Cisco CUCM to Alcatel PBX SIP calling issues
Hi All
I have configured a SIP trunk between my cucms and an Alcatel old pbx on a remote site, they are all identical configs.
However one of them, the remote site Alcatel can call my cucm and voice is ok
But when we try to dial from the CUCM to the Alcatel we are getting the fast busy tone!
Codecs are set etc! as it works one way fine!
any ideas what thsi could be ?
cheers
Hi here is a snip of the trace for the call
the calling phone was ext 448 the called number over the sip trunk is 88044615
cheers
16
2015/01/23 08:15:31.897|CC|REJECT|26821723|26821724|476|8804615|8804615|1
2015/01/23 08:15:43.577|CC|RELEASE|26821726|26821727|16
2015/01/23 08:15:54.907|CC|SETUP|26821728|26821729|476|88044615|88044615
2015/01/23 08:15:54.909|CC|OFFERED|26821728|26821729|476|88044615|88044615|SEPC4641301122E|DELHI-SIP-TRUNK
2015/01/23 08:15:55.273|SIPT|26821729|TCP|OUT|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,13,53766610.2^*^*|13409968|[email protected]|INVITE
2015/01/23 08:15:55.640|SIPT|26821729|TCP|IN|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651678^172.20.65.5^*|13409969|[email protected]|100 Trying
2015/01/23 08:15:56.012|SIPT|26821729|TCP|IN|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651679^172.20.65.5^*|13409970|[email protected]|403 Forbidden
2015/01/23 08:15:56.012|SIPT|26821729|TCP|OUT|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651679^172.20.65.5^*|13409971|[email protected]|ACK
2015/01/23 08:15:58.668|CC|RELEASE|26821728|26821729|67108885
2015/01/23 08:16:02.133|CC|SETUP|26821730|26821731|4133320804|467|467
2015/01/23 08:16:02.136|CC|OFFERED|26821730|26821731|4133320804|467|467|172.24.32.38|SEPC464130114C0
2015/01/23 08:16:18.712|CC|SETUP|26821733|26821734|568|487|487
2015/01/23 08:16:18.714|CC|OFFERED|26821733|26821734|568|487|487|SEPC464130117E9|SEPC4641301147E
2015/01/23 08:16:20.151|CC|SETUP|26821730|26821737|4133320804|467|1999
2015/01/23 08:16:20.157|CC|OFFERED|26821730|26821737|4133320804|467|1999|172.24.32.38|CiscoUM1-VI54
2015/01/23 08:16:20.159|CC|RELEASE|26821731|0|0
2015/01/23 08:16:28.151|CC|RELEASE|26821730|26821737|16
2015/01/23 08:16:31.997|CC|SETUP|26821738|26821739|476|88044615|88044615
2015/01/23 08:16:31.998|CC|OFFERED|26821738|26821739|476|88044615|88044615|SEPC4641301122E|DELHI-SIP-TRUNK
2015/01/23 08:16:31.999|SIPT|26821739|TCP|OUT|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,13,51956482.95772^172.24.48.180^SEPC4641301122E|13409978|[email protected]|INVITE
2015/01/23 08:16:32.366|SIPT|26821739|TCP|IN|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651682^172.20.65.5^*|13409979|[email protected]|100 Trying
2015/01/23 08:16:32.764|SIPT|26821739|TCP|IN|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651683^172.20.65.5^*|13409980|[email protected]|403 Forbidden
2015/01/23 08:16:32.764|SIPT|26821739|TCP|OUT|172.24.32.34|5060|DELHI-SIP-TRUNK|172.20.65.5|5060|1,100,63,1.4651683^172.20.65.5^*|13409981|[email protected]|ACK
2015/01/23 08:16:39.148|CC|RELEASE|26821738|26821739|67108885
2015/01/23 08:16:44.261|CC|SETUP|26821740|26821741|4133320804|465|465
2015/01/23 08:16:44.263|CC|OFFERED|26821740|26821741|4133320804|465|465|172.24.32.38|SEPC46413011466
2015/01/23 08:17:26.622|CC|RELEASE|26821733|26821734|16
2015/01/23 08:17:33.320|CC|SETUP|26821743|26821744|568|536|536
2015/01/23 08:17:33.322|CC|OFFERED|26821743|26821744|568|536|536|SEPC464130117E9|SEPC46413011477
2015/01/23 08:17:44.673|CC|RELEASE|26821706|26821707|16
2015/01/23 08:18:38.248|CC|RELEASE|26821713|26821714|16
2015/01/23 08:18:51.306|CC|RELEASE|26821740|26821741|16
2015/01/23 08:18:53.509|CC|SETUP|26821746|26821747|447|033255961|033255961
2015/01/23 08:18:53.513|CC|OFFERED|26821746|26821747|447|033255961|033255961|SEPC46413011482|172.24.32.38
2015/01/23 08:18:53.739|SIPL|0|TCP|IN|172.24.32.34|50
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Calling issue with Cisco 7937 conference station
Hi Friends,
I am facing issue wiht Cisco 7937 conference station, our customer have various branch offices accross the world. All branches are connected over MPLS through service provider( SIP service provider) . there is a centralized CUCM and remote office have SIP Voice gateways .
When making calls from once remote site to another using Cisco 6921 phones calls working fine
When making calls from once remote site to another using Cisco 7937 conference station to make call any phone at remote office, calls are getting disconneted, remote phone rings when calls, but its gets fast busy tone when other party picks up the phone and not able to talk.
I suspect the issue with Codec but we have configured transcoders in VG and registered with CUCM
Please help me if any one experience such issue earlier.
Regards
Sivahi Basant,
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Call Flow --> Phone A ---->CUCMRouterpattern--> SIP trunk ----> Voice gateway--->Service provider cloud---> Respective Voice Gateway---> CUCM -- Phone B
Show Run
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.02.27 15:14:52 =~=~=~=~=~=~=~=~=~=~=~=
sh run
Building configuration...
Current configuration : 12139 bytes
! Last configuration change at 06:35:59 UTC Tue Feb 25 2014
! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname eucamvgw01
boot-start-marker
boot system flash:c2900-universalk9-mz.SPA.151-4.M5.bin
boot-end-marker
card type e1 0 0
logging buffered 51200 warnings
no logging console
no aaa new-model
no network-clock-participate wic 0
no ipv6 cef
ip source-route
ip traffic-export profile cuecapture mode capture
bidirectional
ip cef
ip multicast-routing
ip domain name drreddys.eu
ip name-server 10.197.20.1
ip name-server 10.197.20.2
multilink bundle-name authenticated
stcapp ccm-group 2
stcapp
stcapp feature access-code
stcapp feature speed-dial
stcapp supplementary-services
port 0/1/0
fallback-dn 5428025
port 0/1/1
fallback-dn 5428008
port 0/1/2
fallback-dn 5421462
port 0/1/3
fallback-dn 5421463
isdn switch-type primary-net5
crypto pki token default removal timeout 0
voice-card 0
dsp services dspfarm
voice call send-alert
voice call disc-pi-off
voice call convert-discpi-to-prog
voice rtp send-recv
voice service voip
ip address trusted list
ipv4 10.198.0.0 255.255.255.0
ipv4 152.63.1.0 255.255.255.0
address-hiding
allow-connections sip to sip
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
fax-relay ans-disable
sip
rel1xx supported "track"
privacy pstn
no update-callerid
early-offer forced
call-route p-called-party-id
voice class uri 100 sip
host 41.206.187.71
voice class codec 10
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 ilbc
codec preference 4 g729r8
codec preference 5 g729br8
voice class codec 20
codec preference 1 g729br8
codec preference 2 g729r8
voice moh-group 1
moh flash:moh/Panjo.alaw.wav
description MOH G711 alaw
multicast moh 239.1.1.2 port 16384 route 10.198.2.9
voice translation-rule 1
rule 1 /^012237280\(..\)/ /54280\1/
rule 2 /^012236514\(..\)/ /54214\1/
rule 3 /^01223651081/ /5428010/
rule 4 /^01223506701/ /5428010/
voice translation-rule 2
rule 1 /^00\(.+\)/ /+\1/
rule 2 /^0\(.+\)/ /+44\1/
rule 3 /^\([0-9].+\)/ /+\1/
voice translation-rule 3
rule 1 /^9\(.+\)/ /\1/
rule 2 /^\+44\(.+\)/ /0\1/
rule 3 /^\+\(.+\)/ /00\1/
voice translation-rule 4
rule 1 /^54280\(..\)/ /12237280\1/
rule 2 /^54214\(..\)/ /12236514\1/
rule 3 /^\+44\(.+\)/ /\1/
rule 4 /^.54280\(..\)/ /12237280\1/
rule 5 /^.54214\(..\)/ /12236514\1/
voice translation-rule 9
rule 1 /^\(....\)/ /542\1/
voice translation-rule 10
voice translation-rule 11
rule 1 /^\+44122372\(....\)/ /542\1/
rule 2 /^\+44122365\(....\)/ /542\1/
voice translation-rule 12
voice translation-rule 13
rule 1 /^\([18]...\)/ /542\1/
voice translation-rule 14
voice translation-profile MPLS-incoming
translate calling 10
translate called 9
voice translation-profile MPLS-outgoing
translate calling 11
translate called 12
voice translation-profile PSTN-incoming
translate calling 2
translate called 1
voice translation-profile PSTN-outgoing
translate calling 4
translate called 3
voice translation-profile SRST-incoming
translate calling 14
translate called 13
license udi pid CISCO2921/K9 sn FGL145110RE
hw-module ism 0
hw-module pvdm 0/0
username administrator privilege 15 secret 5 $1$syu5$DsxdOgfS7Wltx78o4PV.60
redundancy
controller E1 0/0/0
ip tcp path-mtu-discovery
ip scp server enable
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description internal LAN
ip address 10.198.2.9 255.255.255.0
duplex auto
speed auto
interface ISM0/0
ip unnumbered GigabitEthernet0/0
service-module ip address 10.198.2.8 255.255.255.0
!Application: CUE Running on ISM
service-module ip default-gateway 10.198.2.9
interface GigabitEthernet0/1
description to TATA NGN
ip address 115.114.225.122 255.255.255.252
duplex auto
speed auto
interface GigabitEthernet0/2
description SIP Trunks external
ip address 79.121.254.83 255.255.255.248
ip access-group SIP-InBound in
ip traffic-export apply cuecapture size 8000000
duplex auto
speed auto
interface ISM0/1
description Internal switch interface connected to Internal Service Module
no ip address
shutdown
interface Vlan1
no ip address
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 10.198.2.1
ip route 10.198.2.8 255.255.255.255 ISM0/0
ip route 41.206.187.0 255.255.255.0 115.114.225.121
ip route 77.37.25.46 255.255.255.255 79.121.254.81
ip route 83.245.6.81 255.255.255.255 79.121.254.81
ip route 83.245.6.82 255.255.255.255 79.121.254.81
ip route 95.223.1.107 255.255.255.255 79.121.254.81
ip route 192.54.47.0 255.255.255.0 79.121.254.81
ip access-list extended SIP-InBound
permit ip host 77.37.25.46 any
permit ip host 83.245.6.81 any
permit ip host 83.245.6.82 any
permit ip 192.54.47.0 0.0.0.255 any
permit icmp any any
permit ip host 95.223.1.107 any
deny ip any any log
control-plane
voice-port 0/1/0
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/1
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/2
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/3
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
no ccm-manager fax protocol cisco
ccm-manager music-on-hold bind GigabitEthernet0/0
ccm-manager config server 152.63.1.19 152.63.1.100 172.27.210.5
ccm-manager sccp local GigabitEthernet0/0
ccm-manager sccp
mgcp profile default
sccp local GigabitEthernet0/0
sccp ccm 10.198.2.9 identifier 3 priority 3 version 7.0
sccp ccm 152.63.1.19 identifier 4 version 7.0
sccp ccm 152.63.1.100 identifier 5 version 7.0
sccp ccm 172.27.210.5 identifier 6 version 7.0
sccp
sccp ccm group 2
bind interface GigabitEthernet0/0
associate ccm 4 priority 1
associate ccm 5 priority 2
associate ccm 6 priority 3
associate ccm 3 priority 4
associate profile 1002 register CFB_UK_CAM_02
associate profile 1001 register XCODE_UK_CAM_02
associate profile 1000 register MTP_UK_CAM_02
dspfarm profile 1001 transcode
codec ilbc
codec g722-64
codec g729br8
codec g729r8
codec gsmamr-nb
codec pass-through
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 18
associate application SCCP
dspfarm profile 1002 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dspfarm profile 1000 mtp
codec g711alaw
maximum sessions software 200
associate application SCCP
dial-peer cor custom
name SRSTMode
dial-peer cor list SRST
member SRSTMode
dial-peer voice 100 voip
description *** Inbound CUCM ***
translation-profile incoming PSTN-incoming
incoming called-number .
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 500 voip
description *** Inbound TATA MPLS ***
translation-profile incoming MPLS-incoming
session protocol sipv2
session target sip-server
incoming called-number ....
incoming uri from 100
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 510 voip
description *** Outbound TATA MPLS ***
translation-profile outgoing MPLS-outgoing
destination-pattern 54[013-9]....
session protocol sipv2
session target ipv4:41.206.187.71
session transport udp
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 520 voip
description *** Outbound TATA MPLS ***
translation-profile outgoing MPLS-outgoing
destination-pattern 5[0-35-9].....
session protocol sipv2
session target ipv4:41.206.187.71
session transport udp
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 200 voip
description *** Inbound M12 *** 01223651081, 01223651440 - 01223651489
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 0122365....
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 201 voip
description *** Inbound M12 *** 012237280XX
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 012237280..
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 202 voip
description *** Inbound M12 *** 01223506701
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 01223506701
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 210 voip
description *** Outbound M12 ***
translation-profile outgoing PSTN-outgoing
destination-pattern +...T
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 211 voip
description *** Outbound ISDN for SRST and emergency ***
translation-profile outgoing PSTN-outgoing
destination-pattern 9.T
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 212 voip
description *** Outbound ISDN for emergency ***
translation-profile outgoing PSTN-outgoing
destination-pattern 11[02]
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 2000 voip
description *** Outbound to CUCM Primary ***
preference 1
destination-pattern 542....
session protocol sipv2
session target ipv4:152.63.1.19
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 2001 voip
description *** Outbound to CUCM Secondary ***
preference 2
destination-pattern 542....
session protocol sipv2
session target ipv4:152.63.1.100
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 2002 voip
description *** Outbound to CUCM Teritiary ***
preference 3
destination-pattern 542....
session protocol sipv2
session target ipv4:172.27.210.5
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 999010 pots
service stcapp
port 0/1/0
dial-peer voice 999011 pots
service stcapp
port 0/1/1
dial-peer voice 999012 pots
service stcapp
port 0/1/2
dial-peer voice 999013 pots
service stcapp
port 0/1/3
sip-ua
no remote-party-id
gatekeeper
shutdown
call-manager-fallback
secondary-dialtone 9
max-conferences 4 gain -6
transfer-system full-consult
ip source-address 10.198.2.9 port 2000
max-ephones 110
max-dn 400 dual-line no-reg
translation-profile incoming SRST-incoming
moh flash:/moh/Panjo.ulaw.wav
multicast moh 239.1.1.1 port 16384 route 10.198.2.9
time-zone 22
time-format 24
date-format dd-mm-yy
line con 0
login local
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 131
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
session-timeout 60
exec-timeout 60 0
privilege level 15
login local
transport input all
line vty 5 15
session-timeout 60
exec-timeout 60 0
privilege level 15
login local
transport input all
scheduler allocate 20000 1000
ntp server 10.1.30.1
end
eucamvgw01#
Sh SCCP
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.03.03 17:57:44 =~=~=~=~=~=~=~=~=~=~=~=
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
IPv4 Address: 10.198.2.9
Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.198.2.9, Port Number: 2000
Priority: 3, Version: 7.0, Identifier: 3
Call Manager: 152.63.1.19, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 4
Trustpoint: N/A
Call Manager: 152.63.1.100, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 5
Trustpoint: N/A
Call Manager: 172.27.210.5, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 6
Trustpoint: N/A
MTP Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1000
Reported Max Streams: 400, Reported Max OOS Streams: 0
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
TLS : ENABLED
Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1001
Reported Max Streams: 36, Reported Max OOS Streams: 0
Supported Codec: ilbc, Maximum Packetization Period: 120
Supported Codec: g722r64, Maximum Packetization Period: 30
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: gsmamr-nb, Maximum Packetization Period: 60
Supported Codec: pass-thru, Maximum Packetization Period: N/A
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
Conferencing Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1002
Reported Max Streams: 16, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
TLS : ENABLED
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070080
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070081
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070082
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070083
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
eucamvgw01# -
Site to Site calling issue - Cisco 2911 Dial Peer Configuration
My customer dials from remote site to main site to their main site number, the call by-passes their auto attendant and goes directly to any random available party.
At first fingers were pointing to the their PBX, however we noticed one of their sites that wasn't managed by our company did not have the issue. We cut that site over to our service and the issue started right up. I believe it is possibly due to the way the dial peers are configured and how the calls route into the PBX. Unfortunately I do not understand much about them and curious to know if anyone has any history on a issue similiar to this or any input whatsoever?
Cisco equipment/Dialpeer config below ........
co IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.2(4)M4, RELEASE SOFTWARE (fc2) - Cisco CISCO2911/K9
dial-peer voice 100 voip
description --- VoIP Dial-Peer ---
translation-profile outgoing 7digit
huntstop
preference 1
service session
destination-pattern .T
progress_ind setup enable 3
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class codec 99
dtmf-relay rtp-nte
fax-relay ecm disable
fax rate 14400
fax nsf 000000
ip qos dscp af41 signaling
no vad
dial-peer voice 150 voip
permission none
description 900 block
huntstop
destination-pattern 1900T
session protocol sipv2
session target sip-server
voice-class codec 99
dtmf-relay rtp-nte
ip qos dscp af41 signaling
no vad
dial-peer voice 151 voip
permission none
description 900 block
huntstop
destination-pattern 900T
session protocol sipv2
session target sip-server
voice-class codec 99
dtmf-relay rtp-nte
ip qos dscp af41 signaling
no vad
dial-peer voice 101 pots
description --- INCOMING Calls from PBX ---
incoming called-number .T
direct-inward-dial
dial-peer voice 1001 pots
description --- Calls to the PBX ---
preference 3
destination-pattern .T
port 0/0/1:23
forward-digits 4
Here is some ISDN debug information
BAD CALL
Protocol Profile = Networking Extensions
0xA11C0201420201008014484152545F20484F54454C535F434C4159544F4E
Component = Invoke component
Invoke Id = 66
Operation = CallingName
Name Presentation Allowed Extended
Name = XXXXXXXXXXX
Display i = ''XXXXXXXXXXX''
Calling Party Number i = 0x2180, ''XXXXXXXXXX''
Plan:ISDN, Type:National
Called Party Number i = 0x80, ''6551''
Plan:Unknown, Type:Unknown
Aug 19 16:10:47.242 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB15
Channel ID i = 0xA98381
Exclusive, Channel 1
Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: RX <- CONNECT pd = 8 callref = 0xAB15
Channel ID i = 0xA98381
Exclusive, Channel 1
Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x2B15
GOOD CALL
Protocol Profile = Networking Extensions
0xA116020144020100800E475245454E204D4F554E5441494E
Component = Invoke component
Invoke Id = 68
Operation = CallingName
Name Presentation Allowed Extended
Name = XXXXXXXXXXXXXXXXXX
Display i = ''XXXXXXXXXXX''
Calling Party Number i = 0x2180, ''XXXXXXXXXX''
Plan:ISDN, Type:National
Called Party Number i = 0x80, 'XXXX''
Plan:Unknown, Type:Unknown
Aug 19 16:15:07.999 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB17
Channel ID i = 0xA98381
Exclusive, Channel 1I done the configration via CCA and the running conf i can see two voip dial peer. this is the site where all trunk line roured. Customer from other site2 needs to call outside by taking line from site1.
dial-peer voice 2100 voip
corlist incoming call-internal
description **CCA*INTERSITE inbound call to SITE 1
translation-profile incoming multisiteInbound
incoming called-number 82...
voice-class h323 1
dtmf-relay h245-alphanumeric
fax protocol cisco
no vad
dial-peer voice 2101 voip
corlist incoming call-internal
description **CCA*INTERSITE outbound calls to SITE2
translation-profile outgoing multisiteOutbound
destination-pattern 81...
session target ipv4:192.168.50.1
voice-class h323 1
dtmf-relay h245-alphanumeric
fax protocol cisco
no vad
no dial-peer outbound status-check pots -
Incoming calls issue in Third Party SIP Phone
Hi,
Yesterday I configured my third party sip phone which is yealink in this case on cucm and successfully registered it with cucm, despite of registration i have some calling issue in this phone. I am able to make outbound calls from this phone to any other phone however issue is related to inbound calls.I tried calling its DN from anywhere but call disconnect after sometime. Also didnt get any proper sip session trace in RTMT. Kindly suggest some step to sortout this issue.
ThanksDear Manish,
Call normally dicsonnected after 30-40 sec with termination code 102 in session trace. PFB SDI trace with 5030 is Thirdparty sip phone and 5033 is c7945. Looking forward for your suggestion.
CallingPartyNumber=5033
|DialingPartition=
|DialingPattern=5030
|FullyQualifiedCalledPartyNumber=5030
|DialingPatternRegularExpression=(5030)
|DialingWhere=
|PatternType=Enterprise
|PotentialMatches=NoPotentialMatchesExist
|DialingSdlProcessId=(0,0,0)
|PretransformDigitString=5030
|PretransformTagsList=SUBSCRIBER
|PretransformPositionalMatchList=5030
|CollectedDigits=5030
|UnconsumedDigits=
|TagsList=SUBSCRIBER
|PositionalMatchList=5030
|VoiceMailbox=
|VoiceMailCallingSearchSpace=PT-LHR-LOCAL:PT-Local:Unityvmpt:PT-F6-Local:PT-ISL-LOCAL:PT-KHI-LOCAL:PT_Operator_LHR:PT_Operator_KHI:PT_Operator_ISL
|VoiceMailPilotNumber=7103
|RouteBlockFlag=RouteThisPattern
|RouteBlockCause=0
|AlertingName=Syed Ahmer
|UnicodeDisplayName=Syed Ahmer
|DisplayNameLocale=1
|OverlapSendingFlagEnabled=0
12:17:38.028 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.16.200.21:[5062]:
[23928282,NET]
INVITE sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 10.100.200.11:5060;branch=z9hG4bK1ca0cc6e317649
From: "Syed Ahmer" ;tag=8787406~039e2a80-8561-4586-8954-d01ed2aa12c8-246211918
To:
Date: Thu, 30 Jan 2014 07:17:38 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Send-Info: conference, x-cisco-conference
Alert-Info:
Contact:
Remote-Party-ID: "Syed Ahmer" ;party=calling;screen=yes;privacy=off
Max-Forwards: 70
Content-Length: 0
|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
12:17:38.028 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xaf299320|*^*^*
12:17:38.028 |EnvProcessUdpPort::fireSignal - SEND, destination = 172.16.200.21:5062|*^*^*
12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 850, 172.16.200.21:5062)|*^*^* -
Cisco ISR G2 SIP Calls Capacity
Dear all,
We're planning for Cisco Voice Gateway configuration with SIP trunk, till now no E1s are used.
I would like to know how can we calculate the number of simulataneous calls that a cisco ISR G2 router (1921. 2921.3945,etc...) can support ?
How much sip simultaneous calls each ISR G2 model can support ?
Is it better to use SIP or we must get into E1 PRI ?
Regards,The Q and A below has the call capacity you are looking for
Table 1. Number of IP-to-IP Calls per Platform
Platform
Maximum Number of Simultaneous Calls (Flow-Through)
Cisco 3945E
2500
Cisco 3925E
2100
Cisco 3945
950
Cisco 3925
800
Cisco 2951
500
Cisco 2921
400
Cisco 2911
200
Cisco 2901
100
Cisco ASR 1004; and Cisco ASR 1006 Router Processor 2 (RP2)
5000; 16000*
Cisco ASR 1002, ASR 1004, and ASR 1006 RP1
1750
Cisco AS5350XM and AS5400XM
600
Cisco 3845
500
Cisco 3825
400
Cisco 2851
225
Cisco 2821
200
Cisco 2811
110
Cisco 2801
55
http://www.cisco.com/en/US/prod/collateral/voicesw/ps6790/gatecont/ps5640/prod_qas09186a00801da69b.html
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared" -
SIP Lync Issues - calls unable to connect
Hi,
For the last couple of weeks I have been unable to make any Lync SIP calls on my home BT Infinity package using HH5.
Calls simply will not connect. However if I swap the router out the calls connect correctly.
This was working before so can only presume that BT have 'changed something'.
Any ideas??Hi Nokkyear. I have been having the same exact issue using my E51 since I got the phone two weeks ago. After plenty of stress and anger, I somehow got it to work just a few minutes ago.
In my case, I made the following change and was finally able to connect and make a test call:
1> Under Connections/SIP Settings, I changed the PUBLIC USERNAME to the full address (i.e. [email protected]). Previously I had the xxxx portion of the address.
I had also removed the appearances of the "sip:" prefix from the proxy address, the registrar address and the public username address. I would say that only after I did that, I was able to connect to my SIP server....HOWVEVER, upon review now as I write this post, I can see that the "sip:" prefix has returned to the entries.
Try this and see how it works.
LS. -
SIP to SIP Call Failures on CME to CME - sip-ua conflict/issue?
Hi,
I have two existing CME systems which I wish to allow internal calls between. These calls will go over an IPSec VPN. However the calls are failing.
Phones DN22xx - London CME 2801 - PIX505 --- Internet ---ASA5505 - India CME 2801 - Phones DN400x
I have configured dial peers on both CME's and the IPSec VPN. I can ping between both systems. The VPN allows traffic between the interface IP's of the CME systems only.
London CME (local SCCP phones 22xx):
interface FastEthernet0/0.100
encapsulation dot1Q 100 native
ip address 10.0.10.250 255.255.255.0
voice class codec 101
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 25 voip
description *** SIP Peer to India ***
answer-address 400.
destination-pattern 400.
voice-class codec 101
session protocol sipv2
session target ipv4:192.168.15.10
incoming called-number 400.
no vad
India CME (Local SSCP phones 400x):
interface FastEthernet0/0
ip address 192.168.15.10 255.255.255.0
voice class codec 100
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 10 voip
description *** SIP Peer to London UK ***
answer-address 22..
destination-pattern 22..
voice-class codec 100
session protocol sipv2
session target ipv4:10.0.10.250
incoming called-number 22..
no vad
The CME system at India also has an existing SIP dial peer to a service provider and has sip-ua configured (username, password, realm and registrar).
A call from India (4005) to London (DN2207) fails, the ccsip debug attached. I'm assuming its because the sip-ua configuration is being used for these calls to when I don't want it to be. The from field shows âFrom: <sip:[email protected]â when I need this to be the internal IP 192.168.15.10.
Can anyone offer any assistance with this?
Regards,
ChrisHi,
thanks for your input however thats not the problem. 201.196.128.56 isn't an address on the router, it only has one IP and its 192.168.15.10.
The 201.196.128.56 address is the NAT'd address on the firewall. So that when a SIP call is made to the internet with sip-ua the from address is the public IP.
Chris -
SIP Calls Drop. Receive Bye From Cube 15min,30min, 45min
Hello,
Running into an odd issue. I've seen several others having this problem with calls dropping after 15min duration. But this is a bit different. Sometimes long duration calls drop at 15min. Some at 30min, others at 45min. And sometimes not at all. Call flow is such.
8831-sip--CUCM--sip--Cube--ITSP
I'm convinced this is likely a problem with the refresh timer. But I can't explain why it wouldn't just fail only at 15min. It's also interesting to note I've only seen this on the 8831. I tried getting the issue with debugs from the cube but of course it didn't happen once I turned on ccsip message.
From the callmanager traces I see the bye arrive from cube with Reason Q.850 cause=102.
The CUCM version is 9.1.2 and cube is 15.2(4)M1. I did see some odd defect in 15.1 related to this where the refresh on the cube would send out 3 invites to the ITSP on an update. I guess it would have only 33% chance of getting it right. Any help someone could provide I'd appreciate it.Thanks for the replies.
So was able to capture it while had debugs running. This time it disconnected after an hour. Same cause=102.
Now here is where it gets interesting in the debugs. I see an invite is sent 3 seconds from callmanager. I assume this is a refresher with the same call-id. Cube receives it and sends out to ITSP. With a new call-id. We then receive a bye from ITSP cause=86. Which then of course is sent to callmanager. Here are the relevent sections of debugs.
Received from cucm to cube:
820421: May 6 09:00:42.976: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:43 2365082: Received:
2014-05-06 09:00:43 2365083: INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
2014-05-06 09:00:43 2365084: Via: SIP/2.0/TCP 10.38.246.136:5060;branch=z9hG4bK28bab16dbd5664
2014-05-06 09:00:43 2365085: From: "Marcos Vazquez" <sip:[email protected]>;tag=3831180~dfbf10b3-6c69-4443-852f-cbf609935a6f-35009402
2014-05-06 09:00:43 2365086: To: <sip:[email protected]>;tag=5EBA2282-19C8
2014-05-06 09:00:43 2365087: Date: Tue, 06 May 2014 15:00:42 GMT
2014-05-06 09:00:43 2365088: Call-ID: [email protected]
2014-05-06 09:00:43 2365089: Supported: 100rel,timer,resource-priority,replaces
2014-05-06 09:00:43 2365090: Min-SE: 1800
2014-05-06 09:00:43 2365091: User-Agent: Cisco-CUCM9.1
2014-05-06 09:00:43 2365092: Allow: INVITE, OPTIONS, I
2014-05-06 09:00:43 2365093: NFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
2014-05-06 09:00:43 2365094: CSeq: 106 INVITE
2014-05-06 09:00:43 2365095: Max-Forwards: 70
2014-05-06 09:00:43 2365096: Expires: 300
2014-05-06 09:00:43 2365097: Allow-Events: presence, kpml
2014-05-06 09:00:43 2365098: Supported: X-cisco-srtp-fallback
2014-05-06 09:00:43 2365099: Supported: Geolocation
2014-05-06 09:00:43 2365100: P-Asserted-Identity: "Marcos Vazquez" <sip:[email protected]>
2014-05-06 09:00:43 2365101: Remote-Party-ID: "Marcos Vazquez" <sip:[email protected]>;party=calling;screen=yes;privacy=off
2014-05-06 09:00:43 2365102: Contact: <sip:[email protected]:5060;transport=tcp>
2014-05-06 09:00:43 2365103: Content-Type: application/sdp
2014-05-06 09:00:43 2365104: Content-Length: 371
2014-05-06 09:00:43 2365105:
2014-05-06 09:00:43 2365106: v=0
2014-05-06 09:00:43 2365107: o=CiscoSystemsCCM-
2014-05-06 09:00:43 2365108: SIP 3831180 1 IN IP4 10.38.246.136
2014-05-06 09:00:43 2365109: s=SIP Call
2014-05-06 09:00:43 2365110: c=IN IP4 10.96.5.28
2014-05-06 09:00:43 2365111: b=TIAS:64000
2014-05-06 09:00:43 2365112: b=AS:64
2014-05-06 09:00:43 2365113: t=0 0
2014-05-06 09:00:43 2365114: m=audio 31146 RTP/AVP 18 0 116 101
2014-05-06 09:00:43 2365115: a=rtpmap:0 PCMU/8000
2014-05-06 09:00:43 2365116: a=ptime:20
2014-05-06 09:00:43 2365117: a=rtpmap:116 iLBC/8000
2014-05-06 09:00:43 2365118: a=ptime:20
2014-05-06 09:00:43 2365119: a=maxptime:60
2014-05-06 09:00:43 2365120: a=fmtp:116 mode=20
2014-05-06 09:00:43 2365121: a=rtpmap:18 G729/8000
2014-05-06 09:00:43 2365122: a=ptime:20
2014-05-06 09:00:43 2365123: a=fmtp:18 annexb=no
2014-05-06 09:00:43 2365124: a=rtpmap:101 telephone-event/8000
2014-05-06 09:00:43 2365125: a=fmtp:101 0-15
2014-05-06 09:00:43 2365126: 5820422: May 6 09:00:42.978: //3024943/CC044C80000B/SIP/Msg/ccsipDisplayMsg:
Sent to ITSP:
Sent: Which looks like 3 are sent.
2014-05-06 09:00:43 2365128: INVITE sip:12.194.190.26:5060;transport=udp SIP/2.0
2014-05-06 09:00:43 2365129: Via: SIP/2.0/UDP 12.17.223.243:5060;branch=z9hG4bK2D699C1126
2014-05-06 09:00:43 2365130: P-Asserted-Identity: "Marcos Vazquez" <sip:[email protected]>
2014-05-06 09:00:43 2365131: From: "Marcos Vazquez" <sip:[email protected]>;tag=5EBA1628-22FC
2014-05-06 09:00:43 2365132: To: <sip:[email protected]>;tag=8088820710430052_c2b05.1.1.1385369448756.0_9843675_19511361
2014-05-06 09:00:43 2365133: Date: Tue, 06 May 2014 15:00:42 GMT
2014-05-06 09:00:43 2365134: Call-ID: [email protected]
2014-05-06 09:00:43 2365135: Supported: 100rel,timer,resource-priority,replaces,sdp-an
2014-05-06 09:00:43 2365136: at
2014-05-06 09:00:43 2365137: Min-SE: 1800
2014-05-06 09:00:43 2365138: Cisco-Guid: 3422833792-0000065536-0000746374-2297832970
2014-05-06 09:00:43 2365139: User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:43 2365140: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
2014-05-06 09:00:43 2365141: CSeq: 105 INVITE
2014-05-06 09:00:43 2365142: Max-Forwards: 70
2014-05-06 09:00:43 2365143: Timestamp: 1399388442
2014-05-06 09:00:43 2365144: Contact: <sip:[email protected]:5060>
2014-05-06 09:00:43 2365145: Expires: 60
2014-05-06 09:00:43 2365146: Allow-Events: telephone-event
2014-05-06 09:00:43 2365147: Content-Type: application/sdp
2014-05-06 09:00:43 2365148: Content-Length: 334
2014-05-06 09:00:43 2365149:
2014-05-06 09:00:43 2365150: v=0
2014-05-06 09:00:43 2365151: o=CiscoSystemsSIP-GW-UserAgent 8182 4488 IN IP4 12.17.223.243
2014-05-06 09:00:43 2365152: s=SIP Call
2014-05-06 09:00:43 2365153: c=IN IP4 1
2014-05-06 09:00:43 2365154: 2.17.223.243
2014-05-06 09:00:43 2365155: t=0 0
2014-05-06 09:00:43 2365156: m=audio 18760 RTP/AVP 18 0 100 101
2014-05-06 09:00:43 2365157: c=IN IP4 12.17.223.243
2014-05-06 09:00:43 2365158: a=rtpmap:18 G729/8000
2014-05-06 09:00:43 2365159: a=fmtp:18 annexb=no
2014-05-06 09:00:43 2365160: a=rtpmap:0 PCMU/8000
2014-05-06 09:00:43 2365161: a=rtpmap:100 X-NSE/8000
2014-05-06 09:00:43 2365162: a=fmtp:100 192-194
2014-05-06 09:00:43 2365163: a=rtpmap:101 telephone-event/8000
2014-05-06 09:00:43 2365164: a=fmtp:101 0-15
2014-05-06 09:00:43 2365165: 5820423: May 6 09:00:42.978: //3024942/CC044C80000B/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:43 2365166: Sent:
2014-05-06 09:00:43 2365167: SIP/2.0 100 Trying
2014-05-06 09:00:43 2365168: Via: SIP/2.0/TCP 10.38.246.136:5060;branch=z9hG4bK28bab16dbd5664
2014-05-06 09:00:43 2365169: From: "Marcos Vazquez" <sip:[email protected]>;tag=3831180~dfbf10b3-6c69-4443-852f-cbf609935a6f-35009402
2014-05-06 09:00:43 2365170: To: <sip:[email protected]>;tag=5EBA2282-19C8
2014-05-06 09:00:43 2365171: Date: Tue, 06 May 2014 15:00:42 GMT
2014-05-06 09:00:43 2365172: Call-ID: [email protected]
2014-05-06 09:00:43 2365173: CSeq: 106 INVITE
2014-05-06 09:00:43 2365174: Allow-Events: telephone-event
2014-05-06 09:00:43 2365175: Server: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:43 2365176: Content-Length: 0
2014-05-06 09:00:43 2365177:
2014-05-06 09:00:43 2365178: 5820424: May 6 09:00:43.479: //3024943/CC044C80000B/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:43 2365179: Sent:
2014-05-06 09:00:43 2365180: INVITE sip:12.194.190.26:5060;transport=udp SIP/2.0
2014-05-06 09:00:43 2365181: Via: SIP/2.0/UDP 12.17.223.243:5060;branch=z9hG4bK2D699C1126
2014-05-06 09:00:43 2365182: P-Asserted-Identity: "Marcos Vazquez" <sip:[email protected]>
2014-05-06 09:00:43 2365183: From: "Marcos Vazquez" <sip:[email protected]>;tag=5EBA1628-22FC
2014-05-06 09:00:43 2365184: To: <sip:[email protected]>;tag=8088820710430052_c2b05.1.1.1385369448756.0_9843675_19511361
2014-05-06 09:00:43 2365185: Date: Tue, 06 May 2014 15:00:43 GMT
2014-05-06 09:00:43 2365186: Call-ID: [email protected]
2014-05-06 09:00:43 2365187: Supported: 100rel,timer,resource-priority,replaces,sdp-an
2014-05-06 09:00:43 2365188: at
2014-05-06 09:00:43 2365189: Min-SE: 1800
2014-05-06 09:00:43 2365190: Cisco-Guid: 3422833792-0000065536-0000746374-2297832970
2014-05-06 09:00:43 2365191: User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:43 2365192: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
2014-05-06 09:00:43 2365193: CSeq: 105 INVITE
2014-05-06 09:00:43 2365194: Max-Forwards: 70
2014-05-06 09:00:43 2365195: Timestamp: 1399388443
2014-05-06 09:00:43 2365196: Contact: <sip:[email protected]:5060>
2014-05-06 09:00:43 2365197: Expires: 60
2014-05-06 09:00:43 2365198: Allow-Events: telephone-event
2014-05-06 09:00:43 2365199: Content-Type: application/sdp
2014-05-06 09:00:43 2365200: Content-Length: 334
2014-05-06 09:00:43 2365201:
2014-05-06 09:00:43 2365202: v=0
2014-05-06 09:00:43 2365203: o=CiscoSystemsSIP-GW-UserAgent 8182 4488 IN IP4 12.17.223.243
2014-05-06 09:00:43 2365204: s=SIP Call
2014-05-06 09:00:43 2365205: c=IN IP4 1
2014-05-06 09:00:44 2365206: 2.17.223.243
2014-05-06 09:00:44 2365207: t=0 0
2014-05-06 09:00:44 2365208: m=audio 18760 RTP/AVP 18 0 100 101
2014-05-06 09:00:44 2365209: c=IN IP4 12.17.223.243
2014-05-06 09:00:44 2365210: a=rtpmap:18 G729/8000
2014-05-06 09:00:44 2365211: a=fmtp:18 annexb=no
2014-05-06 09:00:44 2365212: a=rtpmap:0 PCMU/8000
2014-05-06 09:00:44 2365213: a=rtpmap:100 X-NSE/8000
2014-05-06 09:00:44 2365214: a=fmtp:100 192-194
2014-05-06 09:00:44 2365215: a=rtpmap:101 telephone-event/8000
2014-05-06 09:00:44 2365216: a=fmtp:101 0-15
2014-05-06 09:00:44 2365217: 5820425: May 6 09:00:44.479: //3024943/CC044C80000B/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:44 2365218: Sent:
2014-05-06 09:00:44 2365219: INVITE sip:12.194.190.26:5060;transport=udp SIP/2.0
2014-05-06 09:00:44 2365220: Via: SIP/2.0/UDP 12.17.223.243:5060;branch=z9hG4bK2D699C1126
2014-05-06 09:00:44 2365221: P-Asserted-Identity: "Marcos Vazquez" <sip:[email protected]>
2014-05-06 09:00:44 2365222: From: "Marcos Vazquez" <sip:[email protected]>;tag=5EBA1628-22FC
2014-05-06 09:00:44 2365223: To: <sip:[email protected]>;tag=8088820710430052_c2b05.1.1.1385369448756.0_9843675_19511361
2014-05-06 09:00:44 2365224: Date: Tue, 06 May 2014 15:00:44 GMT
2014-05-06 09:00:44 2365225: Call-ID: [email protected]
2014-05-06 09:00:44 2365226: Supported: 100rel,timer,resource-priority,replaces,sdp-an
2014-05-06 09:00:44 2365227: at
2014-05-06 09:00:44 2365228: Min-SE: 1800
2014-05-06 09:00:44 2365229: Cisco-Guid: 3422833792-0000065536-0000746374-2297832970
2014-05-06 09:00:44 2365230: User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:44 2365231: Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
2014-05-06 09:00:44 2365232: CSeq: 105 INVITE
2014-05-06 09:00:44 2365233: Max-Forwards: 70
2014-05-06 09:00:44 2365234: Timestamp: 1399388444
2014-05-06 09:00:44 2365235: Contact: <sip:[email protected]:5060>
2014-05-06 09:00:44 2365236: Expires: 60
2014-05-06 09:00:44 2365237: Allow-Events: telephone-event
2014-05-06 09:00:44 2365238: Content-Type: application/sdp
2014-05-06 09:00:44 2365239: Content-Length: 334
2014-05-06 09:00:44 2365240:
2014-05-06 09:00:44 2365241: v=0
2014-05-06 09:00:44 2365242: o=CiscoSystemsSIP-GW-UserAgent 8182 4488 IN IP4 12.17.223.243
2014-05-06 09:00:44 2365243: s=SIP Call
2014-05-06 09:00:44 2365244: c=IN IP4 1
2014-05-06 09:00:44 2365245: 2.17.223.243
2014-05-06 09:00:44 2365246: t=0 0
2014-05-06 09:00:44 2365247: m=audio 18760 RTP/AVP 18 0 100 101
2014-05-06 09:00:44 2365248: c=IN IP4 12.17.223.243
2014-05-06 09:00:44 2365249: a=rtpmap:18 G729/8000
2014-05-06 09:00:44 2365250: a=fmtp:18 annexb=no
2014-05-06 09:00:44 2365251: a=rtpmap:0 PCMU/8000
2014-05-06 09:00:44 2365252: a=rtpmap:100 X-NSE/8000
2014-05-06 09:00:44 2365253: a=fmtp:100 192-194
2014-05-06 09:00:44 2365254: a=rtpmap:101 telephone-event/8000
2014-05-06 09:00:44 2365255: a=fmtp:101 0-15
2014-05-06 09:00:45 2365256: 5820426: May 6 09:00:45.147: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:45 2365257: Received:
And then I don't see a response then send out a bye:
Sent:
2014-05-06 09:00:46 2365897: BYE sip:12.194.190.26:5060;transport=udp SIP/2.0
2014-05-06 09:00:46 2365898: Via: SIP/2.0/UDP 12.17.223.243:5060;branch=z9hG4bK2D69A54BC
2014-05-06 09:00:46 2365899: From: "Marcos Vazquez" <sip:[email protected]>;tag=5EBA1628-22FC
2014-05-06 09:00:46 2365900: To: <sip:[email protected]>;tag=8088820710430052_c2b05.1.1.1385369448756.0_9843675_19511361
2014-05-06 09:00:46 2365901: Date: Tue, 06 May 2014 15:00:44 GMT
2014-05-06 09:00:46 2365902: Call-ID: [email protected]
2014-05-06 09:00:46 2365903: User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:46 2365904: Max-Forwards: 70
2014-05-06 09:00:46 2365905: P-Asserted-Identity: "Marcos Vazquez" <sip:[email protected]>
2014-05-06 09:00:46 2365906: Timestamp: 1399388446
2014-05-06 09:00:46 2365907: CSeq: 106 BYE
2014-05-06 09:00:46 2365908: Reason: Q.850;cause=86
2014-05-06 09:00:46 2365909: P-RTP-Stat: PS=180295,OS=3604444,PR=180354,OR=3607080,PL=0,JI=0,LA=0,DU=3603
2014-05-06 09:00:46 2365910: Content-Length: 0
2014-05-06 09:00:46 2365911:
2014-05-06 09:00:46 2365912: 5820458: May 6 09:00:46.479: //3024942/CC044C80000B/SIP/Msg/ccsipDisplayMsg:
2014-05-06 09:00:46 2365913: Sent:
2014-05-06 09:00:46 2365914: BYE sip:[email protected]:5060;transport=tcp SIP/2.0
2014-05-06 09:00:46 2365915: Via: SIP/2.0/TCP 10.38.246.166:5060;branch=z9hG4bK2D69A6E75
2014-05-06 09:00:46 2365916: From: <sip:[email protected]>;tag=5EBA2282-19C8
2014-05-06 09:00:46 2365917: To: "Marcos Vazquez" <sip:[email protected]>;tag=3831180~dfbf10b3-6c69-4443-852f-cbf609935a6f-35009402
2014-05-06 09:00:46 2365918: Date: Tue, 06 May 2014 15:00:42 GMT
2014-05-06 09:00:46 2365919: Call-ID: [email protected]
2014-05-06 09:00:46 2365920: User-Agent: Cisco-SIPGateway/IOS-15.2.4.M1
2014-05-06 09:00:46 2365921: Max-Forwards: 70
2014-05-06 09:00:46 2365922: Timestamp: 1399388446
2014-05-06 09:00:46 2365923: CSeq: 101 BYE
2014-05-06 09:00:46 2365924: Reason: Q.850;cause=102
2014-05-06 09:00:46 2365925: P-R
2014-05-06 09:00:46 2365926: TP-Stat: PS=180239,OS=3604780,PR=180295,OR=3604444,PL=0,JI=0,LA=0,DU=3603
2014-05-06 09:00:46 2365927: Content-Length: 0
2014-05-06 09:00:46 2365928: -
I've been racking my brain on this since 4am, finally went to bed, and I'm at it again.My UC is registered to voip.ms, and can place multiple outbound calls with no issue.Inbound calls were coming in intermittantly last night. Right now it just goes to an announcement: "We're sorry, but due to technical difficulties, we are unable to route your call, please try your call again"I did a debug ccsip calls and I see the calls hitting the UC, and I'm wondering if it's a NAT issue:003446: Jun 28 16:53:39.120: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_validate_own_ip_addr: ReqLine IP addr does not match with host IP addr003447: Jun 28 16:53:39.120: //-1/0F1EA628A9A3/SIP/Error/sact_idle_new_message_invite: Invalid URL in incoming INVITE003448: Jun 28 16:53:39.156: //-1/0F1EA628A9A3/SIP/Call/sipSPICallInfo: The Call Setup Information is:Call Control...
This topic first appeared in the Spiceworks CommunityYou would have a wan address your provider wan and your internal network
Data and VoIP could be two separate networks..
Example 192.168.1,0 for data
192.168.2.0 voice
The outbound is failing because you don't have any dial- peers pointing to your provider
For example
Dial-peer voice 1 VoIP
Destination-partners 9T
Session target ipv4: op address of sip provider
Is your sip provider just dial tone or Internet as well.
If there providing Internet you need to configure nat...
Hope is gives you a right direction
Sent from Cisco Technical Support iPad App -
Hi there,
I'm having problems modifying the 'Dialed Number (DN)' text box under 'Advanced Configuration->Patterns for RNA timeout on outbound SIP calls' of the SIP tab in the Cisco Unified Customer Voice Portal 8.5(1) opsconsole. In a nut shell, I need to change the RNA timeout but some reason when typing into the Dialed Number text box, the input is not taken. The reason I want to change this settings is because my ICM Rona is not working with CVP:
https://supportforums.cisco.com/thread/2031366
Thanks in advance for any help.
Carlos A Trivino
[email protected]Hello Dale,
CVP doesn't allow you to exceed the RNA more than 60 Seconds. If you want to configure the timer for DN Patterns you should do it via OPS console, It would update the sip.properties files in correct way, the above way is incorrect.
Regards,
Senthil -
Incoming sip calls are not working but outgoing is working with cme
I have CME setup with voip.ms on my 2800 router, my outgoing calls are working but my incoming calls are not. Below is my config, please let me know if it is something with my config:
voice translation-rule 3
rule 1 /^9142281\(...\)$/ /\1/
voice translation-profile INCOMING_CALL_1
translate called 3
dial-peer voice 1 voip
translation-profile incoming INCOMING_CALL_1
session protocol sipv2
session target sip-server
incoming called-number .%
voice-class codec 1
dtmf-relay rtp-nte
no vadI made the change, but I am getting no output from debug voip ccapi inout. What does concern me from debug ccsip messages is:
Aug 31 12:42:04.195: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 400 Bad Request - 'Invalid Host'
Via: SIP/2.0/UDP 107.6.67.238:5060;branch=z9hG4bK000d3c36;rport
From: "+19144410197" <sip:[email protected]>;tag=as7439b9c1
To: <sip:[email protected]:1061>;tag=829C8-2532
Date: Sun, 31 Aug 2014 12:42:04 GMT
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow-Events: telephone-event
Reason: Q.850;cause=100
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
I also am getting this:
voicertr2#debug ccsip error
SIP Call error tracing is enabled
voicertr2#
Aug 31 12:45:07.359: //-1/xxxxxxxxxxxx/SIP/Error/sipSPI_validate_own_ip_addr: ReqLine IP addr does not match with host IP addr
Aug 31 12:45:07.359: //-1/78AE76E98009/SIP/Error/sact_idle_new_message_invite: Invalid URL in incoming INVITE -
Cisco CUCM 6.1.5 CodeYellowExit alarm(s) received
Hi,
I am running Cisco CUCM ver.6.1.5 & two CUCMs acting as Publisher & Subscriber, Total Phones registered on CUCM is 94.
The Problem that I am facing i used to receive the below Alarm & I noticed the CPU is hitting 85% on both CUCMs so I found the Process called CCM is utilizing 80% of CPU usage.
From Tue Feb 23 23:38:05 AST 2010 to Tue Feb 23 23:59:05 AST 2010 on node 172.30.145.4, there are 1 CodeYellowEntry alarm(s) and 1 CodeYellowExit alarm(s) received. On Tue Feb 23 23:58:35 AST 2010, the last CodeYellowEntry alarm generated: CodeYellowEntry AverageDelay : 31 EntryLatency : 20 ExitLatency : 8 SampleSize : 10 TotalCodeYellowEntry : 26 HighPriorityQueueDepth : 0 NodeID : CUCM-A
Note:
I had disabled all Traces for all CUCM types but still the problem exist.
I had enough MEM & CPU.
Please Advice !!!
Best Regards,
MohanadDid you recreate your whole cluster, just as it is in production, on UCS???
I did the procedure from 6.1(5) and 7.1(5) and had no problems.
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk -
Question regarding the use of Built-in Bridge SIP call setup to recording device
We have an application that uses Built-in Bridge (BiB) to setup a SIP call to our recording application. Recently the topology of our network was changed and for some reason the RTP for the call is not being sent. What appears to be happenning is that the SIP messages are sent from the UCCM and once the call is established we get an end call event.
We are not sure how to troubleshoot our network to determine the root cause. Any pointers will be appreciated.
Thanks.
-ArunHi
Did you ever figure this out? I am having an issue with recording calls to the PSTN from 9971 SIP phones. Station to station calls seem fine, and 7945 phones running SCCP seem fine to the PSTN.
Thanks,
Aaron -
Can someone explain how h323 to SIP calls work & vice versa.
The following messages are mapped:
SIP <---> H323
INVITE - SETUP
100 Trying - Call Proc
180 Ringing - Alerting
183 Session Progress - Progress
200 OK (for INVITE) - Connect
BYE - Release Complete
With H323 to SIP CUBE, if fast start occurs on one leg, early offer needs to happen on the other (and vice versa). Most SIP devices these days to early offer (SDP in invite) so you typically need fast start enabled on both directions of the H323 leg for this design.
Check out this link for more information:
http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-h323sip_ps5640_TSD_Products_Configuration_Guide_Chapter.html -
Outbound SIP calls to invalid numbers
what is the expected behavior for a sip call to an invalid number? and what would cause it to behave differently?
Hi Ronald,
Whats the call flow ex IP Phone --SCCP-> CUCM --SIP --> CUBE -SIP-> Telco
If called number is invalid then call should not ring. Why is far end responding back with ringing.
Can you grab below debugs from the VG
++ debug voice ccapi inout
++ debug ccsip messages
HTH,
Regards,
Mohammed Noor
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