SIP & H323
Dos anyone have a configuration example for running SIP and H323 on the same gateway?
I am trying to use Broadvoice and route calls to another router with FXS ports on it via H323.
Thanks in advance.
Steve
[partial config]
voice class codec 1
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g711alaw
codec preference 4 g711ulaw
voice class codec 2
codec preference 1 g729r8
dial-peer voice 1001 voip
description description -SIP- VOXBONE
application prepaid
voice-class codec 2
session protocol sipv2
incoming called-number .
dtmf-relay rtp-nte
dial-peer voice 2000 voip
destination-pattern 001.T
progress_ind setup enable 3
progress_ind progress enable 2
progress_ind connect enable 1
translate-outgoing called 100
session target ipv4:1.1.1.10
dtmf-relay h245-signal h245-alphanumeric
dial-peer voice 2001 voip
destination-pattern .T
progress_ind setup enable 3
progress_ind progress enable 2
progress_ind connect enable 1
translate-outgoing called 100
session target ipv4:1.1.1.1
dtmf-relay h245-signal h245-alphanumeric
dial-peer voice 2002 voip
destination-pattern 0057.T
progress_ind setup enable 3
progress_ind progress enable 2
progress_ind connect enable 1
translate-outgoing called 100
session target ipv4:1.1.1.2
dtmf-relay h245-signal h245-alphanumeric
dial-peer voice 2003 voip
destination-pattern 0056.T
progress_ind setup enable 3
progress_ind progress enable 2
progress_ind connect enable 1
translate-outgoing called 102
session target ipv4:1.1.1.3
dtmf-relay h245-signal h245-alphanumeric
sip-ua
credentials username 5512041842 password 107758300B4F405A05572722090A671A2D rea
lm SIP_Auth_Realm
registrar ipv4:1.1.1.1 expires 60
If you need specific info let me know
Similar Messages
-
Reject sip/h323 calls by IP?
i have a few sip/h323 providers. I have also enabled sip/h323 on my as5400xm(this is for my asterisk server). Since i'm using these providers, i have to put their IP in my access-list. my concern is, since my gateway is accepting sip/h323 calls. what if these provider send the calls to my gateway? so i was thinking of a way to restrict this. It could be as simple as tweaking the access-list. but I don't know. Please help.
here's how i have my access-list setup:
access-list 101 permit tcp host 10.10.10.10 any
access-list 101 permit udp host 10.10.10.10 any
access-list 101 permit udp any any range 16384 32767
access-list 101 deny tcp any any
access-list 101 deny udp any any
Thanks in advanceAh, so you just want to restrict VoIP calls from L3 addresses other than your provider?
That's just a simple ACL to open up traffic to your SIP ITSP's IP external addresses, and block anything else.
You can get what IPs and ports are used by your provider, but here is what you need open on the Cisco side inbound for an inbound ACL on a WAN interface:
UDP - ITSP address:ITSP SIP Port to External interface:5060 - For SIP signaling
ITSP address:ITSP RTP Port Range - External interface:16384-32767 - RTP traffic
ITSP's port range could be anything between 1024-65535. SIP usually comes from UDP/5060 from the ITSP, but doesn't have to. Verify with them, or look at a SIP debug or packet capture to verify.
The implicit deny will take care of everything else. -
SIP- h323 in a AS5850 - Not able to send h323 calls coming from a SIP Phone
Dear All!
I have an AS5850 configured as a SIP Gateway and as a H323 Gateway. I'm planning to use this equipment as an interconnection point between PSTN,SIP and H323.
I already have a functional H323 Network with ISDN trunks to the pstn and it is working fine. I added SIP configuration to the AS5850 in order to be able to route calls out to the PSTN or H323 remote ends coming from a SIP Phone registered with a third-party SIP Proxy.
When the calls coming from the SIP Phone goes to a PSTN destination the calls completes properly, but i am having problems trying to send calls coming from the SIP phone to a remote h323 gateway(also cisco)
Attached is my configuration and the error i'm getting in my cdr. It seems that the "ext" number of the phone is being used as destination string in the last call leg, but i'm not sure.
Please Help!
dial-peer voice 100 pots
application session
destination-pattern 5T
port 2/6:D
forward-digits all
dial-peer voice 102 pots
application session
destination-pattern 044T
port 2/6:D
forward-digits all
dial-peer voice 103 voip
application session
incoming called-number 001T
destination-pattern 001T
session protocol sipv2
session target ipv4:20X.21X.17X.1X
tech-prefix 10511
sip-ua
sip-server ipv4:20X.6X.14X.18X
CDR ERROR:
.Mar 24 2004 18:31:42.620 GMT: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2, ConnectionId 9F74CE17 7D2A11D8 82A09B41 D2C3D418, SetupTime .18:31:42.470 GMT Wed Mar 24 2004, ***PeerAddress 2006***, PeerSubAddress , DisconnectCause 3 , DisconnectText no route to destination (3), ConnectTime .18:31:42.620 GMT Wed Mar 24 2004, DisconnectTime .18:31:42.620 GMT Wed Mar 24 2004, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 0, TransmitBytes 0, ReceivePackets 0, ReceiveBytes 0
Thanks.
Attached you can find the debug ccsip messages output.There are 2 solutions here.
1. Use of SIP/H.323 Signalling Gateway as the protocol convertor. Search google will yield heaps of hits on this subject. Product available both commercial and open source, trial, etc. Using this method means that the SIP End Point will communicate with H.323 End Point without going out the PSTN. I believe this is what you want to achieve in the long term. You are trying the AS5xxx as the protocol convertor for you, which it will not work. A call flow will be something like SIP IP Phone->SIP Server->SIP-to-H.323 Gateway->H.323 Gatekeeper->H.323 End Point. Of couse there is a SIP server that do the protocol convertor in the same box but the functionality is the still the same. Performance and concurrent call setup differ from products to products. Going for this solution would require you to find such products and test it on the your network.
2. If you do not wish to try on Soluton 1, this solution is a workaround way by not getting device but using the existing equipment that you have right now. Onto whether this good long term solution for depends on what you want to achieve both in term of commercially and technically. A call flow will be SIP End Point->SIP Server->Voice Gateway (AS5xxx)->PSTN Switch(ISDN/PRI)->Voice Gateway->H.323 Gatekeeper>H.323 End Point. The key is the Voice session must traverse the ISDN link. In other words your dial pattern must be setup is such as way that will go out thru the dial peer pots to pstn switch then come back to another dial-peer pots. I am not saying this is the most efficient way of doing it, I merely suggesting a workable way to achieve your desired goal without soluton 1.
Hopes you get better understanding now.
Thanks
SSng -
SIP -H323 Simultaneously on Cisco AS5350
A Chairde,
I am looking to setup my AS5350 to have H323 on one E1 , and SIP running on the other E1 for testing purposes.
Am having problems with DIAL PEER Commands SESSION PROTOCOL and TRANSPORT , they are not available, also most commands in VOICE CLASS VOIP do not load either.
I have new IOS for AS5350 (12.4) , but was wondering if my idea will work.
Attached is AS5350 script.Cisco gateways provide support for coexistence of SIP and H.323 calls beginning with Cisco IOS Software Release 12.2(2)XB.
http://www.cisco.com/en/US/products/sw/iosswrel/ps5012/prod_release_note09186a0080080a3a.html -
Dear All,
I have a few questions and I would appereciate if someone will answer my questions?
1) May I know pros and cons of using Dual NIC and single NIC with VCSE in DMZ?
2) In order to make H323 and SIP encrypted call, what configuration need to be done on Cisco endpoint, non-cisco endpoint, VCSC and VCSE (both signalling and Media need to be encrypted).
3)let say my VCSE is in DMZ- endpoint A (cisco) and endpoint B (non-cisco) are registered on VCSE. I would like VCSE to use non-traversal call license when A call to B or B call to A regardless of whether H323 or SIP call.How can I force VCSE and endpoints to use non-traversal call. I only want VCSE to handle signalling and media is EP to EP direct. As traversal license is quite expensive.
I found following information from Cisco document.
all Cisco TelePresence endpoints are traversal enabled and so a traversal license will always be needed when at least one of the endpoints involved in the call is a Cisco TelePresence endpoint.
Is it possible to disable traversal client feature of Cisco endpoint?
Thanks and Best Regards,voice register pool 4
add-->codec g711ulaw
voice register global
no create pro
create prof
and enable below debugs and send the logs
deb ccsip mess
deb voip ccapi inou
deb h225 asn1
deb h245 asn1 -
Can someone explain how h323 to SIP calls work & vice versa.
The following messages are mapped:
SIP <---> H323
INVITE - SETUP
100 Trying - Call Proc
180 Ringing - Alerting
183 Session Progress - Progress
200 OK (for INVITE) - Connect
BYE - Release Complete
With H323 to SIP CUBE, if fast start occurs on one leg, early offer needs to happen on the other (and vice versa). Most SIP devices these days to early offer (SDP in invite) so you typically need fast start enabled on both directions of the H323 leg for this design.
Check out this link for more information:
http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-h323sip_ps5640_TSD_Products_Configuration_Guide_Chapter.html -
How can i transfer a call from SIP 9971 to PBX system on CME router
hello everybody,
I have a critical problem about interaction of transfering feature between CME router and pbx panasonic system in some status. let me explain more detail about this issue..i have a SIP 9971(CP-9971) registered on CME at the one site and a voice gateway that is connect with PBX system through a E1 pri trunk connection at the other site. totally the integration between CME and PBX is ok and there is no problem in two direction, i mean i can call pbx system from cp-9971 and vise versa but when i call from a phone which is registered on PBX site to SIP 9971 which is registered on cisco CME call is connected,then when i try to transfer that call to another phone at PBX site, the session is open between two panasonic phones but no audio transmited in two direction. in addition every thing works fine about SCCP phones(transfer feature works fine). here is my configuration file. i hope someone could help me because i've searched a lot but no result help help help plz....
cme router 3845 configuration
VOIP-3845#show running-config
Building configuration...
Current configuration : 12657 bytes
! Last configuration change at 11:44:01 UTC Mon Oct 31 2011 by admin
! NVRAM config last updated at 11:44:02 UTC Mon Oct 31 2011 by admin
! NVRAM config last updated at 11:44:02 UTC Mon Oct 31 2011 by admin
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname VOIP-3845
boot-start-marker
boot-end-marker
no aaa new-model
clock calendar-valid
dot11 syslog
ip source-route
ip cef
no ipv6 cef
multilink bundle-name authenticated
voice-card 0
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
sip
bind control source-interface Loopback10
bind media source-interface Loopback10
registrar server
voice register global
mode cme
source-address 192.168.2.1 port 5060
max-dn 720
max-pool 262
load 9971 sip9971.9-1-1SR1.loads
authenticate register
authenticate realm cisco.com
tftp-path flash:
file text
create profile sync 0063544528862458
camera
video
voice register dn 1
number 500
voice register dn 2
number 600
voice register dn 3
number 700
name test
voice register template 1
softkeys idle Newcall Redial Cfwdall
softkeys connected Confrn Endcall Hold Trnsfer
voice register pool 1
id mac B8BE.BF23.5242
type 9971
number 1 dn 1
template 1
username test password test
camera
video
blf-speed-dial 4 600 label "test"
voice register pool 2
id mac B8BE.BF9C.5476
type 9971
number 1 dn 2
template 1
username bank password bank
camera
video
voice register pool 3
id mac B8BE.BF9C.51D4
type 9971
number 1 dn 3
template 1
username test1 password test1
camera
video
voice register pool 4
id mac B8BE.BF9C.4FA2
number 1 dn 1
camera
video
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-1576175886
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1576175886
revocation-check none
rsakeypair TP-self-signed-1576175886
crypto pki certificate chain TP-self-signed-1576175886
certificate self-signed 01
30820241 308201AA A0030201 02020101 300D0609 2A864886 F70D0101 04050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 31353736 31373538 3836301E 170D3131 31303038 30393034
34365A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D31 35373631
37353838 3630819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
8100D6EC 47BCDC3C 82F43FF3 23522678 2616868D 9910DCD2 E36016B3 D7B40DA7
53A6E339 4978D451 21F051BE B21F8AD5 86B952DC 1ECCE371 3E094B54 26A41E14
A3055C06 AE860756 425E5C50 E62B3287 631B1E87 9BAC2E39 2810E120 DA3BF823
947EA591 81CA5489 1B868239 E835EC7C 0AA7651A 22D6E47F 545EBEF3 A172C9A3
5A0D0203 010001A3 69306730 0F060355 1D130101 FF040530 030101FF 30140603
551D1104 0D300B82 09564F49 502D3338 3435301F 0603551D 23041830 1680146C
934AD072 99DDC600 ECD6F389 8F71E0C2 18EC2E30 1D060355 1D0E0416 04146C93
4AD07299 DDC600EC D6F3898F 71E0C218 EC2E300D 06092A86 4886F70D 01010405
00038181 000E82F6 5FBB847C 49226955 6F7DECE7 0B093513 D57C35D5 4CD22FA7
8144A080 B0D56C8D 86AF8156 0152443A A3FBE59F B1AEFFBC BEB43E09 35757BAD
4C06FC4A 0F3695E0 B00FBD30 4E8F36CE 7748F39C F9602650 7A1D2D48 DBC31237
AE3D63CE 593D31F5 62E4916F D20E30E8 30DC55C0 120FBD26 D2768DBC A67DDC34
5BDB66B1 E3
quit
license udi pid CISCO3845-MB sn FOC14421Q1Y
archive
log config
hidekeys
username admin privilege 15 secret 5 $1$Zf7j$P93opukmmEBIioVpjmHB3.
redundancy
interface Loopback10
ip address 192.168.2.1 255.255.255.0
interface Tunnel1
ip address 172.25.10.1 255.255.255.0
no ip redirects
ip nhrp map multicast dynamic
ip nhrp network-id 10
tunnel source GigabitEthernet0/1.1
tunnel mode gre multipoint
tunnel key 100
interface Tunnel2
ip address 172.25.11.1 255.255.255.0
no ip redirects
ip nhrp map multicast dynamic
ip nhrp network-id 20
tunnel source GigabitEthernet0/1.2
tunnel mode gre multipoint
interface Tunnel14
ip address 192.168.13.129 255.255.255.252
tunnel source GigabitEthernet0/1.1
tunnel destination 10.2.68.25
interface Tunnel18
ip address 192.168.13.137 255.255.255.252
tunnel source GigabitEthernet0/1.1
tunnel destination 10.9.160.236
interface GigabitEthernet0/0
no ip address
shutdown
duplex auto
speed auto
media-type rj45
interface GigabitEthernet0/1
no ip address
duplex auto
speed auto
media-type rj45
interface GigabitEthernet0/1.1
encapsulation dot1Q 10
ip address 10.9.160.25 255.255.255.0
interface GigabitEthernet0/1.2
encapsulation dot1Q 50
ip address 10.10.9.25 255.255.255.0
router eigrp 202
network 172.25.11.0 0.0.0.255
network 192.168.2.0 0.0.0.15
redistribute static route-map MYMAP1
router eigrp 201
network 172.25.10.0 0.0.0.255
network 192.168.2.0 0.0.0.15
redistribute static route-map MYMAP1
ip forward-protocol nd
ip http server
ip http secure-server
ip http path flash:/gui
ip route 10.2.68.0 255.255.255.0 10.9.160.1
ip route 10.10.0.0 255.255.0.0 10.10.9.1
ip route 10.64.164.30 255.255.255.255 10.9.160.1
ip route 192.168.14.0 255.255.255.0 192.168.13.130
ip route 192.168.17.0 255.255.255.0 Tunnel18
ip access-list standard REDIS1
permit 192.168.14.0
permit 192.168.17.0
route-map MYMAP1 permit 10
match ip address REDIS1
snmp-server community test RO
tftp-server flash:term11.default.loads
tftp-server flash:dkern9971.100609R2-9-0-3.sebn
tftp-server flash:kern9971.9-0-3.sebn
tftp-server flash:rootfs9971.9-0-3.sebn
tftp-server flash:sboot9971.111909R1-9-0-3.sebn
tftp-server flash:sip9971.9-0-3.loads
tftp-server flash:skern9971.022809R2-9-0-3.sebn
tftp-server flash:sccp11.9-0-2sr1s
tftp-server flash:SCCP11.9-1-1SR1S.loads
tftp-server flash:apps11.9-1-1TH1-16.sbn
tftp-server flash:cnu11.9-1-1TH1-16.sbn
tftp-server flash:cvm11sccp.9-1-1TH1-16.sbn
tftp-server flash:dsp11.9-1-1TH1-16.sbn
tftp-server flash:jar11sccp.9-1-1TH1-16.sbn
tftp-server flash:term06.default.loads
tftp-server flash:sip9971.9-1-1SR1.loads
tftp-server system:cme/sipphone
tftp-server flash:Desktops/320x212x12/NantucketFlowers.png
tftp-server flash:Desktops/320x212x12/TN-CampusNight.png
tftp-server flash:Desktops/320x212x12/TN-CiscoFountain.png
tftp-server flash:Desktops/320x212x12/TN-Fountain.png
tftp-server flash:Desktops/320x212x12/TN-MorroRock.png
tftp-server flash:Desktops/320x212x12/TN-NantucketFlowers.png
tftp-server flash:Desktops/320x212x12/Fountain.png
tftp-server flash:Desktops/320x212x12/CiscoLogo.png
tftp-server flash:Desktops/320x212x12/TN-CiscoLogo.png
tftp-server flash:Desktops/320x212x12/List.xml
tftp-server flash:Desktops/320x216x16/List.xml
tftp-server flash:Desktops/320x212x16/List.xml
tftp-server flash:gui/admin_user.html
tftp-server flash:gui/admin_user.js
tftp-server flash:gui/CiscoLogo.gif
tftp-server flash:gui/Delete.gif
tftp-server flash:gui/dom.js
tftp-server flash:gui/downarrow.gif
tftp-server flash:gui/ephone_admin.html
tftp-server flash:gui/logohome.gif
tftp-server flash:gui/normal_user.html
tftp-server flash:gui/normal_user.js
tftp-server flash:gui/Plus.gif
tftp-server flash:gui/sxiconad.gif
tftp-server flash:gui/Tab.gif
tftp-server flash:gui/telephony_service.html
tftp-server flash:gui/uparrow.gif
tftp-server flash:gui/xml-test.html
tftp-server flash:gui/xml.template
tftp-server flash:ringtones/Analog1.raw
tftp-server flash:ringtones/Analog2.raw
tftp-server flash:ringtones/AreYouThere.raw
tftp-server flash:ringtones/AreYouThereF.raw
tftp-server flash:ringtones/Bass.raw
tftp-server flash:ringtones/CallBack.raw
tftp-server flash:ringtones/Chime.raw
tftp-server flash:ringtones/Classic1.raw
tftp-server flash:ringtones/Classic2.raw
tftp-server flash:ringtones/ClockShop.raw
tftp-server flash:ringtones/DistinctiveRingList.xml
tftp-server flash:ringtones/Drums1.raw
tftp-server flash:ringtones/Drums2.raw
tftp-server flash:ringtones/FilmScore.raw
tftp-server flash:ringtones/HarpSynth.raw
tftp-server flash:ringtones/Jamaica.raw
tftp-server flash:ringtones/KotoEffect.raw
tftp-server flash:ringtones/MusicBox.raw
tftp-server flash:ringtones/Piano1.raw
tftp-server flash:ringtones/Piano2.raw
tftp-server flash:ringtones/Pop.raw
tftp-server flash:ringtones/Pulse1.raw
tftp-server flash:ringtones/Ring1.raw
tftp-server flash:ringtones/Ring2.raw
tftp-server flash:ringtones/Ring3.raw
tftp-server flash:ringtones/Ring4.raw
tftp-server flash:ringtones/Ring5.raw
tftp-server flash:ringtones/Ring6.raw
tftp-server flash:ringtones/Ring7.raw
tftp-server flash:ringtones/RingList.xml
tftp-server flash:ringtones/Sax1.raw
tftp-server flash:ringtones/Sax2.raw
tftp-server flash:ringtones/Vibe.raw
tftp-server flash:APPS-1.2.1.SBN
tftp-server flash:SYS-1.2.1.SBN
tftp-server flash:GUI-1.2.1.SBN
tftp-server flash:CP7921G-1.2.1.LOADS
tftp-server flash:TNUX-1.2.1.SBN
tftp-server flash:TNUXR-1.2.1.SBN
tftp-server flash:WLAN-1.2.1.SBN
tftp-server flash:apps37sccp.1-2-1-0.bin
tftp-server flash:APPSH-1.3.1.SBN
tftp-server flash:GUIH-1.3.1.SBN
tftp-server flash:CP7925G-1.3.1.LOADS
tftp-server flash:SYSH-1.3.1.SBN
tftp-server flash:TNUXH-1.3.1.SBN
tftp-server flash:WLANH-1.3.1.SBN
tftp-server flash:SCCP11.9-2-1S.loads
tftp-server flash:Desktops/320x212x12/CampusNight.png
tftp-server flash:Desktops/320x212x12/CiscoFountain.png
tftp-server flash:Desktops/320x212x12/MorroRock.png
tftp-server flash:skern9971.022809R2-9-2-1.sebn
tftp-server flash:sip9971.9-2-1.loads
tftp-server flash:sboot9971.031610R1-9-2-1.sebn
tftp-server flash:rootfs9971.9-2-1.sebn
tftp-server flash:dkern9971.100609R2-9-2-1.sebn
tftp-server flash:kern9971.9-2-1.sebn
tftp-server flash:United_States/g4-tones.xml
tftp-server flash:English_United_States/gd-sip.jar
tftp-server flash:sboot9971.031610R1-9-1-1SR1.sebn alias sboot9971.031610R1-9-1-1SR1.sebn
tftp-server flash:rootfs9971.9-1-1SR1.sebn alias rootfs9971.9-1-1SR1.sebn
tftp-server flash:kern9971.9-1-1SR1.sebn alias kern9971.9-1-1SR1.sebn
tftp-server flash:dkern9971.100609R2-9-1-1SR1.sebn alias dkern9971.100609R2-9-1-1SR1.sebn
tftp-server flash:skern9971.022809R2-9-1-1SR1.sebn alias skern9971.022809R2-9-1-1SR1.sebn
control-plane
mgcp profile default
dial-peer voice 1 voip
description connection-trough-PBX
destination-pattern 0....
session target ipv4:192.168.13.130
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 100 voip
description K
destination-pattern 9T
session target ipv4:192.168.13.130
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 5 voip
shutdown
destination-pattern *3709
session protocol sipv2
session target ipv4:192.168.13.130
session transport tcp
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 2 pots
incoming called-number .
dial-peer voice 10 voip
gatekeeper
shutdown
telephony-service
em logout 0:0 0:0 0:0
max-ephones 262
max-dn 400
ip source-address 192.168.2.1 port 2000
load 7911 SCCP11.9-2-1S
max-conferences 12 gain -6
web admin system name admin secret 5 $1$IKnn$tyKyuBcGqXFl6nhxCSu.z0
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
create cnf-files version-stamp 7960 Oct 29 2011 12:39:25
ephone-template 1
softkeys connected Confrn Endcall Trnsfer Hold
keep-conference endcall
ephone-dn 1 dual-line
number 200
label test
name test
ephone-dn 2 dual-line
number 300
label Sepahbod
name Sepahbod
ephone-dn 4 dual-line
number 666
ephone-dn 5 dual-line
number 660
ephone-dn 6 dual-line
number 670
ephone-dn 7 dual-line
number 770
ephone-dn 8 dual-line
number 770
ephone-dn 9 dual-line
number 999
ephone 1
device-security-mode none
mac-address 18EF.639F.BCB0
keep-conference endcall
button 1:1
ephone 2
device-security-mode none
mac-address 0025.8418.B017
ephone-template 1
keep-conference endcall
button 1:2
ephone 3
device-security-mode none
mac-address F04D.A243.3154
keep-conference endcall
button 1:4
ephone 4
device-security-mode none
mac-address 6CF0.496A.69E9
button 1:4
ephone 5
device-security-mode none
mac-address 0015.E987.345F
keep-conference endcall
button 1:5
ephone 6
device-security-mode none
mac-address 0024.1DEA.614A
keep-conference endcall
button 1:6
ephone 9
device-security-mode none
mac-address 001D.7D4D.4DCB
button 1:9
line con 0
line aux 0
line vty 0 4
login local
transport input telnet
scheduler allocate 20000 1000
end
and Voice Gateway connected two PBX system configuration
Current configuration : 3486 bytes
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Voice-GW
boot-start-marker
boot-end-marker
card type e1 0 2
no aaa new-model
network-clock-participate wic 2
dot11 syslog
ip source-route
ip cef
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-net5
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2811 sn FHK1352F0E9
username admin privilege 15 secret 5 $1$O6AN$1kvvqiLdIl3/ZTHoyYRy0/
redundancy
controller E1 0/2/0
framing NO-CRC4
pri-group timeslots 1-31
controller E1 0/2/1
interface Tunnel14
ip address 192.168.13.130 255.255.255.252
tunnel source FastEthernet0/1
tunnel destination 10.9.160.25
interface Tunnel17
ip address 192.168.13.134 255.255.255.252
tunnel source FastEthernet0/1
tunnel destination 10.9.160.25
interface FastEthernet0/0
ip address 192.168.14.252 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
ip address 10.2.68.25 255.255.255.0
duplex auto
speed auto
interface Serial0/2/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn overlap-receiving
isdn incoming-voice voice
no cdp enable
router eigrp 201
network 172.25.10.0 0.0.0.255
network 192.168.14.0
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 10.9.160.0 255.255.255.0 10.2.68.1
ip route 10.128.0.69 255.255.255.255 Tunnel14
ip route 192.168.2.1 255.255.255.255 192.168.13.129
ip route 192.168.17.0 255.255.255.0 Tunnel14
tftp-server flash:SCCP11.9-2-1S.loads
tftp-server flash:jar11sccp.9-2-1TH1-13.sbn
tftp-server flash:dsp11.9-2-1TH1-13.sbn
tftp-server flash:cvm11sccp.9-2-1TH1-13.sbn
tftp-server flash:cnu11.9-2-1TH1-13.sbn
tftp-server flash:apps11.9-2-1TH1-13.sbn
control-plane
voice-port 0/0/0
caller-id enable
voice-port 0/0/1
voice-port 0/0/2
supervisory disconnect dualtone mid-call
dial-type pulse
disc_pi_off
output attenuation 1
echo-cancel coverage 32
timeouts call-disconnect 5
timeouts wait-release 1
timing hookflash-out 50
timing sup-disconnect 50
connection plar 600
caller-id enable
voice-port 0/0/3
caller-id enable
voice-port 0/2/0:15
mgcp profile default
dial-peer voice 1 pots
description connection-to-PBX
destination-pattern 0....
direct-inward-dial
port 0/2/0:15
forward-digits 4
dial-peer voice 10 voip
destination-pattern ...
session target ipv4:192.168.13.129
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 20 pots
description FXO-K
destination-pattern 9T
progress_ind alert enable 8
progress_ind progress enable 8
progress_ind connect enable 8
direct-inward-dial
port 0/0/2
prefix 9
dial-peer voice 30 pots
description FXO-K2
destination-pattern 9T
direct-inward-dial
port 0/0/1
prefix 9
telephony-service
max-ephones 20
max-dn 100
ip source-address 192.168.14.252 port 2000
cnf-file location flash:
load 7911 term11.default.loads
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1
number 770
line con 0
line aux 0
line 1/0 1/15
line vty 0 4
login local
transport input telnet
scheduler allocate 20000 1000
endHaving looked at your spreadsheet I see you're failing H323 transfers back to your ISDN system, but only under certain circumstances. Quite why, I'm not sure, possibly because you haven't codec defined on your H323 dial peers. or it could be something else
I think you may be able to work around the problem by adding
" supplementary-service h450.12 " under voice service voip on your CME router as a quick fix.
reference
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmetrans.html#wpxref44614
worth a try
Adam -
Sip 500 Internal Server Error Reason: Q.850;cause=16
Please help in understanding what is wrong in the config .Incoming calls don't work.
show run:
voice service voip
ip address trusted list
ipv4 87.226.136.164 255.255.255.255
ipv4 172.16.24.0 255.255.255.0
ipv4 188.254.68.66 255.255.255.255
ipv4 188.254.68.67 255.255.255.255
ipv4 188.254.69.66 255.255.255.255
ipv4 188.254.69.67 255.255.255.255
ipv4 46.38.52.68 255.255.255.255
address-hiding
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
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback cisco
sip
voice class codec 1
codec preference 1 g729br8
codec preference 2 g729r8
codec preference 3 g711alaw
codec preference 4 g711ulaw
voice class codec 2
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
voice translation-rule 1
rule 1 /XXX5397962/ /1999/
voice translation-rule 2
rule 1 /XXX55317577/ /1999/
voice translation-rule 3
rule 1 /5555317884/ /1999/
voice translation-profile ROS
translate called 1
voice translation-profile ROS2
translate called 2
voice translation-profile ROS3
translate called 3
interface FastEthernet0/0
ip address 178.208.X.X 255.255.255.248
ip access-group INBOUND in
no ip unreachables
ip verify unicast reverse-path
ip nat outside
ip inspect IPFW in
ip inspect IPFW out
ip virtual-reassembly in
duplex auto
speed auto
no cdp enable
interface FastEthernet0/1
no ip address
ip nat inside
ip virtual-reassembly in
duplex auto
speed auto
interface FastEthernet0/1.1
encapsulation dot1Q 1 native
ip address 10.110.0.200 255.255.255.0
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/1.2
encapsulation dot1Q 2
ip address 172.16.24.254 255.255.255.0
ip nat inside
ip virtual-reassembly in
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.16.24.254
ip dns server
ip nat inside source list NAT interface FastEthernet0/0 overload
ip route 0.0.0.0 0.0.0.0 178.208.X.X
ip route 192.168.0.0 255.255.0.0 Null0 254
sccp local FastEthernet0/1.2
sccp ccm 172.16.24.101 identifier 1 version 7.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register XCODE123456
keepalive retries 1
keepalive timeout 10
switchover method immediate
switchback method immediate
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 6
associate application SCCP
dial-peer voice 10000 voip
tone ringback alert-no-PI
description ROSTELECOM Incoming
translation-profile incoming ROS
destination-pattern 74955397962
session protocol sipv2
session target ipv4:87.226.136.164
session transport udp
incoming called-number XXXX5397962
dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 10010 voip
tone ringback alert-no-PI
description ROSTELECOM Incoming
translation-profile incoming ROS2
destination-pattern XXX55317577
session protocol sipv2
session target ipv4:87.226.136.164
session transport udp
incoming called-number 75555317577
dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 10020 voip
tone ringback alert-no-PI
description ROSTELECOM Incoming
translation-profile incoming ROS3
preference 1
destination-pattern 5555317884
session protocol sipv2
session target ipv4:188.254.68.66
session transport udp
incoming called-number 5555317884
dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 10021 voip
tone ringback alert-no-PI
description ROSTELECOM Incoming
translation-profile incoming ROS
preference 2
destination-pattern 5555317884
session protocol sipv2
session target ipv4:188.254.69.66
session transport udp
incoming called-number 5555317884
dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 2 voip
tone ringback alert-no-PI
description to CUCM_PUB
destination-pattern 1...
session target ipv4:172.16.24.101
voice-class codec 2
dtmf-relay rtp-nte
debug ccsip all:
c2801#
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [188.254.68.66]:9290, local_address:[ - ]
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
06:19:26: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 188.254.68.66:9290;branch=z9hG4bK-6110d60075c89eab-a(STATE_IDLE, SUBSTATE_NONE)
06:19:26: //-1/xxxxxxxxxxxx/SIP/T3c000c-1
Call-ID: isbc6994325518770806443-1385214296-16204
Fransport/sipSPIUpdateResponseInfo: Dialog Transaction Address 188.254.68.66,Port 9290, Transport 1, SentBy Port 5060
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone MSK to SIP default timezone = GMT
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 188.254.68.66,Port 929rom:
<sip:[email protected];user=phone>;tag=sbc09026994325from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 188.254.68.66,Port 9290, Transport 1, SentBy Port 5060
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Converting TimeZone MSK to SIP default timezone = GMT
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 188.254.68.66,Port 518770806443
ddress_to_bind: return addr 178.208.X.Xone>
06:19:26: //-1/EE5EC9DD8170/SIP/State/sipSPIChangeState: 0x6A874E70 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 188.254.68.66,Port 9290, Transport 1, SentBy Port 5060
0
CSeq: 1 INVITE
Min-SE: 90
Session-Expires: 3600;refresher=u6:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetDateHeader: Convac
Contact: <sip:[email protected]:9290;user=phone>
A //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: rellow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO,PRACK
Supported:turn addr 178.208.X.X
06:19:26: //-1/EE5EC9DD8170/SIP/St timer,100rel
Diversion: <sip:[email protected]>;privacyate/sipSPIChangeState: 0x6A874E70 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 188.254.68.66,Port 9290, Transport 1,
Sen=off;screen=no;reason=unknown,<sip:[email protected]>;priv6:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponsacy=off;screen=no;reason=unknown
Max-Forwards: 70
User-AgenteInfo: Dialog Transaction Address 188.254.68.66,Port 9290, Tra: VCS 5.8.2.56-03
Content-Length: 393
Content-Type: applicatnsport 1, SentBy Port 9290ion/sdp
v=0
o=- 12060 26053 IN IP4 188.254.68.67
s=SBC call
c=IN IP4 188.254.68.67
t=0 0
m=audio 24402 RTP/AVP 8 0 18 98 96 97 101
a=rtpmap:98 G.729a/8000
a=rtpmap:96 G.729ab/8000
a=rtpmap:97 G.729b/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
a=ptime:10
a=X-vrzcap:vbd Ver=1 Mode=FaxPr ModemRtpRed=0
a=X-vrzcap:identification bin=DSR2866 Prot=mgcp App=MG
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAddContextToTable: Added context(0x6A874E70) with key=[52] to table
06:19:26: //-1/000000000000/SIP/Info/sipSPI_ipip_vcc_Initialization: Entry...
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipSPIUpdateResponseInfo: Dialog Transaction Address 188.254.68.66,Port 9290, Transport 1, SentBy Port 9290
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling ip_best_local_address()
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 178.208.X.X
06:19:26: //-1/EE5EC9DD8170/SIP/State/sipSPIChangeState: 0x6A874E70 : State change from (STATE_NONE, SUBSTATE_NONE) to
c2801#L
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIUaddCcbToUASReqTable: ****Adding to UAS Request table.
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x6
c2801#a
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIMatchSrcIp
c2801#mat: VIA URL:sip:188.254.68.66:9290, Host:188.254.68.66
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetShrlPeer: Try match incoming dialpeer for Calling number: : 9067259847
06:19:26:ched for incoming call
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported h
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetFromCalledPartyId: P-Called-Party-ID header not found
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetPeerByCalledPartyId: P-Called-Party-ID not found or parse error
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: No match found for P-Called-Party-ID
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: Peer tag 10020 matched for incoming call
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling reg_invoke_ip_first_hop()
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: calling ip_best_local_address()
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_sig_ip_address_to_bind: return addr 178.208.X.X
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: Precondition tag absent in Require/Supported header
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: Checking Video Type Rate=-1 video_codec_allowed=1F
06:19:26: //-1/EE5EC9DD8170/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetModemInfoPerCall: peer_callID=0
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: xcoder high-density disabled
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: Flow Mode set to FLOW_THROUGH
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIGetCallConfig: Media forking disabled
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIContinueNewMsgInvite: Calling name , number 9067259847, Calling oct3 0x00, oct_3a 0x80, ext_priv 0x00, Called number
5555317884, oct3 0x00
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIContinueNewMsgInvite: Carrier id code , prev_cid NONE, next_cid NONE, prev_tgrp NONE, next_tgrp NONE
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIContinueNewMsgInvite: Requires reliable-provisional support
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIValidateRequestUri: Not Enabled
06:19:26: //-1/EE5EC9DD8170/SIP/Info/sipSPIRscmsmAvail: Value returned by check is = 0
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_IsSDPPassthruEnabled: - 0
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_GetHdrPassthruCfg: Hdr passthrough config:1 tag:0
06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ
06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLMain: dir:2, method:102, resp_code:0, container:6A01759C
06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLExtractSessionExpiresHdr:
Session-Expires value: 3600 refresher: uac
06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLExtractMinSEHdr: Min-SE Duration: 90
06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLGetInternalSREvent: E_STSL_INITIAL_SR_REQ
06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLInitialSRReqPeerEventGen: sending received session expires to the peer leg
06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLPrintTDContainer: Peer-Event: E_STSL_PASS_ST_PARAMS, SE Value:3600, SE Refresher:uac, Min-SE Value:1800,
flags:2001
06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLMain:
SE: 3600;refresher:uac peer refresher:none, flags:2001, posted event:E_STSL_INVALID_PEER_EVENT, reason:4
Configured SE:1800, Configured Min-SE:1800
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIProcessHistoryInfoHeader: No HI headers recvd from app container
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIProcessReplacesHeader: No replaces hdr found
SIP: Warning: Unrecognized attribute (X-vrzcap)
SIP: Warning: Unrecognized attribute (X-vrzcap)
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIValidateConnectionAddress: Dest port = 24402
SIP: (129) Attribute mid, level 1 instance 1 not found.
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: calling reg_invoke_ip_first_hop()
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: calling ip_best_local_address()
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/resolve_media_ip_address_to_bind: return addr 178.208.X.X
06:19:26: //129/EE5EC9DD8170/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 178.208.X.X
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(98) reserved for codec g729r8
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(98) reserved for codec g729r8
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(96) reserved for codec g729abr8
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(96) could not be reserved
as its in use by other codec g729abr8
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPT: Requested payload-Type (96) is reserved by another application
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 96
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAllocateFreeDynamicPT: Allocating free Dynamic Payload : 99 for Codec:
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(97) reserved for codec g729br8
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(97) could not be reserved
as its in use by other codec g729br8
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPT: Requested payload-Type (97) is reserved by another application
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 97
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAllocateFreeDynamicPT: Allocating free Dynamic Payload : 102 for Codec:
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) reserved for codec No Codec
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 99
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) could not be reserved
as its in use by other codec No Codec
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPT: Requested payload-Type (101) is reserved by another application
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 103
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIUpdateDynamicPayloadunused: Unreserving dynamic payload type 101
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIAllocateFreeDynamicPT: Allocating free Dynamic Payload : 101 for Codec:
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711ulaw) Negotiation Successful on Static Payload for m-line 1
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoPtimeNegotiation: One ptime attribute found - value:10
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711ulaw ptime :10, codecbytes: 80
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :80, ptime: 10
06:19:26: //129/EE5EC9DD8170/SIP/Media/sipSPIDoPtimeNegotiation: Offered ptime:10, Negotiated ptime:10 Negotiated codec bytes: 80 for codec g711ulaw
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPISetFaxFlags: FAX_PASSTHROUGH = 0, END_FAX_PASSTHROUGH = 0
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(101) reserved for codec
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIReserveRtpNtePayload: Reserved the payload type 101 for RTP-NTE
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoDTMFRelayNegotiation: RTP-NTE DTMF relay option
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of partial named event(NE) match in fmtp list of events.
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: V150 NSE payload = 0, SSE payload = 0, SPRT payload=0
06:19:26: //129/EE5EC9DD8170/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-
line:1 and num-a-lines:0
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1
payload_type=0, codec_bytes=80, codec=g711ulaw, dtmf_relay=rtp-nte
stream_type=voice+dtmf (1), dest_ip_address=188.254.68.67, dest_port=24402
06:19:26: //129/EE5EC9DD8170/SIP/State/sipSPIChangeStreamState: Stream (callid = -1) State changed from (STREAM_DEAD) to (STREAM_ADDING)
06:19:26: //129/EE5EC9DD8170/SIP/Media/sipSPIUpdCallWithSdpInfo:
Preferred Codec : g711ulaw, bytes :160
Preferred DTMF relay : rtp-nte
Preferred NTE payload : 101
Early Media : No
Delayed Media : No
Bridge Done : No
New Media : No
DSP DNLD Reqd : No
06:19:26: //129/EE5EC9DD8170/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr
06:19:26: //129/EE5EC9DD8170/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 178.208.X.X
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_report_media_to_peer:
callId 129 peer 0 flags 0x201 state STATE_IDLE
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_vcc_ProcessXcoderNeeded: xcoder_attached not yet initialised for this call.
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_report_media_to_peer: Xcoder not yet used for the call
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
CallID 129, Peer CallID 0, sdp 0x69EC3234 channels 0x6A8763C4
06:19:26: //129/EE5EC9DD8170/SIP/Info/copy_channels:
callId 129 size 0 ptr 0x6899F6D4)
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
CCB t38 version 0 ipip_caps version 0
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
CCB fax rate 2 ipip_caps rate 14400
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: reset the switch..
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 8 mline 1
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711alaw
06:19:26: //129/EE5EC9DD8170/SIP/Info/codec_found:
Codec to be matched: 6
06:19:26: //129/EE5EC9DD8170/SIP/Info/codec_found: No match for the codecs found..
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 0 mline 1
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711ulaw
06:19:26: //129/EE5EC9DD8170/SIP/Info/codec_found:
Codec to be matched: 5
06:19:26: //129/EE5EC9DD8170/SIP/Info/codec_found: codecs[i] = 5 & codec = 5 are same..
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: ADD AUDIO CODEC 5
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :80, ptime: 10
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation done: stream->negotiated_ptime=10,stream->negotiated_codec_bytes=80,
coverted ptime=10 stream->mline_index=1, media_ndx=1
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
failed to update call entry
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Adding codec 5 ptype 0 time 10, bytes 80 as channel 0 mline 1 ss 1 188.254.68.67:24402
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 18 mline 1
06:19:26: //129/EE5EC9DD8170/SIP/Media/sipSPISelectCodecVersion: Codec (g729r8) is not in preferred list
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: An exact codec match not configured, using interoperable codec g729r8
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g729r8
06:19:26: //129/EE5EC9DD8170/SIP/Info/codec_found:
Codec to be matched: 16
06:19:26: //129/EE5EC9DD8170/SIP/Info/codec_found: No match for the codecs found..
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 98 mline 1
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 96 mline 1
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 97 mline 1
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:
Hndl ptype 101 mline 1
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: setting ipip_caps DTMF to RFC2833: callid = 129, dtmf = 6
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[channel_ndx].codec
= 5
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[channel_ndx].codec
= -1
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_report_media_to_peer:
callId 129 flags 0x100 state STATE_IDLE
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_report_media_to_peer:
Report initial call media
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_report_media_to_peer: ccb->flags 0x804000C, ccb->pld.flags_ipip 0x201
06:19:26: //129/EE5EC9DD8170/SIP/Info/copy_channels:
callId 129 size 240 ptr 0x69E20A34)
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_report_media_to_peer:
CCSIP: Unable to report channel ind
06:19:26: //129/EE5EC9DD8170/SIP/Info/ccsip_update_srtp_caps: 5798: Posting Remote SRTP caps to other callleg.
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_report_media_to_peer: do cc_api_caps_ind()
06:19:26: //129/EE5EC9DD8170/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : voice+dtmf
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : -1
Negotiated Codec : g711ulaw, bytes :80
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated DTMF relay : rtp-nte
Negotiated NTE payload : 101 (tx), 101 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [178.208.X.X]:0
Media Dest Addr/Port : [188.254.68.67]:24402
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIHandleInviteMedia:
Negotiated Codec : g711ulaw, bytes :80
Preferred Codec : g711ulaw, bytes :160
Preferred DTMF relay 1 : 6
Preferred DTMF relay 2 : 0
Negotiated DTMF relay : 6
Preferred and Negotiated NTE payloads: 101 101
Preferred and Negotiated NSE payloads: 100 0
Preferred and Negotiated Modem Relay: 0 0
Preferred and Negotiated V150.1 Modem Passthrough: 0 0
Preferred and Negotiated V150.1 Modem Relay: 0 0
Preferred and Negotiated Modem Relay GwXid: 1 0
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoQoSNegotiationWithMediaLine: QOS negotiation for mline_index 1
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIDoStreamQoSNegotiation: Best effort
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPICanSetFallbackFlag: Local Fallback is not active
06:19:26: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 17550 for stream 1
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIUpdateSrcSdpFixedPart: Reserving rtp port for stream 1, src_port=17550
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPISetMediaDirectionForStream: Setting Media direction SENDRECV for stream 1
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Setting stream 1 portnum to 17550
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIUpdateSrcSdpVariablePart:
SIP update src sdp, negoitated codec 5, payload type 0
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIUpdateSrcSdpVariablePart: Negotiated method of dtmf relayand pyld: 6 101
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIAddBillingInfoToCcb: sipCallId for billing records = isbc6994325518770806443-1385214296-16204
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentCPA: No CPA found in inbound container
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIProcessCPA: No x-cisco-cpa content found
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_GetHdrPassthruCfg: Hdr passthrough config:1 tag:0
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_IsContentPassthruEnabled: - 0
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_ExtractPassthruContentFromSipContainer: Passthru Content Not Enabled
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_store_channel_info: Store channelInfo in CallInfo
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_store_channel_info: dtmf negotiation done, storing negotiated dtmf = 6,
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIShrlCall: Check peer: 10020 for Shared-Line call, callid: 129
06:19:26: //129/EE5EC9DD8170/SIP/Info/ccsip_set_bearer_capability:
Bearer Capability: Speech (0x00)
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQSIG: No QSIG Body found in inbound container
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentQ931: No RawMsg Body found in inbound container
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPICreateNewRawMsg: No Data to form The Raw Message
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIContinueNewMsgInvite: ccsip_api_call_setup_ind returned: SIP_SUCCESS
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIUaddccCallIdToTable: Adding call id 81 to table
06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:100, container:6A0173E4
06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLValidateSessRefreshMsg: Ignoring 1xx response for session timer processing
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPISendInviteResponse: Associated container=0x6A0173E4 to Invite Response 100
06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipSPITransportSendMessage: msg=0x6A5A1A34, addr=188.254.68.66, port=9290, sentBy_port=9290, local_addr=, is_req=0,
transport=1, switch=0, callBack=0x0
06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipTransportLogicSendMsg: Trying to send resp=0x6A5A1A34 to default port=9290
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostRequestConnection: Posting UDP conn create request for addr=188.254.68.66, port=9290, context=0x68ABB118
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportSetConnWaitTimer: Wait timer set for connection=0x68ABCB0C,addr=188.254.68.66, port=9290
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportSetConnWaitTimer:
Wait Conn Timer started for 5000 msec
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipCreateConnInstance: Created new initiated conn=0x68ABCB0C, connid=-1, addr=188.254.68.66, port=9290, local_addr=,
transport=UDP
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:188.254.68.66, rport:9290 with laddr:
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipInstanceGetConnectionId: Registering gcb=0x6A874E70 with connection=0x68ABCB0C
06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipTransportLogicSendMsg: Waiting for Connection for sending msg=0x6A5A1A34
06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipSPITransportSendMessage: Deferred sending msg=0x6A5A1A34
06:19:26: //129/EE5EC9DD8170/SIP/State/sipSPIChangeState: 0x6A874E70 : State change from (STATE_IDLE, SUBSTATE_NONE) to (STATE_RECD_INVITE, SUBSTATE_NONE)
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIProcessContactInfo: Previous Hop 188.254.68.66:9290
06:19:26: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_PROCEEDING
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: switch(ev.ev_id: 165)
06:19:26: //129/EE5EC9DD8170/SIP/Info/ccsip_event_handler:
ccsip_event_handler: peer ID 130 chans 0x6780D478 event 165 flags 0x844001C 0x100 0x601 data 0x6780D478
06:19:26: //129/EE5EC9DD8170/SIP/Info/ccsip_event_handler:
ccsip_event_handler: CC_EV_H245_SET_MODE: peer ID 130 chans 0x6780D478 event 165 flags 0x844001C 0x100 0x601 data 0x6780D478, type = 1
06:19:26: //129/EE5EC9DD8170/SIP/Info/ccsip_gw_set_sipspi_mode: Setting SPI mode to SIP-H323
06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SET_MODE
06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLMain:
SE: 3600;refresher:uac peer refresher:none, flags:2001, posted event:E_STSL_INVALID_PEER_EVENT, reason:4
Configured SE:1800, Configured Min-SE:1800
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_event_handler: CC_R_SUCCESS_WITH_CONFIRMED
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 3
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 58
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWConnectionCreated: context=0x68ABB118
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessConnCreated: gConnTab=0x68ABB118, addr=188.254.68.66, port=9290, local_addr=, connid=3,
transport=UDP
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerProcessConnCreated: connection instance created for addr:188.254.68.66, port:9290 local_addr=
local_port=57282
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportStopConnWaitTimer: Wait timer stopped for connection=0x68ABCB0C,addr=188.254.68.66, port=9290
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipInstanceHandleConnectionCreated: Moving connection=0x68ABCB0C, connid=3 state to established. local_addr=,
local_port=57282
06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipTransportPostInternalMsg: Posting Internal Msg type=0
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 63
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x6A5A1A34, addr=188.254.68.66, port=9290, local_addr=, connId=3 for UDP
06:19:26: //129/EE5EC9DD8170/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 188.254.68.66:9290;branch=z9hG4bK-6110d60075c89eab-a3c000c-1
From: <sip:[email protected];user=phone>;tag=sbc09026994325518770806443
To: <sip:[email protected];user=phone>
Date: Sat, 23 Nov 2013 13:42:29 GMT
Call-ID: isbc6994325518770806443-1385214296-16204
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
06:19:26: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_ALERTING
06:19:26: //-1/xxxxxxxxxxxx/SIP/Error/ccsip_call_service_msg: ccb NULL, unable to update the callinfo ui parameters
06:19:26: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_MEDIA_EVENT
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 5
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIAddCiscoGcid: Fatal Error in parsing CCB/Msg
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIStoreTunnelData: Container /RawMessage Absent
06:19:26: //129/EE5EC9DD8170/SIP/Error/sipSPI_ipip_set_history_info_header: Not SIP2SIP mode
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIUaddCcbToUASRespTable: ****Adding to UAS Response table.
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPIUaddCcbToTable: Added to table. ccb=0x6A874E70 key=isbc6994325518770806443-1385214296-1620415B6280-0
06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_RESP
06:19:26: //129/EE5EC9DD8170/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:180, container:6A017B1C
06:19:26: //129/EE5EC9DD8170/SIP/Info/Session-Timer/sipSTSLValidateSessRefreshMsg: Ignoring 1xx response for session timer processing
06:19:26: //129/EE5EC9DD8170/SIP/Event/sipSPICreateRpid: Received Octet3A=0x00 -> Setting ;screen=no ;privacy=off
06:19:26: //129/EE5EC9DD8170/SIP/Info/sipSPISendInviteResponse: Associated container=0x6A017B1C to Invite Response 180
06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipSPISendInviteResponse: Sending 180 Response to the Transport Layer
06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipSPITransportSendMessage: msg=0x6A5A1A34, addr=188.254.68.66, port=9290, sentBy_port=9290, local_addr=, is_req=0,
transport=1, switch=0, callBack=0x618A57B8
06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipSPITransportSendMessage: Proceedable for sending msg immediately
06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipTransportLogicSendMsg: switch transport is 0
06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipTransportLogicSendMsg: Trying to send resp=0x6A5A1A34 to default port=9290
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipConnectionManagerGetConnection: connection required for raddr:188.254.68.66, rport:9290 with laddr:
06:19:26: //129/EE5EC9DD8170/SIP/Transport/sipTransportLogicSendMsg: Connection obtained...sending msg=0x6A5A1A34
06:19:26: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportPostSendMessage: Posting send for msg=0x6A5A1A34, addr=188.254.68.66, port=9290, local_addr=, connId=3 for UDP
06:19:26: //129/EE5EC9DD8170/SIP/Info/sentInviteResponse18x: Sent a 18x Response
06:19:26: //129/EE5EC9DD8170/SIP/Info/sentInviteResponse18x: Transaction active. Facilities will be queued.
06:19:26: //129/EE5EC9DD8170/SIP/State/sipSPIChangeState: 0x6A874E70 : State change from (STATE_RECD_INVITE, SUBSTATE_NONE) to (STATE_SENT_ALERTING, SUBSTATE_NONE)
06:19:26: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 30ty-ID:
<sip:[email protected]>;party=called;screen=no;privacy=off
Contact: <sip:[email protected]:5060>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
06:19:27: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [188.254.68.66]:9290, local_address:[ - ]
06:19:27: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [188.254.68.66]:9290, local_address:[ - ]
06:19:27: //129/EE5EC9DD8170/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 188.254.68.66:9290;branch=z9hG4bK-6110d60075c89eab-a3c000c-2
From: <sip:[email protected];user=phone>;tag=sbc09026994325518770806443
To: <sip:[email protected];user=phone>;tag=15B6280-0
Date: Sat, 23 Nov 2013 13:42:30 GMT
Call-ID: isbc6994325518770806443-1385214296-16204
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 2 PRACK
Content-Length: 0
06:19:27: //129/EE5EC9DD8170/SIP/Msg/ccsipDisplayMsg:
Sent:
UPDATE sip:[email protected]:9290;user=phone SIP/2.0
Via: SIP/2.0/UDP 178.208.X.X:5060;branch=z9hG4bK120
From: <sip:[email protected];user=phone>;tag=15B6280-0
To: <sip:[email protected];user=phone>;tag=sbc09026994325518770806443
Date: Sat, 23 Nov 2013 13:42:30 GMT
Call-ID: isbc6994325518770806443-1385214296-16204
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Timestamp: 1385214150
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 UPDATE
Contact: <sip:[email protected]:5060>
Min-SE: 1800
Remote-Party-ID: <sip:[email protected]>;party=called;screen=yes;privacy=off
Content-Length: 0
06:19:27: //129/EE5EC9DD8170/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 188.254.68.66:9290;branch=z9hG4bK-6110d60075c89eab-a3c000c-2
From: <sip:[email protected];user=phone>;tag=sbc09026994325518770806443
To: <sip:[email protected];user=phone>;tag=15B6280-0
Date: Sat, 23 Nov 2013 13:42:30 GMT
Call-ID: isbc6994325518770806443-1385214296-16204
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 2 PRACK
Content-Length: 0
06:19:31: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [188.254.68.66]:9290, local_address:[ - ]
06:19:31: //129/EE5EC9DD8170/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 188.254.68.66:9290;branch=z9hG4bK-6110d60075c89eab-a3c000c-1
From: <sip:[email protected];user=phone>;tag=sbc09026994325518770806443
To: <sip:[email protected];user=phone>;tag=15B6280-0
Date: Sat, 23 Nov 2013 13:42:30 GMT
Call-ID: isbc6994325518770806443-1385214296-16204
CSeq: 1 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=16
Content-Length: 0
06:19:31: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [188.254.68.66]:9290, local_address:[ - ]
06:19:31: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1
06:19:31: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x0
06:19:31: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 188.254.68.66:9290;branch=z9hG4bK-6110d60075c89eab-a3c000c-1
Call-ID: isbc6994325518770806443-1385214296-16204
From: <sip:[email protected];user=phone>;tag=sbc09026994325518770806443
To: <sip:[email protected];user=phone>;tag=15B6280-0
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
06:19:31: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog
06:19:31: //129/EE5EC9DD8170/SIP/Info/sipSPIFindCcbUASRespTable: *****CCB found in UAS Response table. ccb=0x6A874E70
06:19:31: //129/EE5EC9DD8170/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x6A874E70 key=isbc6994325518770806443-1385214296-1620415B6280-0
06:19:31: //129/EE5EC9DD8170/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd
06:19:31: //129/EE5EC9DD8170/SIP/Info/sipSPIStopRequestPendingTimer: Stopping Request Pending Timer
06:19:31: //129/EE5EC9DD8170/SIP/Info/sipSPI_ipip_free_codec_profile: Codec Profiles Freed
06:19:31: //129/EE5EC9DD8170/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 6A874E70
06:19:31: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContextFromTable: NO context for key[52]
c2801#
c2801#Hi Ahmed,
Looking at the logs, it seems some important messages are missing..
can you please again collect the following debugs ?
- debug voip ccapi inout
- debug ccsip all
- debug voice translation
Thanks,
Piyush -
SiP Phone wont dial inbound or outbound
I have 9971 phone and was dialing sip to sip and sip h323 on the network, but now I get Apr 20 14:17:58.911: %VOICE_IEC-3-GW: Application Framework Core: Internal error
(Toll fraud call rejected): IEC=1.1.228.3.31.0 on callID 12 GUID=BCBC7FIHi Wharrison,
can you please provide the call flow and where do you see this error.
I am guessing the call is from an IP phone regsitered to CUCM --> SIP truk --> CUBE --> provider.. Is this right?
Please let me know where do you see the error.
Thanks,
Manoj -
CallManager 4.x to SIP ITSP
Hello,
I am trying to find out if a CallManager 4.x system can communicate to an IP Telephony Service Provider over SIP to their Sessions Border Controller. From what it looks like, I will need an IP2IP gateway to talk H323 to the CCM and SIP to the ITSP. Has anyone successfully done this before?
Any help or experiences would be greatly appreciated.We have tested this in our lab, and this was working well. An Cisco 2811 with ver. 12.4 IP2IP was used for this test and H323 to SIP, H323 to H323 and SIP to H323 was working well.
Config Cisco router :
Building configuration...
Current configuration : 2983 bytes
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
enable password cisco
no aaa new-model
ip cef
no ip dhcp use vrf connected
no ip dhcp conflict logging
ip dhcp excluded-address 10.193.25.1 10.193.25.65
ip dhcp excluded-address 172.16.1.1 172.16.1.9
ip dhcp excluded-address 10.193.25.70 10.193.25.80
ip dhcp pool 10.193.25.0
network 10.193.25.0 255.255.255.0
option 150 ip 10.193.25.113
default-router 10.193.25.111
lease infinite
multilink bundle-name authenticated
voice-card 0
no dspfarm
dsp services dspfarm
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
call start interwork
sip
interface Loopback0
ip address 10.0.0.1 255.255.255.255
interface FastEthernet0/0
ip address 10.193.25.111 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 10.193.25.111
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface Content-Engine1/0
no ip address
shutdown
ip route 0.0.0.0 0.0.0.0 10.193.25.1
no ip http server
control-plane
dial-peer voice 10 voip
description VoIP to live callmanager
destination-pattern 3...
progress_ind connect enable 8
session target ipv4:10.193.1.5
dtmf-relay h245-alphanumeric
codec g711alaw
dial-peer voice 20 voip
description VoIP to Test Callmanager
tone ringback alert-no-PI
destination-pattern 2...
progress_ind setup enable 3
progress_ind progress enable 8
progress_ind connect enable 8
session target ipv4:10.193.25.113
dtmf-relay h245-alphanumeric
codec g711alaw
dial-peer voice 30 voip
description to VoIP/AA at Test Callmanager
destination-pattern 500.
session target ipv4:10.193.25.113
dtmf-relay h245-alphanumeric
codec g711alaw
dial-peer voice 1 voip
description to H323 External GW
destination-pattern 0T
session target ipv4:10.193.1.4
dtmf-relay h245-alphanumeric
codec g711alaw
dial-peer voice 200 voip
description to SIP Soft IP-Phone
destination-pattern 1999
session protocol sipv2
session target ipv4:10.193.10.9
dtmf-relay rtp-nte
codec g711alaw
dial-peer voice 100 voip
tone ringback alert-no-PI
description 3th party hardware SIP IPPhone
destination-pattern 1...
session protocol sipv2
session target ipv4:10.193.25.200:5060
dtmf-relay rtp-nte h245-alphanumeric
codec g711alaw
no vad
sip-ua
retry options 0
gatekeeper
shutdown
line con 0
line aux 0
line 66
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120
line vty 0 4
password cisco
login
scheduler allocate 20000 1000
end
Router# -
Call/video not working between Cisco jabber for Windows and VCS control C40s
Hello,
I've been struggling with no luck how to make a call using Cisco Jabber for Windows 9.6.0 registered to CM 8.6.2 with intercluster ICT to another CM 8.6.2 where we have a VCS Control 7.0.2 via GK H225, and all C40s are registered as H.323.
The VCS has interworking between H323 and SIP, however not sure if there is any problem with that. Assuming it is ok, not sure either if I'm facing any interoperability issue because in my remote site I have C40 (H323 registered at VCS and SIP listening mode) and cisco jabber for windows which is SIP based.
If is not possible, would I be able to change my C40 from H323 to SIP at VCS, or have both H323/SIP registered at VCS? If so, will I need to change as well instead of GK I'll have to establish a SIP Trunk between the CM and VCS?
Another thing I do not believe either I would be able to have one VCS connected with two clusters, right?
I'm just trying to find a solution in case my current topology is not compatible, but feel free if you have any better idea to make it work.
Anyway here is what is happening:
When I make a call from my cisco jabber windows to C40 using alias number. The call is being redirected just fine to the C40 and it rings, however when someoene or the auto answer picks it up, the call dropped right away.
However, if I enabled the MTP in my CSF device, the call gets longer before dropping. I was even able to see my jabber " start video" turns green, before was grayed out all the time and the call dropped faster. I hear a fast busy tone.
I'm able to provide SDI traces, logs, diagnostic sip/h323 calls from VCS in order to know for sure if this is an incompatible issue or something I can workaround.
Let me know if someone of you are interested in read these logs or could point me on the right direction.
Thanks!Ok,
I have looked at both logs. I have to mentinon though that you didnt
provide the log that shows the h323 setup between cucm and the VCS. This
is most likely because the call originated from a different cucm than
the ones you provided the logs from.
The call would have orginated from the first cucm in the cucm group of
this trunk: Name=RL_TRUNK_VIDEO
The cucm ip will be : 10.252.53.10.
This is the VCS log that confirms where the h323 request originated
from:
pr 10 22:50:29 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:29,187"
Module="network.h323" Level="DEBUG": Src-ip="10.252.53.10" Src-
port="54000"
Received RAS PDU:
Having said that here is my analysis of the logs that you sent..
Jabber sent an INVITE to CUCM and advertised all the codecs (audio and
video it can support)..
Observer that Jabber says it doesnt support G729 anexB
21:55:16.576 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message
from 10.223.20.73 on port 54677 index 90661 with 2220 bytes:
[862370,NET]
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP 10.223.20.73:54677;branch=z9hG4bK000029d3
From: "4122107" <sip:[email protected]>;tag=00059a3c78000011000070b0
-00000e65
To: <sip:[email protected]>
Call-ID: [email protected]
Max-Forwards: 70
Date: Fri, 11 Apr 2014 01:55:16 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CSF/9.4.1
m=audio 19252 RTP/AVP 0 8 18 105 104 101
c=IN IP4 10.223.20.73
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 28878 RTP/AVP 97
c=IN IP4 10.223.20.73
++++Now lets observer the capabilites exchange during h245 negotiation
between cucm and VCS++++
Here CUCM advertises its caps to VCS (afterreceiving caps from VCS)
Note that G729A, G729AB, G729 is all advertised..
Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,017"
Module="network.h323" Level="DEBUG": Src-ip="10.252.53.10" Src-
port="45660"
Received H.245 PDU:
value MultimediaSystemControlMessage
::= request : terminalCapabilitySet
capabilityTableEntryNumber 2,
capability receiveAudioCapability :
g729wAnnexB : 6
capabilityTableEntryNumber 3,
capability receiveAudioCapability : g729AnnexAwAnnexB : 6
capabilityTableEntryNumber 4,
capability
receiveAudioCapability : g729 : 6
capabilityTableEntryNumber 5,
capability receiveAudioCapability :
g729AnnexA : 6
++++++
After doing MSD (master slave determination, we move to the OLC phas e..
Here we see that the far end..c40 wants to use G729AB for media++++
Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,783"
Module="network.h323" Level="DEBUG": Src-ip="10.224.114.11" Src-
port="11163"
Received H.245 PDU:
value MultimediaSystemControlMessage
::= request : openLogicalChannel :
forwardLogicalChannelNumber 1,
forwardLogicalChannelParameters
dataType audioData :
g729AnnexAwAnnexB : 20,
multiplexParameters
h2250LogicalChannelParameters :
+++Next VCS sends G729AB as the codec to use to CUCM+++
Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,784"
Module="network.h323" Level="DEBUG": Dst-ip="10.252.53.10" Dst-
port="45660"
Sending H.245 PDU:
value MultimediaSystemControlMessage
::= request : openLogicalChannel :
forwardLogicalChannelNumber 1,
forwardLogicalChannelParameters
dataType audioData :
g729AnnexAwAnnexB : 20,
multiplexParameters
h2250LogicalChannelParameters :
++++The next thing we get is an OLC reject from CUCM and this is where
th call drops++
Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,790"
Module="network.h323" Level="DEBUG": Src-ip="10.252.53.10" Src-
port="45660"
Received H.245 PDU:
value MultimediaSystemControlMessage
::= response : openLogicalChannelReject :
forwardLogicalChannelNumber 1,
cause dataTypeNotSupported : NULL
Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,790"
Module="network.h323" Level="INFO": Dst-ip="10.224.114.11" Dst-
port="11163"
Detail="Sending H.245 OpenLogicalChannelRejResponse
+++We then receive a call release from cucm with cause code of 47:
resource unavailable++++
Apr 10 22:50:32 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:32,365"
Module="network.h323" Level="DEBUG": Src-ip="10.252.53.10" Src-
port="50913"
Received H.225 PDU:
Q931
Message Type: Release
Complete
Call reference flag: Message sent from originating side
Call reference value: 0x7b
Info Element : Cause
Location: Usr
Cause Value: Resource unavailable
Info Element : User User
Length = 22
Suggestions:
Change the region setting between the ICT trunk to VCS and Jabber to use
G711 and test again. -
CISCO Jabber 8.6.2 and CME 8.6
Hello,
I want to use Cisco Jabber 8.6.2 with Call manager Express 8.6
I configured the IPhone on CME and is working ok on local wireless LAN,
When I’m using the VPN I can place call's inside the network but I can't use on outside line. I have no sound.
Also if I place a call and the I end the call, the called phone rings and is not stopping. So I think is a call disconnect problem.
Bellow is a part of the configuration.
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
call start slow
sip
session transport tcp
registrar server
voice class codec 1
codec preference 1 g729r8
voice class custom-cptone romania
dualtone busy
frequency 450
cadence 170 170
dualtone disconnect
frequency 450
cadence 170 170
voice register global
mode cme
source-address 10.12.4.252 port 5060
max-dn 10
max-pool 10
authenticate register
hold-alert
tftp-path flash:
create profile sync 0000545624458818
voice register dn 1
number 5146
call-forward b2bua all 5108
call-forward b2bua busy 5160
call-forward b2bua noan 5160 timeout 18
name Ioan Stanciu
shared-line
label Ioan
voice register pool 10
registration-timer max 720 min 660
id mac 68A8.6D91.3FE0
session-transport tcp
type CiscoMobile-iOS
number 1 dn 1
username 5146 password 5146
voice-port 0/0/0
supervisory disconnect dualtone mid-call
supervisory custom-cptone romania
no battery-reversal
timeouts call-disconnect 2
timeouts wait-release 2
connection plar opx immediate 5150
caller-id enable
voice-port 0/0/1
supervisory disconnect dualtone mid-call
supervisory custom-cptone romania
no battery-reversal
timeouts call-disconnect 2
timeouts wait-release 2
connection plar opx immediate 5150
caller-id enable
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
codec g722-64
maximum sessions 1
associate application SCCP
dial-peer voice 2 pots
preference 2
destination-pattern 9T
incoming called-number .
direct-inward-dial
port 0/0/1
telephony-service
video
maximum bit-rate 256
no auto-reg-ephone
em logout 0:0 0:0 0:0
max-ephones 25
max-dn 120
ip source-address 10.12.4.252 port 2000
max-redirect 7
auto assign 1 to 8
service phone videoCapability 1
service phone paramEdibility 1
no service directed-pickup
timeouts interdigit 6
timeouts ringing 15
Any idea what can be the problem ?Hello,
Is the connect over 3G enabled on your iPhone? Try this.
Btw, what have you put as device ID on your iPhone? the MAC address?
To my knowledge, the device ID is TCTXXX for CUCM, there is no info related on how to add it for CME 8.6
Regards -
Unity Connection Voice Mail Issue
Hi,
I have a weird Unity Connection Voice Mail issue and would appriciate any help i can get. Many thanks in advance.
I have Unity connection 7.0 and CUCM 7.0 integrated in a lab enviroment and here is what happens.
When I place a call internally say from 2001 to 2002, things work as expected, 2002 rings and it goes to voicemail where I can leave voicemail and listen to it from 2002.
However, if I place call from the PSTN to the same number 2002 (or any other number in other sites etc) the call again rings in 2002 and goes to voicemail, unity cnx plays the greetings for 2002 and says record your message as usual. Everything up to point is fine, then when the time on the PSTN phone is showing around 14 seconds into the call unity starts playing, "to send this message press one", if I press one nothing happens.
I have rebuilt unity and cucm, even just configured the bare minimum in the lab and still getting the same result. I tried calling from E1 connection and T1 connection but with the same results.
I have run out of ideas...
===============================
Here is a call trace from an internal call:
CallData, 1, CallerId=2003, CalledId=2002, RedirectingId=2002, Origin=16, Reason=4, CallGuid=8B7859FD6C16417A9A07F507418DD25B, CallerName=, LastRedirectingId=2002, LastRedirectingReason=4, PortDisplayName=PhoneSystem-1-001
Application, 1, 2003, AttemptForward
State, 1, 2003, State - AttemptForward.cde!Dummy
State, 1, 2003, Event is [NULL]
Application, 1, 2003, PHTransfer
State, 1, 2003, State - PHTransfer.cde!LoadInfo
State, 1, 2003, Event is [TrueEvent]
Application, 1, 2003, PHGreeting
State, 1, 2003, State - PHGreeting.cde!PlayGreeting
Display, 1, 2003, Call answered if needed
Display, 1, 2003, Playing greeting for Subscriber: hq2
Display, 1, 2003, No DTMF received
Display, 1, 2003, Playing greeting for Subscriber: hq2
State, 1, 2003, Event is [RecordMsgEvent]
State, 1, 2003, State - PHGreeting.cde!RecordMsg
State, 1, 2003, Event is [NULL]
State, 1, 2003, State - PHGreeting.cde!RunEditMsg
Application, 1, 2003, -->MessageEditing
State, 1, 2003, State - MessageEditing.cde!CheckMsgMenuOpt
State, 1, 2003, Event is [EditMessageMenuEvent]
State, 1, 2003, State - MessageEditing.cde!PlayEditMenu
State, 1, 2003, Event is [HangupEvent]
State, 1, 2003, State - MessageEditing.cde!CheckMsgLength
State, 1, 2003, Event is [ManyEvent]
State, 1, 2003, State - MessageEditing.cde!SendMsg
State, 1, 2003, Event is [TrueEvent]
State, 1, 2003, State - MessageEditing.cde!ConfirmSend
State, 1, 2003, Event is [HangupEvent]
Application, 1, 2003, <--MessageEditing
State, 1, 2003, Event is [HangupEvent]
Display, 1, 2003, Idle
Display, 1, , Dialing (MWI) '2002'
Display, 1, , Idle
and here is a trace from an external (PSTN) call
Trying 142.100.64.13, 5000 ... Open
CallData, 1, CallerId=911, CalledId=2002, RedirectingId=2002, Origin=16, Reason=4, CallGuid=CA3DFD90846C4FE7B0D68298A7698287, CallerName=PSTN Emergency, LastRedirectingId=2002, LastRedirectingReason=4, PortDisplayName=PhoneSystem-1-001
Application, 1, 911, AttemptForward
State, 1, 911, State - AttemptForward.cde!Dummy
State, 1, 911, Event is [NULL]
Application, 1, 911, PHTransfer
State, 1, 911, State - PHTransfer.cde!LoadInfo
State, 1, 911, Event is [TrueEvent]
Application, 1, 911, PHGreeting
State, 1, 911, State - PHGreeting.cde!PlayGreeting
Display, 1, 911, Call answered if needed
Display, 1, 911, Playing greeting for Subscriber: hq2
Display, 1, 911, No DTMF received
Display, 1, 911, Playing greeting for Subscriber: hq2
State, 1, 911, Event is [RecordMsgEvent]
State, 1, 911, State - PHGreeting.cde!RecordMsg
State, 1, 911, Event is [NULL]
State, 1, 911, State - PHGreeting.cde!RunEditMsg
Application, 1, 911, -->MessageEditing
State, 1, 911, State - MessageEditing.cde!CheckMsgMenuOpt
State, 1, 911, Event is [EditMessageMenuEvent]
State, 1, 911, State - MessageEditing.cde!PlayEditMenu
State, 1, 911, Event is [HangupEvent]
State, 1, 911, State - MessageEditing.cde!CheckMsgLength
State, 1, 911, Event is [NULL]
Application, 1, 911, <--MessageEditing
State, 1, 911, Event is [NULL]
State, 1, 911, State - PHGreeting.cde!AfterMsg
State, 1, 911, Event is [NULL]
Display, 1, 911, IdleSounds like one way audio from PSTN to your Unity Connection, couple of things to check:
1. ensure your protocols are bound properly on the GW, i.e. SIP/H323/MGCP
2. Make sure IP routing is OK between Unity and the voice gateway
HTH,
Chris -
No ringbacktone for inbound calls with cucm 8.6
Hi,
we have this problem from many days...
we have two branches with cucm cluster(Publisher and Subscriber) at Head office and cisco untiy.The branches are connected to Head office through MPLS vpn and all the ip phones are registred to publisher located at headoffice.
our setup is like below
HO and BR2 having SIP lines and BR1 has PSTN Lines.
we implement greetings for head office and 2 branches at Headoffice Unity.
when any call comes to headoffice gateway the greetings will be played and call will be diverted to the appropriate extension.everything is fine.
But the problem is when the call comes to Branch gateway and the greetings will be played and the call gets diverted to the IP phone to which the caller dialed the extension. but the caller is not hearing the ringback tone while the extension is ringing. and the caller cannot know whether the extension is ringing or the call got disconnected.
i tried to change the " Send h225 User Information Message" in service parameters from "Use ANN for Ring Back" to H225 Info for call Progress Tone"
whenever i am changing to "H225 Info for call Progress Tone" then the branches problem getting solved but Headoffice getting the same problem.
please can anyone help............................Hi Carlo,
Thankyou for the Response...
here is the Runn config for BR1 Connected to PSTN lines....
voice-card 0
dspfarm
dsp services dspfarm
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
voice class h323 1
h225 timeout tcp establish 3
interface Tunnel100
description " Tunnel JED-RYD "
bandwidth 2048
ip address 10.10.0.1 255.255.255.252
tunnel source 172.31.217.202
tunnel destination 172.31.3.18
interface FastEthernet0/0
description DAMMAM Local LAN
no ip address
duplex auto
speed auto
interface FastEthernet0/0.20
description JEDDAH Local LAN
encapsulation dot1Q 20
ip address 192.168.20.5 255.255.255.0
interface FastEthernet0/0.21
description JEDDAH VOICE VLAN
encapsulation dot1Q 21
ip address 192.168.21.5 255.255.255.0
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.21.5
interface FastEthernet0/1
ip address 172.31.217.202 255.255.255.252
duplex auto
speed auto
router eigrp 200
network 10.10.0.0 0.0.0.3
network 192.168.20.0
network 192.168.21.0
no auto-summary
router bgp 65412
no synchronization
bgp log-neighbor-changes
neighbor 172.31.217.201 remote-as 65000
no auto-summary
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.20.1
ip route 192.168.20.50 255.255.255.255 192.168.20.1
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
access-list 23 permit 10.10.10.0 0.0.0.7
control-plane
voice-port 0/0/0
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/0/1
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/0/2
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/0/3
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/2/0
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
connection plar 2022
shutdown
impedance complex2
description STC
voice-port 0/2/1
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
shutdown
impedance complex2
description STC
voice-port 0/3/0
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/3/1
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/3/2
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
voice-port 0/3/3
supervisory disconnect dualtone mid-call
no battery-reversal
input gain -3
output attenuation -3
echo-cancel coverage 32
cptone BE
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
timing hookflash-out 500
timing guard-out 300
timing sup-disconnect 50
connection plar opx 2050
impedance complex2
description STC
caller-id alerting dsp-pre-allocate
sccp local FastEthernet0/0.21
sccp ccm 192.168.12.190 identifier 1 priority 1 version 5.0.1
sccp ccm 192.168.12.189 identifier 2 priority 2 version 5.0.1
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register CONFJEDRAW
associate profile 2 register TRNJED
associate profile 3 register MTPJED
switchover method immediate
switchback method immediate
switchback interval 15
dspfarm profile 2 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 2
associate application SCCP
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
shutdown
dspfarm profile 3 mtp
codec g729r8
maximum sessions software 250
associate application SCCP
shutdown
dial-peer voice 1 pots
dial-peer voice 1000 voip
description To CallManager - SBWPMPUB
destination-pattern [1-5]...
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.190
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 9001 pots
description ** 02-6140294(outgoing) **
destination-pattern [^2].T
port 0/0/1
dial-peer voice 9002 pots
description ** 02-6140295(outgoing) **
destination-pattern [^2].T
port 0/0/2
dial-peer voice 9003 pots
description ** 02-6140296(outgoing) **
destination-pattern [^2].T
port 0/0/3
dial-peer voice 9004 pots
description ** 02-6140293(outgoing) **
destination-pattern [^2].T
port 0/0/0
dial-peer voice 290 pots
incoming called-number .
direct-inward-dial
dial-peer voice 9006 pots
description ** 02-6529323(local) **
destination-pattern [^0].T
port 0/3/0
dial-peer voice 9010 pots
description ** 02-6578249(local) **
destination-pattern [^0].T
port 0/3/1
dial-peer voice 9011 pots
description "to pstn service"
shutdown
destination-pattern 0.T
port 0/3/3
dial-peer voice 9009 pots
description "to pstn service"
shutdown
destination-pattern [^0].T
port 0/3/2
dial-peer voice 9005 pots
destination-pattern .T
dial-peer voice 1001 voip
description To CallManager - Subscriber
destination-pattern [1-5]...
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.189
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 1002 voip
description " TO Unity Greetings"
destination-pattern 2050
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.190
dtmf-relay h245-alphanumeric
no vad
dial-peer voice 1003 voip
description " TO Unity Greetings"
destination-pattern 2050
progress_ind progress enable 8
voice-class codec 1
voice-class h323 1
session target ipv4:192.168.12.189
dtmf-relay h245-alphanumeric
no vad -
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