Hi David, You said the calls

sam saeed · ‎08-28-2015

This is the lowdown when a caller from the outside (SIP TRUNK) calls into the AA in CUC and it transfers to the Hunt pilot in CM the call connects but no audio on both sides. Just to see if it was an issue with inbound calls I changed the translation rule to bypass the AA and go straight to my extension. Call connects and audio both sides. This is a new install so it never worked before. Yesterday calls transferred from the AA weren't transferring to the pilots or phones which was pain to figure out. I realized I needed to add pstn partitions to the voicemail CSS. Thats what I get for assuming the configuration guide on setting up AA on CUC mentioned everything I needed to know to get it working.

It looks like I'm very close to get this up and running I just need calls connected through AA transfers to have audio and I'll be good to go. To add there is audio when I call the AA from within the network and the call connects through a transfer. It is also a CUCM CUC sccp integration.

This is the call flow

SIP TRUNK -> V Gateway 2911 -> CUCM -> CUC AA->hander transfers to Hunt Pilot

I don't think it would be a CSS issue since calls do connect its some sort of RTP issue but I can't wrap my head around why it only doesn't work when AA is doing the transferring versus an outside caller calling directly my extension.

Would a pattern in the restricted table in CUC prevent rtp from flowing?

Shadi Shami · ‎08-28-2015

David,

As a quick suggestion try checking mtp required check box under the sip trunk configuration on cucm, make sure the trunk has an MRGL contains MTP.

To figure out where the issue is exactly, we need (debug ccsip messages) and (show run) from the voice gateway together with Detaild CUCM traces.

Thank you,

Shadi

sam saeed · ‎08-28-2015

@Shadi,

Ok I'll try enabling MTP software on the cucm and see if that'll do it. What do I need to run to get the cucm traces? I know my way around the cucm but this is the first time I've been given a full build out I am more of CME smb guy. So I am learning a boatload of stuff through these hiccups.

@Lea

Yes I know more often then not no audio is usually a routing/nat issue depending on the scenario. But I can't make sense why it works if I change the translation pattern on the GW pointing to my extension versus the call getting processed by AA and handing it back to CUCM.

I confirmed that I get audio if I call the AA within the network on a deskphone, transfer and connects. I can ping from GW and cuc os admin to my extensions. Its not an isolated issue to one phone this issue effects all of the internal phones.

Mohammed al Baqari · ‎08-28-2015

Hi David,

You said the calls are working if you dial the phone directly bypassing the AA. From your CUBE, I need to see the output of the command 'sh call act vo brief called-number xxxx' in both cases (without AA and with AA).

sam saeed · ‎10-03-2015

@Mohammed

I'm thinking my issue may be related to a software bug. The CUCM version my manager downloaded was 9.1(1.10000.11)

Cisco Bug: CSCus91721 - No audio on call through SIP trunk to CUBE due to MTP hanging
one of the effected versions: 9.1(1.10000.11)

WITH AA
TEST2911#
TEST2911#show call act voi bri

Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
1ED4 : 691 14216390ms.1 (16:36:31.625 UTC/EST Sat Oct 3 2015) +450 pid:9 Answer 9085776815 connected
dur 00:01:11 tx:1210/193600 rx:1189/185560 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 216.86.41.7:30666 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off Transcoded: No
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a

1ED4 : 692 14216400ms.1 (16:36:31.635 UTC/EST Sat Oct 3 2015) +430 pid:7 Originate 2010 connected
dur 00:01:11 tx:1189/185560 rx:1203/192480 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 10.10.10.15:4000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off Transcoded: No
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a

Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2

TEST2911#
TEST2911#

WITHOUT AA

TEST2911#show call act voi bri
Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
1EDB : 693 14386840ms.1 (16:39:22.075 UTC/EST Sat Oct 3 2015) +5860 pid:9 Answer 9085776815 active
dur 00:00:01 tx:69/11040 rx:60/9600 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 216.86.41.5:6320 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off Transcoded: No
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a

1EDB : 694 14386850ms.1 (16:39:22.085 UTC/EST Sat Oct 3 2015) +5840 pid:7 Originate 114 active
dur 00:00:01 tx:60/9600 rx:69/11040 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 10.10.10.15:26204 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off Transcoded: No
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a

Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2

I looked through the call trace using Translator X on the PUB (using AA) and noticed this during the time that I punched the extensions during the AA playback:

Also how do I upload the traces on here. I tried but it said txt.gz extensions can't be uploaded.

This is the show run:

TEST2911#sh run
Building configuration...

Current configuration : 10229 bytes
!
! Last configuration change at 16:53:38 UTC/EST Sat Oct 3 2015 by
version 15.2
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname TEST2911
!
boot-start-marker
boot system flash1:c2900-universalk9-mz.SPA.152-4.M4.bin
boot-end-marker
!
!
logging buffered 10000000
!
no aaa new-model
clock timezone UTC -5 0
clock summer-time UTC/EST recurring
!
ip cef
!
!
!
!

!
ip dhcp excluded-address 10.10.10.1 10.10.10.10
ip dhcp excluded-address 192.168.2.1 192.168.2.10
ip dhcp excluded-address 192.168.3.1 192.168.3.10
ip dhcp excluded-address 192.168.3.101 192.168.3.255
!
ip dhcp pool voice
network 10.10.10.0 255.255.255.0
default-router 10.10.10.1
option 150 ip 10.10.10.242
!
ip dhcp pool Public_WIFI
network 192.168.3.0 255.255.255.0
default-router 192.168.3.1 255.255.255.0
dns-server 209.244.0.3
!
!
!
ip domain name test.local
ip name-server 8.8.8.8
ip name-server 8.8.4.4
ip name-server 151.198.0.38
ip name-server 151.202.0.85
ipv6 multicast rpf use-bgp
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
!
!
!
voice-card 0
dsp services dspfarm
!
!
!
voice service voip
ip address trusted list
ipv4 X.X.X.X
mode border-element
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/0.10
bind media source-interface GigabitEthernet0/0.10
registrar server expires max 1200 min 300
early-offer forced
midcall-signaling passthru
!
voice class media 1
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
voice class custom-cptone JOIN
dualtone conference
frequency 1000 2000
cadence 100 100 100
!
voice class custom-cptone LEAVE
dualtone conference
frequency 600 800
cadence 100 100 100
!
!
!
!
voice translation-rule 1
rule 1 /2016....../ /2010/
!
voice translation-rule 2
rule 1 /^911$/ /911/
rule 2 /^8$.*$/ /\1/
!
voice translation-rule 3
rule 1 /^.*/ /20168...../
!
voice translation-rule 4
rule 1 /^8(.......)$/ /201\1/
rule 2 /2010/ /20168...../
rule 4 /^8(...)$/ /2016\1/
rule 5 /^8(.*)/ /\1/
!
voice translation-rule 5
rule 1 reject /9549783900/
!
voice translation-rule 6
rule 1 /^201$.*$/ /8\1/
rule 2 /$..........$/ /81\1/
!
!
voice translation-profile BLOCKED-INCOMING
translate calling 5
!
voice translation-profile CUE_Voicemail/AutoAttendant
translate called 1
!
voice translation-profile PSTN_CallForwarding
translate redirect-target 4
translate redirect-called 4
!
voice translation-profile PSTN_Outgoing
translate calling 3
translate called 2
translate redirect-target 4
translate redirect-called 4
!
voice translation-profile filter_Incoming
translate calling 6
!
!
!
license udi pid CISCO2911/K9 sn xxxxxxxx
hw-module pvdm 0/0
!
!
!
redundancy
!
!
!
!
!
!
!
!
!
!
!
!
!
!
interface Loopback0
ip address 1.1.1.1 255.255.255.255
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
!
interface GigabitEthernet0/0.1
description $FW_INSIDE$
encapsulation dot1Q 1 native
ip address 10.1.1.201 255.255.255.0
!
interface GigabitEthernet0/0.10
description $FW_INSIDE VOICE NETWORK$
encapsulation dot1Q 10
ip address 10.10.10.1 255.255.255.0
!
interface GigabitEthernet0/0.50
description $DATA_NETWORK$
encapsulation dot1Q 50
ip address 192.168.2.1 255.255.255.0
ip helper-address 192.168.2.240
!
interface GigabitEthernet0/0.51
encapsulation dot1Q 51
ip address 192.168.3.1 255.255.255.0
ip access-group VLAN51 in
!
interface GigabitEthernet0/0.52
description MANAGEMENT
encapsulation dot1Q 52
ip address 192.168.4.1 255.255.255.0
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
!
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 192.168.2.251
!
!
control-plane
!
!
!
!
!
ccm-manager music-on-hold
!
!
mgcp profile default
!
sccp local Loopback0
sccp ccm 10.10.10.241 identifier 1 priority 1 version 7.0
!
sccp ccm group 1
associate profile 2 register R2-CONF
associate profile 1 register R1-XCODE
!
dspfarm profile 1 transcode
description SiP-AA trial
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
codec g729br8
codec g729r8
maximum sessions 6
associate application SCCP
!
dspfarm profile 2 conference
codec g729br8
codec g729r8
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
maximum sessions 2
associate application SCCP
!
dial-peer voice 1 voip
description **Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
preference 1
destination-pattern 8[2-9]..[2-9]......
session protocol sipv2
session target dns:SIP ISP
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 2 voip
description **911 Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 911
session protocol sipv2
session target dns:SIP ISP
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 3 voip
description **Emergency Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 8911
session protocol sipv2
session target dns:SIP ISP
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 4 voip
description **911/411 Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 8[2-9]11
session protocol sipv2
session target dns:SIP ISP
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 5 voip
description **Star Code to SIP Trunk**
destination-pattern *..
session protocol sipv2
session target dns:SIP ISP
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 9 voip
description **Incoming Call from SIP Trunk**
translation-profile incoming CUE_Voicemail/AutoAttendant
call-block translation-profile incoming BLOCKED-INCOMING
call-block disconnect-cause incoming call-reject
session protocol sipv2
session target dns:SIP ISP
incoming called-number .%
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 10 voip
description **International Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 8011T
session protocol sipv2
session target dns:SIP ISP
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 11 voip
description **Outgoing Call to SIP Trunk**
translation-profile outgoing PSTN_Outgoing
destination-pattern 81[2-9]..[2-9]......
session protocol sipv2
session target dns:SIP ISP
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 7 voip
description **CUCM SUB **
translation-profile outgoing PSTN_CallForwarding
preference 1
destination-pattern 2010
session protocol sipv2
session target ipv4:10.10.10.242
voice-class codec 1
dtmf-relay sip-kpml
no vad
!
dial-peer voice 8 voip
description **CUCM PUB **
translation-profile outgoing PSTN_CallForwarding
preference 1
destination-pattern 2010
session protocol sipv2
session target ipv4:10.10.10.241
dtmf-relay sip-kpml
codec g711ulaw
no vad
!
!
sip-ua
no remote-party-id
retry invite 2
retry register 10
retry options 5
timers connect 100
sip-server dns:omitted
connection-reuse
!
!
!
gatekeeper
shutdown
!
!
call-manager-fallback
max-conferences 8 gain -6
transfer-system full-consult
moh "music-on-hold.au"
multicast moh 239.1.1.254 port 16384 route 10.10.10.1 10.10.10.241
!
!
!
line con 0
exec-timeout 61 0
logging synchronous
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
exec-timeout 5041 0
logging synchronous
login local
length 0
transport input telnet ssh
!
scheduler allocate 20000 1000
ntp master
ntp update-calendar
!
end

TEST2911#

Leo Salcie Tejeda · ‎08-28-2015

Hi,

Audio issue mostly is cause by routing issue. Are you phone reachable from CUC and Gateway router?

Use ping to confirm.

Regards

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PSTN to CUC AA no audio after call transfers and answers