ITSP SIP Trunk audio issue
Hi Guys,
call flow:
external caller > service provider SIP Trunk >CUBE VG>CUCM>User ip phone.
no firewall between
we are not facing this audio issue for all the calls but also for few calls , i can say 3 out of 10 calls.
under VG bind media and control command recently added by TAC guys instruction but no use.
recently we changed our office but no changes for device or configuration
also attached debug log for the issue call.
ONE THING I NOTICE 2 HOUR TIME DIFFERENCE IN VOICE GATEWAY than actual time.
Voice gateway show run: ---------
aaa session-id common
memory-size iomem 10
clock timezone CET 1 0
clock summer-time CEST recurring last Sun Mar 2:00 last Sun Oct 2:00
network-clock-participate wic 0
dot11 syslog
ip source-route
ip traffic-export profile tac mode capture
ip traffic-export profile sniffer mode capture
bidirectional
ip traffic-export profile Test mode capture
bidirectional
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 172.18.122.1 172.18.122.50
ip dhcp pool PHONES
network 172.18.122.0 255.255.255.0
domain-name ldhenergy.net
option 150 ip 172.18.122.10
default-router 172.18.122.8
no ip domain lookup
ip domain name ldhenergy.com
ip host ld-lsn-cm-01 172.18.122.10
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-net5
voice call send-alert
voice call convert-discpi-to-prog
voice call carrier capacity active
voice rtp send-recv
voice service voip
ip address trusted list
ipv4 172.18.122.11 255.255.255.255
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
h323
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
voice translation-rule 20
rule 1 /044578\(....\)$/ /\1/ type any unknown plan any unknown
voice translation-rule 30
rule 1 /021343\(....\)$/ /\1/ type any unknown plan any unknown
voice translation-rule 40
rule 1 /^\(.*\)/ /0\1/
voice translation-profile SIPIN
translate called 30
voice-card 0
dspfarm
dsp services dspfarm
crypto pki token default removal timeout 0
controller E1 0/0/0
interface FastEthernet0/0
ip address 172.18.122.3 255.255.255.0
ip helper-address 193.73.102.255
duplex auto
speed auto
interface FastEthernet0/1
ip address 10.128.18.9 255.255.255.0
duplex auto
speed auto
interface Integrated-Service-Engine1/0
ip unnumbered FastEthernet0/0
service-module ip address 172.18.122.11 255.255.255.0
!Application: CUE Running on NME
service-module ip default-gateway 172.18.122.8
no keepalive
router ospf 1005
network 172.18.122.0 0.0.0.255 area 0.0.0.1
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 172.18.122.8
ip route 10.20.0.0 255.255.0.0 172.18.122.8
ip route 172.18.122.11 255.255.255.255 Integrated-Service-Engine1/0
ip tacacs source-interface FastEthernet0/0
control-plane
ccm-manager fallback-mgcp
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 172.18.122.10
ccm-manager config
mgcp call-agent 172.18.122.10 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
mgcp profile default
sccp local FastEthernet0/0
sccp ccm 172.18.122.10 identifier 1 version 5.0.1
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 10 register HW-MTP
associate profile 20 register TRANSCODE
dspfarm profile 20 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 4
associate application SCCP
dspfarm profile 10 mtp
codec g711alaw
maximum sessions hardware 24
associate application SCCP
dial-peer voice 343 voip
translation-profile incoming SIPIN
session protocol sipv2
incoming called-number .
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte sip-notify sip-kpml cisco-rtp h245-signal h245-alphanumeric
codec g711alaw
no vad
dial-peer voice 344 voip
destination-pattern 0T
session protocol sipv2
session target ipv4:62.2.46.4
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay rtp-nte sip-notify sip-kpml cisco-rtp h245-signal h245-alphanumeric
codec g711alaw
no vad
dial-peer voice 1600 voip
destination-pattern 16..
session protocol sipv2
session target ipv4:172.18.122.10
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
codec g711alaw
no vad
dial-peer voice 1616 voip
destination-pattern 1616
session protocol sipv2
session target ipv4:172.18.122.10
dtmf-relay rtp-nte sip-notify sip-kpml cisco-rtp h245-signal h245-alphanumeric
codec g711alaw
no vad
dial-peer voice 1699 voip
destination-pattern 1699
session protocol sipv2
session target ipv4:172.18.122.10
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
codec g711alaw
no vad
sip-ua
call-manager-fallback
max-conferences 4 gain -6
transfer-system full-consult
ip source-address 172.18.122.3 port 2000
max-ephones 42
max-dn 144
Regards
Vigeesh
I suggest do a network capture or enable debug ccsip mesages.
look for conneciion ip address inside sdp field and check that are recheacble.
regards
Similar Messages
-
Hi,
What dial-peer fax statements would be advised in a case where the ITSP (SIP Trunking) only supports g711a pass though for fax?
We are having a hard time making outbound faxes work reliably from a ATA187.
How much 'relay' can/should the gateway do in this case (to my understanding there is no 'relay' in this case)?
Should we rely solely on the ATA187 configuration in CUCM and the settings on the fax machine ? (Speed, ECM, etc)
Gateway/CUBE 15.3, CUCM 9.1.1a
Regards,
Erik
Sent from Cisco Technical Support iPhone AppThe following is an example config you can make use of with an SIP enviroment.
voice service voip
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
no supplementary-service sip refer
fax protocol pass-through g711alaw
h323
modem passthrough nse codec g711alaw
sip
registrar server expires max 3600 min 3600
asserted-id pai
privacy pstn
options-ping 60
midcall-signaling passthru
dial-peer voice 200 voip
description ----- WYCHODZACE SIP-T ------
destination-pattern [1-9]........
modem passthrough nse codec g711alaw
session protocol sipv2
session target ipv4:SIP-SRV:5020
voice-class codec 23
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
voice-class sip pass-thru headers unsupp
voice-class sip pass-thru content unsupp
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
req-qos controlled-load audio
fax-relay ecm disable
fax protocol pass-through g711alaw
no vad
dial-peer voice 101 voip
description **Outgoing Call to CUCM**
destination-pattern 895216...
session target ipv4:CCM1
voice-class codec 1
voice-class h323 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay h245-alphanumeric
fax protocol pass-through g711alaw -
SIP trunk incoming and outgoing calls issue
Hi Everyone,
We recently installad a SIP trunk and terminated on CUBE and CUCM but we have issues on incoming and outgoing calls, When someone dial in from outside he keeps listening the dailing ring even after we pick up the phone and at the end the callers time exipres and call gets disconnected.
For Dailing out, the dialed number rings and caller hear the dailing ring as well but if someone pick the phone it apprears that call is connected but no audio in it, dead air.
Our call flow is as
IP Phones => CUCM --->SIPTRUNK--->CUBE=>SIPTRUNK=>SP
I have attached the config for CUBE and debug ccsip messages output for both incoming and outgoing calls.
Please if some help in sorting out this issue, Thanks in Advance
TasneemInbound call>>>>
The reason you are experiencing this is that your CUBE is requesting PRACK and your provider is not responding to it..
Here we have your cube sending 180 ringing with "Require 100rel"..This was sent several times and your ITSP didnt respond probably because they do not support 100rel...(It is Huwaei after all, they do what they like)
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKkpw18wp35hfw2c23h51ww6a8aT19871
From: ;tag=sbc080552fph4hp-CC-25
To: ;tag=256F3440-12C
Date: Thu, 16 Jan 2014 13:31:34 GMT
Call-ID: isbc6818c4kfhaa4ca1k5f8k1awsh52f1ccw@SoftX3000
CSeq: 1 INVITE
Require: 100rel
RSeq: 2507
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: "TEST STC" ;party=called;screen=yes;privacy=off
Contact:
Record-Route:
Server: Cisco-SIPGateway/IOS-15.2.4.M2
Content-Length: 0
AFter the CUBE didnt get any response, it then replied with Gateway Timeout...
Jan 16 13:31:54.550: //31880/5A4406E48184/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 504 Gateway Timeout
Via: SIP/2.0/UDP 10.205.20.50:5060;branch=z9hG4bKkpw18wp35hfw2c23h51ww6a8aT19871
From: ;tag=sbc080552fph4hp-CC-25
To: ;tag=256F3440-12C
Call-ID: isbc6818c4kfhaa4ca1k5f8k1awsh52f1ccw@SoftX3000
CSeq: 1 INVITE
Reason: Q.850;cause=102
Content-Length: 0
I suggest you disable this parameter..and test again
voice service voip
sip
rel1xx disable
Please rate all useful posts
"The essence of christianity is not the enthronement but the obliteration of self --William Barclay" -
Callcentric SIP Trunk (ITSP -- 2811 CUBE -- CUCM 8.6
I have a SIP trunk from call centric that goes into my lab gear - they appear to be a good sip service due to cost but I'm having some trouble getting calls to route correctly. The call flow is Callcentric.com ITSP (SIP) --> 2811 (acting as cube) -->SIP Trunk --> CUCM 8.6. Phones are registered to CUCM.
I have the sip trunk registered and calls come in to the router (I see them in ccsip message/call debugs) The 2811 running 15.1(4)M7). Callcentric sends the username of the customer in the sip Invite instead of the called number, the called number is in the TO field. I have several DID’s from Callcentric (18452055544, 18452055545, 18452055546) for my lab. There are a few configs on here for CME where the customer number (17772253754) is simply translated to their phone DN - which is fine if you only have 1 DN with callcentric but more than 1 and thats not feasible since every inbound did will be matched to that 17772253754 translation/phone dn.
I’m using the a guide from http://tblog.cisco.be/2011/02/17/cube-conditional-sip-profiles/ using the Copy function as described http://www.cisco.com/c/en/us/products/collateral/ios-nx-os-software/ios-software-release-15-1-3-t/product_bulletin_c25-635704.html
I haven’t been able to find anything where they actually explain all the header fields so Its mostly trial and error.. so far mostly error. I think I’m close.. but who knows. Any assistance would be greatly appreciated
voice class sip-profiles 1
request INVITE peer-header sip TO copy ".sip:(.*)@." u01
request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
CUCM (single/pub)- 192.168.1.200
2811 acting as cube - 192.168.1.203
Calling Number - 18165297500
Called Number - 18452055544
vrtr1#show sip register status
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
17772253754 -1 20 yes
vrtr1#
The Call Setup Information is:
Call Control Block (CCB) : 0x49646C28
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 18165297500
Called Number : 17772253754 (my customer number not called number)
Source IP Address (Sig ): 192.168.1.203 (my 2811 router)
Destn SIP Req Addr:Port : 204.11.192.159:5080
Destn SIP Resp Addr:Port : 204.11.192.159:5080
Destination Name : 204.11.192.159
Feb 14 11:20:53.303: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
f: <sip:[email protected]>;tag=3601387252-874282
t: <sip:[email protected]>
i: [email protected]
CSeq: 1 INVITE
Max-Forwards: 8
m: <sip:[email protected]:5080;transport=udp>
Supported: timer
c: application/sdp
l: 350
v=0
o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.159
s=sip call
c=IN IP4 204.11.192.159
t=0 0
m=audio 61094 RTP/AVP 18 0 8 101
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
From: <sip:[email protected]>;tag=3601387252-874282
To: <sip:[email protected]>
Date: Fri, 14 Feb 2014 17:20:53 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Feb 14 11:20:53.327: //936/310B294680AD/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
From: <sip:[email protected]>;tag=3601387252-874282
To: <sip:[email protected]>;tag=35399D8-63
Date: Fri, 14 Feb 2014 17:20:53 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=1
Content-Length: 0
Feb 14 11:20:53.419: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-805ff2443b18502ff96181045b62dd74
f: <sip:[email protected]>;tag=3601387252-874282
t: <sip:[email protected]>;tag=35399D8-63
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 10
l: 0
u all
Feb 14 11:20:57.067: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:18452055544;cic=0288;rn=6465471001;[email protected]:5070 SIP/2.0
v: SIP/2.0/UDP 204.11.192.159:5080;branch=z9hG4bK-6bceae47efe9f53b4234698a32ac8beb
f: <sip:[email protected]>;tag=3601387252-874282
t: <sip:[email protected]>;tag=35399D8-63
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 8
l: 0
************************** Running Config **************************
sh run
vrtr1#sh running-config
Building configuration...
Current configuration : 4189 bytes
! Last configuration change at 00:34:03 CST Fri Feb 14 2014
! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
! NVRAM config last updated at 20:26:58 CST Thu Feb 13 2014
version 15.1
service timestamps debug datetime msec localtime
service timestamps log datetime msec localtime
no service password-encryption
hostname vrtr1
boot-start-marker
boot system flash:
boot system flash flash:c2800nm-ipvoicek9-mz.151-4.M7.bin
boot-end-marker
card type t1 0 0
logging buffered 4096 notifications
enable password cisco
no aaa new-model
memory-size iomem 5
clock timezone CST -6 0
clock summer-time CST recurring
no network-clock-participate wic 0
dot11 syslog
ip source-route
ip cef
ip name-server 192.168.1.9
no ipv6 cef
multilink bundle-name authenticated
voice service voip
ip address trusted list
ipv4 192.168.1.0 255.255.255.0
ipv4 204.11.192.0 255.255.255.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
registrar server expires max 1800 min 1800
localhost dns:callcentric.com
outbound-proxy dns:callcentric.com
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
voice class sip-profiles 1
request INVITE peer-header sip TO copy ".sip:(.*)@." u01
request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:\u01@\1"
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2811 sn FTX1133A4QR
controller T1 0/0/0
cablelength long 0db
interface FastEthernet0/0
description ** LAN **
ip address 192.168.1.203 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.1.203
interface FastEthernet0/1
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 0.0.0.0 192.168.1.1
snmp mib persist circuit
control-plane
voice-port 0/1/0
voice-port 0/1/1
voice-port 0/1/2
voice-port 0/1/3
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server 192.168.1.200
ccm-manager config
mgcp
mgcp call-agent 192.168.1.200 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp package-capability fxr-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp rtp payload-type g726r16 static
mgcp bind control source-interface FastEthernet0/0
mgcp bind media source-interface FastEthernet0/0
mgcp profile default
dial-peer voice 999100 pots
service mgcpapp
port 0/1/0
dial-peer voice 999101 pots
service mgcpapp
port 0/1/1
dial-peer voice 999102 pots
service mgcpapp
port 0/1/2
dial-peer voice 999103 pots
service mgcpapp
port 0/1/3
dial-peer voice 999010 pots
service mgcpapp
port 0/1/0
dial-peer voice 6 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 17772253754
voice-class sip profiles 1
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 7 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 1845205554[4-5]
voice-class sip profiles 1
dtmf-relay h245-alphanumeric
no vad
sip-ua
credentials username 17772253754 password 7 106C1B49111F17194D realm callcentric.com
authentication username 17772253754 password 7 08035E1E1D11000553 realm callcentric.com
no remote-party-id
retry invite 2
retry register 10
timers connect 100
mwi-server dns:callcentric.com expires 3600 port 5060 transport udp
registrar dns:callcentric.com expires 3600
sip-server dns:callcentric.com
host-registrar
line con 0
line aux 0
line vty 0 4
password cisco
login
transport input all
scheduler allocate 20000 1000
ntp server 199.102.46.72
ntp server 23.227.162.123 prefer
end
exitThank you for the reply. I've updated the dial-peers as sugested. I'm now seeing an invite go out to my CUCM however the call fails with a 403 (forbidden) which appears to come from the ITSP (Callcentric). I've included a new set of ccsip message debugs and the dial-peers as adjusted. Please let me know what you think.
dial-peer voice 6 voip
description ## INBOUND CALL from ITSP ##
session protocol sipv2
session target sip-server
incoming called-number 17772253754
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
dial-peer voice 100 voip
description ## INBOUND DID to CUCM ##
destination-pattern 17772253754
session protocol sipv2
session target ipv4:192.168.1.200
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
dial-peer voice 7 voip
description ## INBOUND DID to CUCM ##
session protocol sipv2
session target ipv4:192.168.1.200
incoming called-number 1845205554[4-5]
voice-class sip profiles 1
dtmf-relay rtp-nte
no vad
Feb 15 10:18:11.424: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
f: ;tag=3601469891-655
t: [email protected]>
i: [email protected]
CSeq: 1 INVITE
Max-Forwards: 8
m:
Supported: timer
c: application/sdp
l: 350
v=0
o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.164
s=sip call
c=IN IP4 204.11.192.164
t=0 0
m=audio 61782 RTP/AVP 18 0 8 101
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
a=setup:actpass
Feb 15 10:18:11.456: //2419/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
From: ;tag=3601469891-655
To: [email protected]>
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Feb 15 10:18:11.460: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:@192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2570262061-2509443555-2182021079-2501285341
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1392481091
Contact:
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 7
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273
v=0
o=CiscoSystemsSIP-GW-UserAgent 2786 1511 IN IP4 192.168.1.203
s=SIP Call
c=IN IP4 192.168.1.203
t=0 0
m=audio 18168 RTP/AVP 18 101
c=IN IP4 192.168.1.203
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Feb 15 10:18:11.552: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35;rport=57100;received=24.123.98.94
f: [email protected]>;tag=8408644-12C8
t:
i: [email protected]
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="callcentric.com", domain="sip:callcentric.com", nonce="8ae6b7b1cea74cf401e8a26fd3c7371b", opaque="", stale=TRUE, algorithm=MD5
l: 0
Feb 15 10:18:11.560: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9A91F35
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Feb 15 10:18:11.560: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:@192.168.1.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 2570262061-2509443555-2182021079-2501285341
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1392481091
Contact:
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="17772253754",realm="callcentric.com",uri="sip:[email protected]:5060",response="a381f10fbbfbd255b444569fef0dddfe",nonce="8ae6b7b1cea74cf401e8a26fd3c7371b",opaque="",algorithm=MD5
Max-Forwards: 7
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 273
v=0
o=CiscoSystemsSIP-GW-UserAgent 2786 1511 IN IP4 192.168.1.203
s=SIP Call
c=IN IP4 192.168.1.203
t=0 0
m=audio 18168 RTP/AVP 18 101
c=IN IP4 192.168.1.203
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Feb 15 10:18:11.648: //2420/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Incorrect Authentication
v: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3;rport=57100;received=24.123.98.94
f: [email protected]>;tag=8408644-12C8
t:
i: [email protected]
CSeq: 102 INVITE
l: 0
Feb 15 10:18:11.660: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK9AA1BA3
From: [email protected]>;tag=8408644-12C8
To:
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
Feb 15 10:18:11.660: //2419/9933162D820E/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
From: ;tag=3601469891-655
To: [email protected]>;tag=8408714-B60
Date: Sat, 15 Feb 2014 16:18:11 GMT
Call-ID: [email protected]
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=57
Content-Length: 0
Feb 15 10:18:11.752: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
apsc-vrtr1#ACK sip:[email protected]:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-d78945b444598bc22c8509d069f4789d
f: ;tag=3601469891-655
t: [email protected]>;tag=8408714-B60
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 10
l: 0
vrtr1#u al
Feb 15 10:18:14.776: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:18452055544;cic=0288;rn=6465471001;[email protected]:5070 SIP/2.0
v: SIP/2.0/UDP 204.11.192.164:5080;branch=z9hG4bK-e437c2c5cac5f1a6e147c1cd7c98aad7
f: ;tag=3601469891-655
t: [email protected]>;tag=8408714-B60
i: [email protected]
CSeq: 1 ACK
Max-Forwards: 8
l: 0 -
NICE recorder - one way or garbled audio over SIP trunk.
Customer with CUCM 10.5.2 trying to integrate with a NICE recording solution. Everything configured per NICE documentation and rechecked that several times with the NICE vendor. However, if calling to/from the PSTN or legacy Nortel PBX to a Cisco 78xx or IP Communicator soft phone - we get one-way audio pushed out the SIP trunk to the NICE system. If it's a Cisco to Cisco phone call - the audio is garbled.
Has anyone experienced this issue with this type of integration - or the same issue with a SIP trunk to the CUCM to another system at all? We're at a loss here.
Thank you.This document should help you:
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a008009484b.shtml -
Issue with instant ringback when using sip trunk to SP
Hi all,
We use CUCM 8.0.2.
We have a SIP trunk to a SP connected via one of our Cisco 2911 routers configured as a CUBE.
Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.0(1)M3, RELEASE SOFTWARE (fc2)
c2900-universalk9-mz.SPA.150-1.M3.bin
Cisco CISCO2911/K9 (revision 1.0)
Technology Package License Information for Module:'c2900'
Technology Technology-package
Current Type
ipbase ipbasek9 Permanent
security securityk9 Permanent
uc uck9 Permanent
data None None
We also have several ISDN lines that run out via various Cisco routers configured as H323 gateways.
We use 7945 and CIPC for our phones.
We're having an issue with calls going via the SIP trunk where we hear ringing instantly after dialling - but before the actual device at the other end starts ringing (considerable difference).
Using the SIP trunk: If I make a call to my mobile phone - I hear ringing instantly - about 3 rings before my mobile phone actually starts ringing - undesireable.
Using H323 gateway: If I make a call to my mobile phone - I hear silence for a bit - then ringing when the mobile starts ringing - desired.
Using SIP trunk: If I make a call to a landline that is ready - it rings instantly for at least 1 ring - before the actual phone I'm calling starts ringing - undesireable.
Using H323 gateway: There is a momentary pause before hearing ringing on my phone and the phone I dialled - desired.
Using SIP trunk: If I make a call to a landline that is off-hook (with no call-waiting/etc.) - it rings once and then returns the busy signal (the worst issue) - undesireable.
Using H323 gateway: There is a momentary pause before hearing busy signal - desired.
Phone to phone internally (same network): Operates as expected (instantly rings locally and on the phone I'm calling). Between phones that utilise the SIP trunk and phones that utilise the H323 gateways within the same network - communication is instant and as expected.
Any ideas why this happens and how to stop it?
I want it to not ring until the situation is known and that it can provide the appropriate feedback (ringing/busy/etc.).
Some possibly relevant config (note that there is a known bug with this IOS that meant I had to declare the codec in each dial-peer as the voice class would not work):
voice service voip
address-hiding
mode border-element
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
header-passing error-passthru
early-offer forced
midcall-signaling passthru
interface GigabitEthernet0/0
ip address x.x.x.x 255.255.255.252
ip access-group acl.SIP-IN in
no ip redirects
no ip unreachables
ip verify unicast reverse-path
ip virtual-reassembly
duplex full
speed 100
no cdp enable
gateway
timer receive-rtp 1200
sip-ua
connection-reuse
gatekeeper
shutdown
dial-peer voice 1 voip
description *** INBOUND CALLS FROM CARRIER ***
translation-profile incoming SIPTRUNK-INCOMING
session protocol sipv2
incoming called-number #blah blah#
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 61 voip
description **** WA, SA AND NT NUMBERS ****
destination-pattern 0[8]........
session protocol sipv2
session target ipv4:<MY SP's SIP SERVER>
incoming called-number 0[8]........
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 81 voip
description **** MOBILE NUMBERS ****
destination-pattern 0[4]........
session protocol sipv2
session target ipv4:<MY SP's SIP SERVER>
incoming called-number 0[4]........
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
dial-peer voice 500 voip
description *** INBOUND SIP TRUNK TO CUCM PUB ***
translation-profile outgoing SIPTRUNK-CALLING-ADD-0
preference 1
destination-pattern 5[12]..
session protocol sipv2
session target ipv4:<OUR CUCM PUBLISHER IP>
dtmf-relay rtp-nte
codec g711alaw
ip qos dscp cs5 media
no vad
Any help or a point in the right direction would be greatly appreciated.
Cheers,
BrettI ended up resolving this issue as follows:
In CUCM, under Device > Device Settings > SIP Profile.
I modifed the profile relevant to my SIP trunk, under the "Trunk Specific Configuration", I set "SIP Rel1XX Options" from "Disabled" to "Send PRACK if 1xx Contains SDP".
Now, I get the expected delay before hearing ringback.
Solved! -
DTMF issues on SIP trunk to Verizon
Were you able to resolve this problem? I am having an identical issue also with Verizon.
Our topology and symptoms are as follows:
Outside phone -> PSTN -> Vzn SBC -> Vzn SIP trunk -> CUBE -> CUCM / VM system
DTMF tones generated by an IP phone are heard and recognized by an outside (off-net) phone/system as you would expect. However, DTMF tones generated by an outside (off-net) phone are not recognized by our voice mail system. When listening to the DTMF tone on an IP phone, it sounds very distorted and faint. A sniffer trace performed on the CUBE shows RFC 2833 NTEs being received from Verizon, and they appear to be properly relayed by the CUBE to the destination. Payload type negotiated for both legs is 101.
We are running CUCM 6.1.5. We have a CUBE router between CUCM and the Verizon SIP trunk. The CUBE router is running 12.4(24)T3 with the IPIPGW feature set. Our voice mail system is an AVST CallXpress system running v7.9 software. To CUCM the AVST voice mail ports appear as DNs assigned to several SCCP 7940 phones (DNs are part of a hunt group, hunt pilot = vm pilot). The AVST masquerades and registers as the 7940 phones.
I tried applying the "dtmf-interworking rtp-nte" both globally and at the dial-peer level with no success. Attached is the debug output you suggested. -
SIP trunk INCOMING/OUTGOING calling issue
Dears,
i am facing issue i.e inbound / outbound calls are not wokring after configuring SIP trunk . the call flow is
SERVICEPROVIDER--------sip trunk------VG --------------sip trunk ----------CUCM
For Outbound calls ---error-----tot tot tot
For Inbound Calls -----error--- Silence
Below are the configuration and debugs:Appreciate if some provide me Sip configuration and troubleshooting guide...Thankyou
-
Unable to place call on calls on hold - SIP Trunk from CUCM to CUBE and from CUBE to ISTP
Hi Cisco Community,
I have a SIP Trunk setup between the CUCM and CUBE and another SIP Dial Peers from the CUBE to the ITSP. All incoming/outgoing calls, DTMF-Relay works well except one thing which is the ability to hold the call.
On the SIP Trunk from the CUCM to the CUBE, I did not select “MTP” because when I do so, I am forced to select my preferred MTP codec which when selected G.729/G.729a, all my outgoing calls goes out using G.729r8. This codec works well for most of the calls until the ITSP replies back with G.729br8. When this condition occur, my call simply fails (this is very intermittent and only some random numbers).
That said, I have some issues with DTMF Relay when I select MTP on the SIP Trunk. DTMF Relay only works if the call is G.729r8 all the way from the CUCM to the ITSP. If the ITSP replies back with G.729br8, the call might established but will simply be “voice-only”.
The current setup is no MTP is selected and everything is working perfectly. I am happy with that until I place a call on hold, which when I do so, the call immediately terminate. Could you please help me understand why?
I have all media resources configured such as G.729r8 MTP, G.729br8 MTP, G.711u MTP, Transcoding with all codecs, etc. All MRG and MRGL are configured on all devices and SIP Trunks.
Below is an example of a call that is connected with the current setup:
Note:
IP: 10.18.81.2 (CUBE)
IP: 10.18.81.11 (CUCM SUB)
IP: 10.111.111.254 (ITSP SBC)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:29.938: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:10.18.81.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
Session-Expires: 1800
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Content-Length: 0
Nov 30 11:44:29.942: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:29.946: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1417347869
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 301
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6958 IN IP4 10.18.81.2
s=SIP Call
c=I
PM-HO-VG-01#N IP4 10.18.81.2
t=0 0
m=audio 22256 RTP/AVP 18 0 8 101
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Nov 30 11:44:29.950: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Nov 30 11:44:30.658: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:30.662: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:31.226: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Session Progress
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Session: Media
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
X-BroadWorks-Correlation-Info: bbf9
PM-HO-VG-01#4839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC72218
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 101 INVITE
Timestamp: 1417347869
Supported:
Contact: <sip:[email protected]:5060;transport=udp>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp,application/xml
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 355
v=0
o=BroadWorks 316169737 1 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.630: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC81D00
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:31.634: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72063a5aba5d
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2747 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:31.726: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72075e3a02c1
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:29 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Type: application/sdp
Content-Length: 236
v=0
o=CiscoSystemsCCM-SIP 9082578 1 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.80.40
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
PM-HO-VG-01#
Nov 30 11:44:31.730: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 10.18.80.40
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
PM-HO-VG-01#sh sip
PM-HO-VG-01#sh sip-ua call
PM-HO-VG-01#sh sip-ua calls
Total SIP call legs:2, User Agent Client:1, User Agent Server:1
SIP UAC CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 27218091323
Called Number : 0862000000
Bit Flags : 0xC04018 0x10000100 0x0
CC Call ID : 64511
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.111.111.254]:5060
Destn SIP Resp Addr:Port: [10.111.111.254]:5060
Destination Name : 10.111.111.254
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64511
Stream Type : voice+dtmf (0)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22256
Media Dest IP Addr:Port : [10.111.111.254]:20074
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Client(UAC) calls: 1
SIP UAS CALL INFO
Call 1
SIP Call ID : [email protected]
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 0218091323
Called Number : 0862000000
Bit Flags : 0xC0401E 0x10000100 0x80004
CC Call ID : 64510
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : [10.18.81.11]:5060
Destn SIP Resp Addr:Port: [10.18.81.11]:5060
Destination Name : 10.18.81.11
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 64510
Stream Type : voice+dtmf (1)
Stream Media Addr Type : 1
Negotiated Codec : g729br8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [10.18.81.2]:22350
Media Dest IP Addr:Port : [10.18.80.40]:21928
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Server(UAS) calls: 1
PM-HO-VG-01#
PM-HO-VG-01#
PM-HO-VG-01#
As you can see, the call is connected and everything is working perfectly. When I press the hold button, here is what I get:
NOTE: I have # debug ccsip messages and #debug ccsip calls (running)
PM-HO-VG-01#
PM-HO-VG-01#
Nov 30 11:44:49.210: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 9082578 2 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:8000
b=AS:8
t=0 0
m=audio 21928 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Nov 30 11:44:49.218: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.218: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.18.81.2:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7676 6959 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22256 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3EC9241
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 102 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
X-BroadWorks-Correlation-Info: bbf94839-a234-4237-95e6-a7037322f0f4
Content-Type: application/sdp
Content-Length: 360
v=0
o=BroadWorks 316169737 2 IN IP4 10.111.111.254
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 20074 RTP/AVP 18 101 100
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
a=inactive
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_ACTIVE
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.278: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.282: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECA2633
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.282: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720852ab8b92
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 271
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2748 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 0.0.0.0
t=0 0
m=audio 22350 RTP/AVP 18 101
c=IN IP4 0.0.0.0
a=inactive
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Nov 30 11:44:49.282: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e72094953dfea
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
Nov 30 11:44:49.290: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Contact: <sip:[email protected]:5060>
Content-Length: 0
Nov 30 11:44:49.294: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Content-Length: 0
Nov 30 11:44:49.294: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1020645888-0000065536-0000124117-0189862410
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1417347889
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Length: 0
Nov 30 11:44:49.338: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECB16F3
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
CSeq: 103 INVITE
Timestamp: 1417347889
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Supported:
Accept: application/media_control+xml,application/sdp,application/xml
Contact: <sip:[email protected]:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 306
v=0
o=BroadWorks 316169737 3 IN IP4 10.111.111.254
s=-
c=IN IP4 10.111.111.254
t=0 0
m=audio 20074 RTP/AVP 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
Nov 30 11:44:49.342: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 2
PM-HO-VG-01#00 OK
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720a6918040f
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:[email protected]:5060>
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-15.2.4.M5
Supported: timer
Content-Type: application/sdp
Content-Length: 289
v=0
o=CiscoSystemsSIP-GW-UserAgent 7965 2749 IN IP4 10.18.81.2
s=SIP Call
c=IN IP4 10.18.81.2
t=0 0
m=audio 22350 RTP/AVP 18 101 19
c=IN IP4 10.18.81.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:19 CN/8000
a=ptime:20
Nov 30 11:44:49.350: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.11:5060;branch=z9hG4bK2e720b594cd517
From: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
To: <sip:[email protected]>;tag=3C365010-1E42
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 213
v=0
o=CiscoSystemsCCM-SIP 9082578 3 IN IP4 10.18.81.11
s=SIP Call
c=IN IP4 10.18.81.10
t=0 0
m=audio 4000 RTP/AVP 18
a=X-cisco-media:umoh
a=rtpmap:18 G729/8000
a=ptime:20
a=fmtp:18 annexb=no
a=sendonly
Nov 30 11:44:49.354: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 101 BYE
Reason: Q.850;cause=86
P-RTP-Stat: PS=874,OS=17480,PR=872,OR=17440,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.354: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Max-Forwards: 70
Timestamp: 1417347889
CSeq: 104 BYE
Reason: Q.850;cause=65
P-RTP-Stat: PS=872,OS=17440,PR=952,OR=19040,PL=0,JI=0,LA=0,DU=17
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 Race Condition
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECD1ECD
From: "Bianca Africa" <sip:[email protected]>;tag=3C364D44-9E2
To: <sip:[email protected]>;tag=71913148-1417348035284
Call-ID: [email protected]
Timestamp: 1417347889
CSeq: 104 BYE
Content-Length: 0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D7B1458
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 27218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.111.111.254:5060
Destn SIP Resp Addr:Port : 10.111.111.254:5060
Destination Name : 10.111.111.254
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22256
Destn IP Address (Media): 10.111.111.254
Destn IP Port (Media): 20074
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.374: //64511/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 65
Disconnect Cause (SIP) : 200
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.18.81.2:5060;branch=z9hG4bK3ECC55
From: <sip:[email protected]>;tag=3C365010-1E42
To: "Bianca Africa" <sip:[email protected]>;tag=9082578~cdf4c5a6-dd2b-4c71-bca0-b262ad997720-44517224
Date: Sun, 30 Nov 2014 11:44:49 GMT
Call-ID: [email protected]
CSeq: 101 BYE
Content-Length: 0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x3D816D70
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 0218091323
Called Number : 0862000000
Source IP Address (Sig ): 10.18.81.2
Destn SIP Req Addr:Port : 10.18.81.11:5060
Destn SIP Resp Addr:Port : 10.18.81.11:5060
Destination Name : 10.18.81.11
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g729br8
Negotiated Codec Bytes : 20
Nego. Codec payload : 18 (tx), 18 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.18.81.2
Source IP Port (Media): 22350
Destn IP Address (Media): 0.0.0.0
Destn IP Port (Media): 21928
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 30 11:44:49.406: //64510/3CD5D2000001/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 86
Disconnect Cause (SIP) : 200
PM-HO-VG-01#Hi Manish,
Again, excellent feedback. Much appreciated.
I will try the commands suggested above and see if I can get DTMF to work correctly while interworking H.323 and SIP.
But my ultimate goal is to have SIP all way from the CUCM to the CUBE and from the CUBE to the ITSP.
If I enable SIP Early Offer with MTP on the CUCM going to the CUBE, all SIP Invite sent from the CUCM to the CUBE uses G.729r8 as the codec and once the call is established using G.729r8 should the ITSP reply with that codec, the call succeed and I am able to see an active MTP session using G.729 when I issue the command # show sccp connections.
One thing that I saw is that my ITSP love so much sending G.729br8 most of the times, so even if using SIP EO with MTP on the SIP Trunk to the CUBE, when I sent my INVITE out from the CUCM to the CUBE using G.729r8, specially on call center numbers such as 0800 numbers, you will see that the call established but the codec being negotiated is G.729br8 which is voice only (missing DTMF).
I will be doing some intensive test again later on this week and will send the logs.
Here is my question to both of you:
Which is the best way of having a proper SIP to SIP setup all the way that will not pose any problem?
Do I have to enable Early Offer on the SIP Profile used by the CUBE SIP Trunk or should I use the normal Standard SIP Profile? Do I need to enable MTP on my CUBE SIP Trunk or not?
From the CUBE point of view, I have a voice class codec that support G.729r8 or G.729br8 and the DTMF Relay method supported by the ITSP is RFC 2833.
I will send more logs for each scenario. I think that we are getting close to the resolution of this problem.
Thanks again for your support fellows. -
Configuring Level3 SIP trunk with Lync 2013
Hi, I ran into some issues trying to configure SIP trunk from Level 3 and I was hoping someone here can help. We have our mediation server collocated with FE and SIP traffic goes from public IP, port 5060 via NAT, to local IP on FE, port 5060.
Level 3 provided us with one signaling IP and two RTP IPs.
I tried multiple trunk configuration settings and I can see that when I'm placing a call from Lync to an outside number I'm getting INVITE from Level 3 signaling IP, the session is established, phone rings, but there is no audio on either side. There's also
a METHOD NOT ALLOWED message coming from them, which doesn't tell me much about what's happening.
If I call to a Level 3 DID (assigned to my Lync user account) there's also INVITE from their side, but later I receive a CANCEL from them due to idle session. The phone never rings.
Questions:
1) Does anyone have Level 3 SIP trunks configured and can share their Get-TrunkConfiguration settings? What settings should I have for encryption, refer, sessionTimer / RTCP, and others? Level 3 refuses to provide any additional information besides IPs.
2) Do I understand this correctly that when configuring PSTN gateways in topology, one of the RTP IPs should be entered in the "alternate media IP" field? We have SIP trunks from another provider (which work fine), and they only use one IP
for everything, so I don't have any experience configuring separate SIP and media IPs with Lync.
Thanks, and let me know if I should provide additional info.Hi,
On Lync topology PSTN gateways interface, please check if you enter gateway listening port 5060 and enable TCP option.
Please also check if you enable refer support on Lync Server Control Panel, if you enable it please uncheck it.
You can compare the trunk configuration for Level 3 in the part “Sample Trunk Configuration for Level 3” in the link below with yours’, it is for Lync server 2010 but similar for Lync server 2013:
http://blogs.technet.com/b/nexthop/archive/2013/04/10/configuring-lync-2010-server-to-work-with-level-3-sip-trunking-services.aspx
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support -
Hi,
I have uc320w with 2.3.2 (6) firmware and 2x SPA504g, 1x 508g, 1xspa303, 1 annalog phone @ fxs port. Both 504g are conected via wifi module. All phones are asign and work well in PBX system and ansfering all external calls even tranfers works well. Only internal dialing and internal conference calls doesn't work, phones ring, but audio not, or just hear myself. Whe is an issue?Yes they have cisco wireless-N bridge for phone adapters. Right no way audio between any phones, even ones they have no more the 10ft of cat5e utp. there is no swich at all, they are are conected directly to the uc320w and have own power adapters. For external calls is used SIP trunk only. WAN connection is direct to provider passing cisco DPC3825 cable modem in bridge mode (SHAW canada) with static IP address. Uc320w is used to route only voice and has WAN connection 50Mbit down and 5Mbit up, 711u is used for SIP trunks. Ping to SIP provider server is around 12ms. SIP is one line(one number) with 4 DIDs
Regards
Eric -
Video only enabled when call is initiated from one direction across SIP Trunk
wonder If anyone can shed some light on this.
I have an issue between two cucm clusters, tied together with a SIP trunk.
If we dial from Australia to the US there is two way video and audio. If the US calls Australia, there is only audio. I have run a test call from the US through VLT and have found the following SDP's (see below). When The US make a video enabled call to australia the message "Video is not available, Remote party has video off" on the US phone screen.
Both clusters have the SIP trunk set up with the same codec settings and video bandwidth between reqions and locations. the SIP trunk is configured pretty much stock standard and identical at both ends, yet the SDP seem to want to negotiate different Video Parameters (again see SDP's below). CUCM in australia is 10.61.2.82.
what other settings can I check to get video to work when calls get initiated from either direction,...................
both phones are SIP 8941's, again audio is no problem in both directions.
=======this is from the phone in Australia to the CUCM in australia phone IP 10.61.4.112======================================
45870304.002 |09:02:07.941 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.61.4.112 on port 34271 index 53563 with 2089 bytes:
[344530309,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.61.2.82:5060;branch=z9hG4bKe0103892bbb75
From: "Anonymous" <sip:[email protected]>;tag=109791678~1b5af941-cea2-4a00-a0bd-15a532224d7d-59374526
To: <sip:[email protected]>;tag=5057a887bfdd550c0d321a20-7f843426
Call-ID: [email protected]
Date: Wed, 29 Apr 2015 23:02:07 GMT
CSeq: 101 INVITE
Server: Cisco-CP8941/9.4.2
Contact: <sip:[email protected]:34271;transport=tcp>;video
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Dennis Mink - 33935" <sip:[email protected]>;party=called;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Content-Length: 966
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 28123 0 IN IP4 10.61.4.112
s=SIP Call
t=0 0
m=audio 16736 RTP/AVP 0 8 18 102 9 116 101
c=IN IP4 10.61.4.112
a=trafficclass:conversational.audio.avconf.aq:admitted
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 16738 RTP/AVP 126 97
c=IN IP4 10.61.4.112
b=TIAS:2000000
a=trafficclass:conversational.video.avconf.aq:admitted <----this is missing from US SDP
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=428014;packetization-mode=1;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200;max-rcmd-nalu-size=1300
a=imageattr:126 send * recv [x=640,y=480]
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=428014;packetization-mode=0;level-asymmetry-allowed=1;max-mbps=36000;max-fs=1200
a=imageattr:97 send * recv [x=640,y=480]
a=rtcp-fb:* ccm tmmbr
a=sendrecv
============below is coming from the US (phone IP is 10.1.109.81)================
04/30/2015 09:02:08.169 Send 10.61.4.112 SIP ACK bfa99a00-541162ed-71da57-52023d0a NotAvail
SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.61.4.112 on port 34271 index 53563
[344530326,NET]
ACK sip:[email protected]:34271;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.61.2.82:5060;branch=z9hG4bKe010481f320b08
From: "Anonymous" <sip:[email protected]>;tag=109791678~1b5af941-cea2-4a00-a0bd-15a532224d7d-59374526
To: <sip:[email protected]>;tag=5057a887bfdd550c0d321a20-7f843426
Date: Wed, 29 Apr 2015 23:02:05 GMT
Call-ID: [email protected]
User-Agent: Cisco-CUCM10.0
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 456
SDP Message
====================================================
v=0
o=CiscoSystemsCCM-SIP 109791678 1 IN IP4 10.61.2.82
s=SIP Call
c=IN IP4 10.1.109.81
b=TIAS:8000
b=AS:8
t=0 0
m=audio 16412 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=trafficclass:conversational.audio.aq:admitted <---what does this do here, and how?
m=video 0 RTP/SAVP 31 34 96 97 <-----------port 0. why?
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:96 H263-1998/90000
a=rtpmap:97 H264/90000
a=content:main
a=inactiveHi Dennis,
On US phone SDP media attribute is inactive.
a=rtpmap:97 H264/90000
a=content:main
a=inactive
Are you sure that audio works ? Can you please share all the SIP messages of both the scenarios.
Thanks
Manish -
Cucm , Cube via Sip and Sip Trunk to ISP , Outgoing calls not working
Hi
We have issue with the outgoing calls to sip trunk
Below is the config and the debugs
It will be great if you give your thoughts since we have stuck here
My thoughts are:
i see that for unknown reason the called number is going with 4 digits instead of 8 digits
i dont see any sip message comming from ISP
Maybe the call not going there ? to isp trunk? From the trace the call hit the correct dialpeer 888 but i see 4 digits as a called number , but i dodnt understant the reason to translated in 4 digits the called number.Not apply a translation rule for that
confused!!!
Calling Numbner:22324086
Called Number: 23823690
CUCM:192.168.1.241 and 242
CUBE:192.168.1.10
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*\(....\)/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^\(9\)\(.......$\)/ /99\2/
rule 3 /^\(2\)\(.......$\)/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*\(....\)/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^\(9\)\(.......$\)/ /99\2/
rule 3 /^\(2\)\(.......$\)/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vadHi Aok
I change the default value for IPVMS from g711ulaw to g711alaw but the results remained the same
Also i have restarted the IPVMS
SIP-GW#
SIP-GW#
*Mar 5 14:19:57.854: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 38874 2 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:64000
b=AS:64
t=0 0
m=audio 24784 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:57.878: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1362493197
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3807 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 101
c=IN IP4 10.249.13.130
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.878: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:57.926: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 102 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 213
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 2 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=inactive
*Mar 5 14:19:57.942: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5774 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 101
c=IN IP4 192.168.1.10
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.946: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3562A4
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:57.946: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK798246ab3597
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
*Mar 5 14:19:58.146: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Length: 0
*Mar 5 14:19:58.158: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1362493198
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:58.158: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:58.218: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 103 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 216
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 3 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 18 96 99
a=rtpmap:96 AMR/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
a=sendrecv
*Mar 5 14:19:58.234: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 283
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5775 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 18 101
c=IN IP4 192.168.1.10
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:58.242: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7985648033f2
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 192
v=0
o=CiscoSystemsCCM-SIP 38874 3 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 192.168.1.241
t=0 0
m=audio 4000 RTP/AVP 8
a=X-cisco-media:umoh
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendonly
*Mar 5 14:19:58.262: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK358582
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3808 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 99
c=IN IP4 10.249.13.130
a=sendonly
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
SIP-GW#
SIP-GW#sh voip rtp connections
VoIP RTP active connections :
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 716 717 19314 4000 192.168.1.10 192.168.1.241
2 717 716 19234 54932 10.249.13.130 10.224.42.72
Found 2 active RTP connections -
Hi
We are migrating from Analogue to IP Telephony. I have recieved the following guidlines to configure the SIP Trunk:
*For signaling: use IP : x.x.211.70 ( SIP ) on PORT 5060
*Regarding Numbering Format, use the following:
• For outgoing Calls :
The originating Number (A#), should be 96611510XXXX format.
The Destination Number should be 0NXXXXXX (N area code) or 00XXXXXXXXX (for international)
• For incoming Calls:
The Destination Number (B#), should 011510XXXX Format.
The originating Number (A#), will be 0NXXXXXXX or 00XXXXXXXXXXX Format
*Use Audio Codec's G711-aLaw ; G711-uLaw & G729
*Use T.38 For FAX
*set DTMF to RFC2833
*Make sure to reply with 200Ok for our OPTIONS messages ( ping messages for the SIP)
* configure the following SIP Timers: “Min-SE=1800 “’ & “Expires=300”
For connectivity consider the following:
SIP CE: 10.65.13.110 (it might be needed to translate this IP to the PBX local IP).
SIP GW: 10.65.13.109
Subnet mask: /30
SIP VLAN: 1191
Notes:
Kindly make sure to have GO SIP GW (x.x.211.70) routed to SIP GW (10.65.13.109) as next-hop.
Kindly make sure to have SIP CE IP addresses are in VLAN 1191.
Can please anyone explain what have to done?
RegardsAhmed,
Wao..Where do I start...This information is required for configuration on your CUBE..which will be your 2921 router...
Ahmed, here are some pointers I wrote a while ago..
In addition to these points, you will need to configure your cube to be able to route traffic to your ITSP using all the information given to you
1. Configure CUBE for media flow through. In this Mode CUBE acts as a true B2BUA. Advantages you get include address hiding and security becaue CUBE terminates and re-originate both signalling and Media. In this mode CUBE becomes a point of demarcation from th external world.
2. Configure CUCM to use Delayed Offer and CUBE to convert delayed offer to ealry offer...This prevents the need for you to use MTP to send Early offer on CUCM
voice service voip
early-offer forced
3. Configure DTMF signalling method on sip trunk to "No preference" This setting allows Unified CM to make an optimal decision for DTMF and to minimize MTP allocation.
4. Configure your CUBE to meet the requirements of your ITSP. Ask if they have configuration templates or any specific configuration they like you to use. This will save you time troubleshooting. Most of them dont use the default port 5060 because of security, confirm with your proivider what ports they use.
voice service voip
allow-connections sip to sip
sip
early-offer forced
header-passing
error-passthru
5. Use SIP to SIP...Use end to end sip. CUCM---sip---CUBE--sip----ITSP
6. Create a Trusted list of IP addresses on your CUBE is your CUBE IOS is 15.1 .2(T) and above.
voice service voip
ip address trusted list
ipv4 203.0.113.100 255.255.255.255
ipv4 192.0.2.0 255.255.255.0
This is imprtant because sometimes your ITSP will send you a single ip address for signalling and will then send media on a different IP adress. So get all the IP address your ITSP is using and add them to the trust list as shown above
7. Configure your inbound and outbound dial-peer approriately
Inbound Dial-Peer for calls from CUCM to CUBE (CUCM sending 9 +all digits dialled to CUBE)
dial-peer voice 100 voip
description *** Inbound LAN side dial-peer ***
incoming called-number 9T
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
Outbound Dial-Peer for calls from CUBE to CUCM (SP will be sending 10 digits inbound)
dial-peer voice 200 voip
description *** Outbound LAN side dial-peer ***
destination-pattern [2-9].........
session protocol sipv2
session target ipv4:
codec g711ulaw
dtmf-relay rtp-nte
Note: If more than 1 CUCM cluster exists, you will have to create multiple such LAN dial-peers with “preference CLI” for CUCM redundancy/load balancing
Inbound Dial-Peer for calls from SP to CUBE
dial-peer voice 100 voip
description *** Inbound WAN side dial-peer ***------------------(catch-all for all inbound PSTN calls)
incoming called-number [2-9].........
session protocol sipv2
codec g711ulaw
dtmf-relay rtp-nte
Outbound Dial-Peer for calls from CUBE to SP
dial-peer voice 200 voip
description *** Outbound WAN side dial-peer ***
translation-profile outgoing Digitstrip
destination-pattern 9[2-9].........
session protocol sipv2
voice-class sip bind control source gig0/1
voice-class sip bind media source gig0/1
session target ipv4::XXXX (where XXXX is the port number your provider is using if different from 5060)
codec g711ulaw
dtmf-relay rtp-nte
8. SIP Normalization:
You may need to configure sip normnalization to modify sip headers, CLI etc. A good example is during call forwarding. You may need to change your diversion headers to match the CLI your provider is expecting.. if your call forwarding is failing during testing this may be the reason..We can help you with this.
9. Media Resources
Plan your solution properly. Consider if you will need Xcoders, MTP, Conference bridge etc. You may avoid the need for xcoders if you confure your regions properly and use voice class codecs on your sip profiles. It is important to know if there are any endpoints in your network that do not support dtmf relay rtp-nte. You can avoid the use of MTP if you configure your dial-peers to have multiple dtmf types for thos phones that do not support rtp-nte
e.g
dial-peer voice 1 voip
session protocol sipv2
dtmf-relay rtp-nte digit-drop sip-kpml (if your phones support kpml..then this will be used)
If in your environment you will need to do xcoding or CFB then ensure you have PVDMS
.10.FAX
If you have FAX in your network, determine what fax protocol your sip provider supports. Dont assume. Ask them and confirm in writing what they support. I have seen legal cases because of fax failures over sip trunks
Configure your FAX devices in seperate device pools and use porefix to route calls using G711 only. Even if you are using T38, ensure your fax use G711 to establish the voice calls
Finally
11. Have a detailed and carefully planned TEST Plan. Test the FF:
Inbound and outbound Local, Long distance, International calls for G711 & G729 codecs (if supported by provider)
Outbound calls to information and emergency services
Caller ID and Calling Name Presentation
Supplementary services like Call Hold, Resume, Call Forward & Transfer
DTMF Tests
Fax calls – T.38, modem pass-through--whichever one you decide to use
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared" -
SIP Trunk not accepting inbound calls
I have a CME setup using Engin as a SIP provider
I am able to dial out with no issue, however my inbound calls do not work, they divert to the Engin voicemail
My SIP registration is OK and the number is configured as the primary DN on one of my phones
Router#sh sip-ua register status
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
038682XXXX -1 1124 yes
101 20001 45 no
102 20003 18 no
103 20005 45 no
104 20006 45 no
I do see the call come in if I debug the dial peer, but it only seems to match an outgoing dp
I am seeing a couple of disconnect cause codes that I cant seem to find any relavent information on in the CCSIP debugs
Router#
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:2, method:102, resp_code:0, container:4F947560
Sep 8 18:15:15.009: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLPrintTDContainer: Peer-Event: E_STSL_PASS_ST_PARAMS, SE Value:1800, SE Refresher:uas, Min-SE Value:1800, flags:2001
Sep 8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
Sep 8 18:15:15.017: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:100, container:4F947DF8
Sep 8 18:15:15.025: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT
Sep 8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
Sep 8 18:15:15.025: //156/83D9548A803F/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:403, container:4F947B38
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4C39C570
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 0417XXXXXX
Called Number : 038682XXXX
Source IP Address (Sig ): 211.30.48.136
Destn SIP Req Addr:Port : 203.161.164.69:5060
Destn SIP Resp Addr:Port : 203.161.164.69:5060
Destination Name : 203.161.164.69
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 211.30.48.136
Source IP Port (Media): 17768
Destn IP Address (Media): 203.161.164.69
Destn IP Port (Media): 18314
Orig Destn IP Address:Port (Media): [ - ]:0
Sep 8 18:15:15.061: //156/83D9548A803F/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 21
Disconnect Cause (SIP) : 403
Any Ideas
DougHi Tapan,
Firstly the topology is as follows
ISP/VOIP provider - Internet - Cable modem - 2800 CME router - IP Phone
The VM is provided by the ISP
debug ccsip messages
Sep 9 10:21:41.488: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Contact:
Supported: 100rel,timer
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: multipart/mixed,application/media_control+xml,application/sdp
Min-SE: 60
Session-Expires: 1800;refresher=uas
Max-Forwards: 9
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 317
v=0
o=BroadWorks 18275729 1 IN IP4 203.161.164.69
s=-
c=IN IP4 203.161.164.69
t=0 0
m=audio 18128 RTP/AVP 18 8 0 101
c=IN IP4 203.161.164.69
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=bsoft: 1 image udptl t38
Sep 9 10:21:41.508: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>
Date: Fri, 09 Sep 2011 00:21:41 GMT
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Sep 9 10:21:41.516: //4375/867C64EB8067/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>;tag=39373A4-586
Date: Fri, 09 Sep 2011 00:21:41 GMT
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
CSeq: 633854439 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=21
Content-Length: 0
Sep 9 10:21:41.544: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 203.161.164.69:5060;branch=z9hG4bKv0hlbo2030i1c85sm7b0.1
CSeq: 633854439 ACK
From: "0417XXXXXX"[email protected];user=phone>;tag=SD1uttd01-353510938-1315527701453-
To: "Doug Goding"[email protected]>;tag=39373A4-586
Call-ID: SD1uttd01-c27fdbf3f1de4ad1f0f2e1342b210494-au418e3
Max-Forwards: 9
Content-Length: 0
Voice Config
Router#
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server expires max 3600 min 3600
localhost dns:mel.byo.engin.com.au
no call service stop
voice class codec 1
codec preference 1 g711ulaw
voice translation-rule 10
rule 1 /^0/ //
voice translation-rule 11
rule 1 /^.*/ /0386821234/
voice translation-profile PSTN_Outgoing
translate calling 11
voice-card 0
dsp services dspfarm
mgcp profile default
sccp local Vlan100
sccp ccm 10.1.100.1 identifier 1 version 7.0
sccp
sccp ccm group 1
bind interface Vlan100
associate ccm 1 priority 1
associate profile 1 register confdsp
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 4
associate application SCCP
dial-peer voice 99 voip
translation-profile outgoing PSTN_Outgoing
destination-pattern .T
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 100 voip
session protocol sipv2
session target dns:mel.byo.engin.com.au
incoming called-number 0386821234
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 110 voip
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
session protocol sipv2
session target dns:mel.byo.engin.com.au
incoming called-number .%
dtmf-relay rtp-nte
no vad
dial-peer voice 90 voip
description Melbourne 03 Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern [89].......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 91 voip
description National Numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 0[278]........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 92 voip
description Vic/Tas 03 numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern [56].......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 93 voip
description Mobile numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 04........
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 94 voip
description 13XXXX numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 13[1-9]...
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 96 voip
description 1300/1800 numbers
translation-profile outgoing PSTN_Outgoing
destination-pattern 1[38]00......
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 98 voip
description Emergency 000
translation-profile outgoing PSTN_Outgoing
destination-pattern 000
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
sip-ua
credentials username 0386821234 password 7 XXXX realm voice.mibroadband.com.au
authentication username 0386821234 password 7 XXXX
nat symmetric role active
nat symmetric check-media-src
no remote-party-id
retry invite 2
retry register 10
timers connect 100
mwi-server dns:mel.byo.engin.com.au expires 3600 port 5060 transport udp unsolicited
registrar dns:mel.byo.engin.com.au expires 3600
sip-server dns:mel.byo.engin.com.au
connection-reuse
telephony-service
sdspfarm conference mute-on #1 mute-off #2
sdspfarm units 2
sdspfarm tag 1 confdsp
conference hardware
max-ephones 42
max-dn 144
ip source-address 10.1.100.1 port 2000
calling-number initiator
service phone videoCapability 1
service phone displayOnDuration 00:01
service phone displayOnTime 08:30
service phone displayOffTime 17:30
service phone displayIdleTimeout 00:01
service phone displayOnWhenIncomingCall 1
system message Cisco CME
load 7941 SCCP41.8-4-2S
load 7942 SCCP42.8-4-2S
load 7945 SCCP45.8-4-2S
load 7961 SCCP41.8-4-2S
load 7962 SCCP42.8-4-2S
load 7965 SCCP45.8-4-2S
load ata ATA030204SCCP090202A
time-zone 48
date-format dd-mm-yy
voicemail 90125200
mwi relay
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
moh music-on-hold.au
web admin system name cisco secret 5 $1$d8/H$glhLiCCWXmFSUp6BtwGho0
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 0.T
create cnf-files version-stamp 7960 Jul 06 2011 10:32:45
ephone-dn 1 dual-line
number 038682XXXX
label 101
name 7965
mwi sip
ephone-dn 2 dual-line
number 102
label 102
name 7941
ephone-dn 3 dual-line
number 103
label 103
name 7920
ephone 1
device-security-mode none
video
mac-address 0023.5EB8.6E4E
type 7965
button 1:2 2:1
ephone 3
device-security-mode none
mac-address 0019.0633.A933
max-calls-per-button 2
type 7920
button 1:3
ephone 10
device-security-mode none
mac-address 0019.E7B7.BAB3
max-calls-per-button 2
type ata
button 1:1
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