11-14-2014 03:09 AM - edited 03-17-2019 12:56 AM
Hello,
we are facing ringback issue when inbound external call to the cisco phone is transferred back to another extenal number.
Call Flow: Cisco phone----CUCM-----SIP-----CUBE----SIP----ITSP.
Investigation done:
- if both PRACK & EO are disabled, we hear the ringback on blind call transfer. In fact it is continuous ringback tone. Though the other end rejects the call, we hear this continuous ringback. Also, we don't hear any announcement played from provider side.
- If PRACK enabled & EO Disabled, same as above
- if PRACK disabled & EO enabled, same as above
- if both either PRACK and Early Offer are enabled, we don't hear the ringback during blind call transfer but we hear announcement from provider
To address both the continuous ringback & provider announcement issues, we enabled the MTP on the SIP Trunk in CUCM to CUBE.
Is there anything we can do to address these issues instead of enabling MTP on the sip trunk for all calls?
I also tried resetting the ANN and assigning dedicated ANNs in the SIP Trunk MRGL.
any help would be much appreciated. Thanks.
Solved! Go to Solution.
11-14-2014 11:51 AM
Suresh,
This appears to be a provider issue from the description. However you can do MIDCALL INVITE/UPDATE consumption, so you do not send re-INVITE to your provider which somehow is allowing them to play announcements..(your CUBE IOS needs to support this feature)
voice service voip
sip
mid-call signaling passthru media-change
Test this and let us know how it goes.
11-14-2014 11:51 AM
Suresh,
This appears to be a provider issue from the description. However you can do MIDCALL INVITE/UPDATE consumption, so you do not send re-INVITE to your provider which somehow is allowing them to play announcements..(your CUBE IOS needs to support this feature)
voice service voip
sip
mid-call signaling passthru media-change
Test this and let us know how it goes.
11-14-2014 11:56 AM
Thanks Deji, I'll try that command.
What would be the settings (PRACK & EO) in the SIP profile?
11-14-2014 11:58 AM
Ideally you should enable PRACK and enable early offer support for voice and video (insert mtp if needed)
11-14-2014 12:20 PM
You are the SIP master Deji. Thanks a ton. That worked :)
11-14-2014 01:13 PM
He definitely is. :)
11-14-2014 01:20 PM
Undoubtedly ;-)
11-14-2014 01:39 PM
George,
I am in your shadows!!! ( Still read a thread of yours today..) :)
11-14-2014 01:42 PM
LOL, here I was thinking where you were. :)
11-14-2014 01:40 PM
Glad to help Suresh :) (don't make my head swell too much)
11-17-2014 03:36 AM
Hi Deji, I think I made a mistake when testing the call day before yesterday. seems I missed out to uncheck the MTP in the SIP Trunk and made changes only to SIP Profile (EO & PRACK enabled).
When I crosschecked this today, I found the MTP was checked. when I unchecked and tried the test calls, found that the ringback issue still persists. Sorry about the wrong update.
any idea how to proceed further?
11-17-2014 03:41 AM
This suggests that the re-INVITE consumption is not working well or has not been configurd properly. It does similar thing as the MTP does. Stops re-INVITE from going out to ITSP.
Please do a test call and send us
debug ccsip messages (include calling and called number and describe what you get) Also send sh run of your cube
11-17-2014 04:51 AM
Yes, I see the midcall invites are going out to ITSP. I had the midcall-signaling passthru media-change configured in global level first that didn't help. also configured it on the dial-peer level, still no luck.
I also tried midcall-signaling block in the dial-peer level that is also not working.
CUBE is 2921 with Version 15.3(3)M.
voice service voip
mode border-element
allow-connections sip to sip
redirect ip2ip
fax protocol pass-through g711ulaw
sip
header-passing
asserted-id pai
asymmetric payload dtmf
midcall-signaling passthru media-change
early-offer forced
g729 annexb-all
sip-profiles 1
!
voice class codec 1
codec preference 1 g711ulaw
!
voice class sip-profiles 1
request INVITE sip-header Min-SE remove
request INVITE sip-header Unsupported modify "Unsupported:" "timer"
request INVITE sip-header Remote-Party-ID remove
response 200 sip-header Remote-Party-ID remove
request INVITE sip-header P-Asserted-Identity modify ">" ";user=phone>"
request INVITE sip-header From modify ">" ";user=phone>"
request INVITE sip-header Min-SE remove
request ANY sip-header Allow-Header modify "UPDATE, " ""
response ANY sip-header Allow-Header modify "UPDATE, " ""
!
!
dial-peer voice 100 voip
description ** Incoming Calls From CUCM **
answer-address .T
voice-class codec 1
voice-class sip g729 annexb-all
voice-class sip early-offer forced
voice-class sip profiles 1
no voice-class sip midcall-signaling passthru media-change
voice-class sip midcall-signaling block
dtmf-relay rtp-nte
dtmf-interworking standard
fax-relay ecm disable
fax-relay sg3-to-g3
fax protocol pass-through g711ulaw
ip qos dscp af21 signaling
no vad
!
dial-peer voice 101 voip
description ** International Outbound Calls to TELUS **
destination-pattern 011T
session protocol sipv2
session target ipv4:172.27.50.23
voice-class codec 1
voice-class sip g729 annexb-all
voice-class sip early-offer forced
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
no voice-class sip midcall-signaling passthru media-change
voice-class sip midcall-signaling block
dtmf-relay rtp-nte
fax-relay ecm disable
fax-relay sg3-to-g3
fax protocol pass-through g711ulaw
ip qos dscp af21 signaling
no vad
!
dial-peer voice 102 voip
description ** National Outbound Calls to TELUS **
destination-pattern 1..........
session protocol sipv2
session target ipv4:172.27.50.23
voice-class codec 1
voice-class sip g729 annexb-all
voice-class sip early-offer forced
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
no voice-class sip midcall-signaling passthru media-change
voice-class sip midcall-signaling block
dtmf-relay rtp-nte
fax-relay ecm disable
fax-relay sg3-to-g3
fax protocol pass-through g711ulaw
ip qos dscp af21 signaling
no vad
!
dial-peer voice 103 voip
description ** Local Outbound Calls to TELUS **
destination-pattern [2-9].........
session protocol sipv2
session target ipv4:172.27.50.23
voice-class codec 1
voice-class sip g729 annexb-all
voice-class sip early-offer forced
voice-class sip profiles 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
no voice-class sip midcall-signaling passthru media-change
voice-class sip midcall-signaling block
dtmf-relay rtp-nte
fax-relay ecm disable
fax-relay sg3-to-g3
fax protocol pass-through g711ulaw
ip qos dscp af21 signaling
no vad
!
we have 2 cubes with active-active connection. so in the nonworkingscr.txt is the one receives the call first from provider and nonworkingsteu.txt is the one you can see the 2nd call sent back to provider.
calling: 918066914754
called: 5816280216 (mask: 6048959000)
Transferred to 918066914714.
Note: These logs are captured when midcall signalling passthrough is configured only on global level. not in dial-peers.
keeping the global config, I tried the passthrough in dial-peer level, that didn't work, so removed it and added the blocking command in the same dial-peers. so you will see both in the config pasted above.
11-17-2014 08:48 AM
Your MIDCALL consumption is definitely working. If you look at the logs, there are several re-INVITEs between 157.171.4.27 and 153.112.89.13, which were not sent to the ITSP. There is only one re-INVITE sent to the ITSP which is the final part to connect the transferred endpoint.
So what do you get with the mid call consumption in place? Do you still get announcement from the provider?
11-17-2014 08:59 AM
We are still not getting the ringback when transferring the call back to ITSP. That's the issue.
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