alaw, Maximum Packetization Period : 30
Codec : g729ar8, Maximum Packetization Period : 60
Codec : g729abr8, Maximum Packetization Period : 60
Total number of DSPFARM DSP channel(s) 0
Also find the attached sh run...
08-02-2015 03:32 AM
conf t
dspfarm profile 1 transcode universal
no shut
exit
wr
sh dspf all
08-02-2015 03:36 AM
updated ,sorry still the same issue,
MOE-Jalalabad-RTR#sh dspf all
Dspfarm Profile Configuration
Profile ID = 1, Service =Universal TRANSCODING, Resource ID = 1
Profile Description :
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 : 5
Number of Resources Out of Service : 0
Codec Configuration: num_of_codecs:4
Codec : g711ulaw, Maximum Packetization Period : 30
Codec : g711alaw, Maximum Packetization Period : 30
Codec : g729ar8, Maximum Packetization Period : 60
Codec : g729abr8, Maximum Packetization Period : 60
SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID PKTS_TXED PKTS_RX ED
0 1 37.2.0 UP N/A FREE xcode 1 - - -
0 1 37.2.0 UP N/A FREE xcode 1 - - -
0 1 37.2.0 UP N/A FREE xcode 1 - - -
0 1 37.2.0 UP N/A FREE xcode 1 - - -
0 1 37.2.0 UP N/A FREE xcode 1 - - -
08-02-2015 03:59 AM
ok. now dspfarm is UP
debug ccsip mess
08-02-2015 04:12 AM
08-02-2015 04:33 AM
Received:
INVITE sip:8@172.18.3.129;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.18.3.138:5060;branch=z9hG4bK361ab983
From: "5002" <sip:5002@172.18.3.129>;tag=e0d173e58d8a001d4cd79cda-29d1b8e6
To: <sip:8@172.18.3.129>
========================
calling number 5002
called number 8
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