topic Re: Phone call falling in Unified Communications Infrastructure

Phone call falling

rafaelrangel — Fri, 24 Jan 2020 19:02:51 GMT

I have a strange situation in my scenario.
I have a CUCM 9.1, and a CUBE 15.0 (1r) M16 running on a 2901.

The scenario would be PHONE <-> CUCM <-> CUBE <-> ITSP

When I receive the call through ITSP and answer on an internal extension, when trying to transfer to a cell phone, the call goes down after pressing the transfer button.

Follow the log.

Re: Phone call falling

Kalliopi Vazima — Sun, 26 Jan 2020 15:01:47 GMT

Hello Rafael,

have you configured a sip trunk dedicated for fowarding with g711ulaw codec?

Kalliopi

Re: Phone call falling

rafaelrangel — Mon, 27 Jan 2020 12:42:20 GMT

Hello Kalliopi,

Yes. It is actually configured with 711alaw.

Re: Phone call falling

Kalliopi Vazima — Mon, 27 Jan 2020 12:55:25 GMT

Hello Rafael,

could you please create another trunk with g711ulaw and check the call again?

Kalliopi

Re: Phone call falling

TONY SMITH — Mon, 27 Jan 2020 15:56:49 GMT

Can you confirm details of the call at issue in that log file? Original called number, calling number and the number that you tried to transfer the call to?

Re: Phone call falling

rafaelrangel — Mon, 27 Jan 2020 18:13:25 GMT

I made it. However, when I try to transfer the call, it drops.

Re: Phone call falling

rafaelrangel — Mon, 27 Jan 2020 18:22:30 GMT

Hello Tony.
Original called number: 2125777755
Calling number: 21997532280
Transfer to: 21995207883

Look my CUBE config:

voice service voip
ip address trusted list
ipv4 172.16.2.0 255.255.255.0
ipv4 192.168.200.0 255.255.255.0
ipv4 192.168.254.0 255.255.255.252
address-hiding
dtmf-interworking rtp-nte
mode border-element license capacity 10
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
header-passing
subscription maximum accept 100
subscription maximum originate 100
registrar server expires max 600 min 60
midcall-signaling passthru media-change
early-offer forced
g729 annexb-all
!
voice class codec 1
codec preference 1 g711ulaw
!
voice class h323 1
h225 timeout tcp establish 3
h225 timeout setup 3
!
!
!
voice class dualtone-detect-params 1111
freq-max-deviation 20
cadence-variation 6
!
voice class dualtone-detect-params 2
!
voice class custom-cptone voice
dualtone disconnect
frequency 450
cadence 355 1995
!
!
!
!
voice translation-rule 1
rule 1 /2123919024/ /3000/
!
voice translation-rule 2
rule 1 /.*/ /2123919024/
!
voice translation-rule 3
rule 1 /2125782060/ /4000/
!
voice translation-rule 4
rule 1 /.*/ /2125782060/
!
voice translation-rule 5
rule 1 /2131722789/ /5000/
!
voice translation-rule 6
rule 1 /.*/ /2131722789/
!
voice translation-rule 7
rule 1 /2131721169/ /6000/
!
voice translation-rule 8
rule 1 /.*/ /2131721169/
!
voice translation-rule 9
rule 1 /21997627223/ /4000/
!
voice translation-rule 10
rule 1 /.*/ /21997627223/
!
voice translation-rule 11
rule 1 /1140404081/ /5000/
!
voice translation-rule 12
rule 1 /.*/ /1140404081/
!
voice translation-rule 13
rule 1 /2140404084/ /5000/
!
voice translation-rule 14
rule 1 /.*/ /2140404084/
!
voice translation-rule 15
rule 1 /2140404042/ /6000/
!
voice translation-rule 16
rule 1 /.*/ /2140404042/
!
voice translation-rule 17
rule 1 /1123919024/ /3000/
!
voice translation-rule 18
rule 1 /.*/ /1123919024/
!
voice translation-rule 19
rule 1 /1123911712/ /7000/
!
voice translation-rule 20
rule 1 /.*/ /1123911712/
!
voice translation-rule 21
rule 1 /2123910391/ /7000/
!
voice translation-rule 22
rule 1 /.*/ /2123910391/
!
voice translation-rule 23
rule 1 /1140404042/ /6000/
!
voice translation-rule 24
rule 1 /.*/ /1140404042/
!
voice translation-rule 25
rule 1 /2125777755/ /4000/
!
voice translation-rule 26
rule 1 /.*/ /2125777755/
!
!

voice translation-profile xxx
translate called 13
!
voice translation-profile xxx
translate called 11
!
voice translation-profile xxx
translate called 15
!
voice translation-profile xxx
translate called 23
!
voice translation-profile xxx
translate called 21
!
voice translation-profile xxx
translate called 19
!
voice translation-profile xxx
translate called 1
!
voice translation-profile xxx
translate called 17
!
voice translation-profile xxx
translate called 25
!
voice translation-profile xxx
translate calling 14
!
voice translation-profile xxx
translate calling 12
!
voice translation-profile xxx
translate calling 16
!
voice translation-profile xxx
translate calling 24
!
voice translation-profile xxx
translate calling 22
!
voice translation-profile xxx
translate calling 20
!
voice translation-profile xxx
translate calling 2
!
voice translation-profile xxx
translate calling 18
!
voice translation-profile xxx
translate calling 26
!
voice translation-profile xxx
translate called 3
!
voice translation-profile xxx
translate called 5
!
voice translation-profile xxx
translate called 7
!
voice translation-profile xxx
translate called 9
!
voice translation-profile xxx
translate called 4
!
voice translation-profile xxx
translate called 6
!
voice translation-profile xxx
translate called 8
!
voice translation-profile xxx
translate called 10
!
!
!
license udi pid CISCO2901/K9 sn FJC1950A2J4
hw-module pvdm 0/0
!
!
!
username admin privilege 15 secret 5 $1$yTah$vW9AcupkqI3qKUD5c8rMJ/
!
redundancy
!
!
!
!
!
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
description TO-CATALYST-0/23
ip address 192.168.200.8 255.255.255.0
ip tcp adjust-mss 1412
duplex auto
speed auto
!
interface GigabitEthernet0/1
description PrimaryWANDesc_TO-VLAN2
ip address dhcp
duplex auto
speed auto
!
no ip classless
ip forward-protocol nd
!
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
!
ip nat inside source list nat-list interface GigabitEthernet0/1 overload
ip route 0.0.0.0 0.0.0.0 192.168.254.1
ip route 0.0.0.0 0.0.0.0 GigabitEthernet0/1
ip route 172.16.2.0 255.255.255.0 192.168.200.1
ip route 172.16.10.0 255.255.255.0 192.168.200.1
ip route 0.0.0.0 0.0.0.0 GigabitEthernet0/1 dhcp
!
dialer-list 1 protocol ip permit
!
!
access-list 3 permit xxx.xxx.xxx.xxx
access-list 23 permit 172.16.2.0 0.0.0.255
access-list 23 permit 192.168.20.0 0.0.0.255
access-list 23 permit 192.168.200.0 0.0.0.255
access-list 23 permit 172.16.10.0 0.0.0.255
!
control-plane
!
!
voice-port 0/0/0
no battery-reversal
compand-type a-law
cptone BR
timeouts interdigit 4
timeouts call-disconnect 3
timeouts wait-release 1
timing hookflash-out 50
connection plar 4000
impedance 600c
description XTECH 25782060
caller-id enable
caller-id alerting line-reversal
caller-id alerting dsp-pre-allocate
!
voice-port 0/0/1
no battery-reversal
compand-type a-law
cptone BR
timeouts interdigit 4
timeouts call-disconnect 3
timeouts wait-release 1
timing hookflash-out 50
connection plar 5000
impedance 600c
description APC LOJA
caller-id enable
caller-id alerting line-reversal
caller-id alerting dsp-pre-allocate
!
voice-port 0/0/2
no battery-reversal
compand-type a-law
cptone BR
timeouts interdigit 4
timeouts call-disconnect 3
timeouts wait-release 1
timing hookflash-out 50
connection plar 6000
impedance 600c
description COMPRAR CISCO
caller-id enable
caller-id alerting line-reversal
caller-id alerting dsp-pre-allocate
!
voice-port 0/0/3
no battery-reversal
compand-type a-law
cptone BR
timeouts interdigit 4
timeouts call-disconnect 3
timeouts wait-release 1
timing hookflash-out 50
connection plar 4000
impedance 600c
description CELULAR XTECH
caller-id enable
caller-id alerting line-reversal
caller-id alerting dsp-pre-allocate
!
!
!
!
!
mgcp
mgcp call-agent 192.168.200.2 service-type mgcp version 0.1
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
!
ccm-manager music-on-hold
!
ccm-manager mgcp
ccm-manager config server 192.168.200.2
!
dial-peer voice 100 voip
description INBOUND - FLUXO TRUNK TO CUCM
huntstop
destination-pattern 3000
session protocol sipv2
session target ipv4:192.168.200.2
dtmf-relay rtp-nte digit-drop
no vad
!
dial-peer voice 101 voip
translation-profile incoming xxx
session protocol sipv2
session target sip-server
incoming called-number 2123919024
dtmf-relay rtp-nte
!
dial-peer voice 103 voip
destination-pattern .
session protocol sipv2
session target sip-server
incoming called-number 0T
voice-class codec 1
no voice-class sip early-offer forced
dtmf-relay rtp-nte
no vad
!
dial-peer voice 104 voip
description INBOUND - FLUXO TRUNK TO CUCM ROTA xxx
huntstop
destination-pattern 4000
session protocol sipv2
session target ipv4:192.168.200.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 105 pots
description INCOMING FXO1
translation-profile incoming PSTN-IN-FXO1
incoming called-number 2125782060
direct-inward-dial
port 0/0/0
forward-digits all
!
dial-peer voice 106 voip
description INBOUND - FLUXO TRUNK TO CUCM ROTA xxx
huntstop
destination-pattern 5000
session protocol sipv2
session target ipv4:192.168.200.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 107 pots
description INCOMING FXO2
translation-profile incoming PSTN-IN-FXO2
incoming called-number 2131722789
direct-inward-dial
port 0/0/1
forward-digits all
!
dial-peer voice 108 voip
description INBOUND - FLUXO TRUNK TO CUCM ROTA xxx
huntstop
destination-pattern 6000
session protocol sipv2
session target ipv4:192.168.200.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 109 pots
description INCOMING FXO3
translation-profile incoming PSTN-IN-FXO3
incoming called-number 2131721169
direct-inward-dial
port 0/0/2
forward-digits all
!
dial-peer voice 110 voip
description INBOUND - FLUXO TRUNK TO CUCM ROTA CELULAR xxx
huntstop
destination-pattern 4000
session protocol sipv2
session target ipv4:192.168.200.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 111 pots
description INCOMING FXO4
translation-profile incoming PSTN-IN-FXO4
incoming called-number 21997627223
direct-inward-dial
port 0/0/3
forward-digits all
!
dial-peer voice 112 voip
description INBOUND xxx - FLUXO TRUNK TO CUCM
huntstop
destination-pattern 5000
session protocol sipv2
session target ipv4:192.168.200.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 113 voip
translation-profile incoming ITSP-IN-xxx
session protocol sipv2
session target sip-server
incoming called-number 2140404084
dtmf-relay rtp-nte
!
dial-peer voice 114 voip
description INBOUND xxx - FLUXO TRUNK TO CUCM
huntstop
destination-pattern 5000
session protocol sipv2
session target ipv4:192.168.200.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 115 voip
translation-profile incoming ITSP-IN-xxx
session protocol sipv2
session target sip-server
incoming called-number 1140404081
dtmf-relay rtp-nte
!
dial-peer voice 116 voip
description INBOUND xxx - FLUXO TRUNK TO CUCM
huntstop
destination-pattern 6000
session protocol sipv2
session target ipv4:192.168.200.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 117 voip
translation-profile incoming ITSP-IN-xxx
session protocol sipv2
session target sip-server
incoming called-number 2140404042
dtmf-relay rtp-nte
!
dial-peer voice 118 voip
description INBOUND xxx - FLUXO TRUNK TO CUCM
huntstop
destination-pattern 6000
session protocol sipv2
session target ipv4:192.168.200.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 119 voip
translation-profile incoming ITSP-IN-xxx
session protocol sipv2
session target sip-server
incoming called-number 1140404042
dtmf-relay rtp-nte
!
dial-peer voice 120 voip
description INBOUND xxx - FLUXO TRUNK TO CUCM
huntstop
destination-pattern 7000
session protocol sipv2
session target ipv4:192.168.200.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 121 voip
translation-profile incoming ITSP-IN-xxx
session protocol sipv2
session target sip-server
incoming called-number 1123911712
dtmf-relay rtp-nte
!
dial-peer voice 122 voip
description INBOUND xxx - FLUXO TRUNK TO CUCM
huntstop
destination-pattern 7000
session protocol sipv2
session target ipv4:192.168.200.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 123 voip
translation-profile incoming ITSP-IN-xxx
session protocol sipv2
session target sip-server
incoming called-number 2123910391
dtmf-relay rtp-nte
!
dial-peer voice 124 voip
description INBOUND xxx - FLUXO TRUNK TO CUCM
huntstop
destination-pattern 3000
session protocol sipv2
session target ipv4:192.168.200.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 125 voip
translation-profile incoming ITSP-IN-xxx
session protocol sipv2
session target sip-server
incoming called-number 1123919024
dtmf-relay rtp-nte
!
dial-peer voice 126 voip
description INBOUND xxx - FLUXO TRUNK TO CUCM
huntstop
destination-pattern 4000
session protocol sipv2
session target ipv4:192.168.200.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 127 voip
translation-profile incoming ITSP-IN-xxx
session protocol sipv2
session target sip-server
incoming called-number 2125777755
dtmf-relay rtp-nte
!
dial-peer voice 128 voip
description INBOUND xxx - FLUXO TRUNK TO CUCM
huntstop
destination-pattern 3000
session protocol sipv2
session target ipv4:192.168.200.2
dtmf-relay rtp-nte
no vad
!
dial-peer voice 131 voip
translation-profile outgoing ITSP-OUT-xxx
preference 1
destination-pattern .
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
!
sip-ua
credentials username xxxxxx password xxxx realm xxxxx

authentication username xxxx password xxxx
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar ipv4:xxxx expires 60
sip-server ipv4:xxxx
!
!
!

Re: Phone call falling

TONY SMITH — Tue, 28 Jan 2020 11:32:11 GMT

Couple of quick comments and checks. It looks as if you may be configured to use different codecs for inbound and outbound calls. Quicker than rummaging through all those dial peers (why so many?), could you do a couple of quick checks.

(1) Place an outbound call and see what codecs are in use, and which dial peers

"sh call act voice comp" and "sh call act voice | i PeerId"

(2) Place an inbound call and do the same.

You have an amazing number of dial peers and lots of them have no codec specified, which means they'll default to G729. You have a codec class with G711u applied to a couple of dial peers. In spite of that your log shows outbound call connects with G711a. It's not quite clear to me how or why that's negotiated, but I think that's the issue.

Inbound call is established as G729

You make an outbound call as G711a

Then the transfer fails as the two calls have different codec.

Re: Phone call falling

Kalliopi Vazima — Tue, 28 Jan 2020 13:33:25 GMT

Hello,

in older deployments i used to create mtp resources (ios) in cucm and assign them as sccp in CUBE for every different sip trunk codec. i also created the necessary mrg and configured sip trunk as attached. Cube configuration example follows

dspfarm profile 2 mtp
codec g711alaw
maximum sessions software 4
associate application SCCP
no shutdown
!
dspfarm profile 3 mtp
no codec g711ulaw
codec g729r8
maximum sessions software 4
associate application SCCP
no shutdown

sccp ccm group 1

associate profile 2 register mtp711_XXX
associate profile 3 register mtp729_XXX

Re: Phone call falling

TONY SMITH — Tue, 28 Jan 2020 14:41:43 GMT

Personally I think you should harden up the existing configuration in the first instance. Then we can see whether you need more stuff. Codecs first.

Inbound - PSTN to CUCM. As far as I can see this will always use G729 as there is no codec defined so the default is used. It's not clear if that is a deliberate choice.

Outbound CUCM to PSTN, in the example given you use G711a. We need to understand how that comes about. It looks as if the call is relaying off an MTP or Transcoder. What are your region settings? How is the trunk configured?

I think we want the same codec in both directions, then review and see what else needs fixing.

If I was taking over that installation I'd be looking to get rid of most of those dial peers, I am sure that matching by IP address and using wildcard destination patterns could consolidate them down. My installs mostly have just two.

Re: Phone call falling

rafaelrangel — Tue, 28 Jan 2020 17:31:49 GMT

Thank you Tony.

Sorted out.
I cleaned the configuration of the dial-peers and included the codec in the legs.

Thankful.

Re: Phone call falling

TONY SMITH — Wed, 29 Jan 2020 09:55:41 GMT

@Kalliopi Vazima wrote:
Hello,
in older deployments i used to create mtp resources (ios) in cucm and assign them as sccp in CUBE for every different sip trunk codec. i also created the necessary mrg and configured sip trunk as attached. Cube configuration example follows

Can I ask why? You could have a single transcoder profile doing all of that. A transcoder can act as an MTP but an MTP can't be a transcoder.