11-11-2013 10:53 AM - edited 03-16-2019 08:20 PM
Hi,
I'm still struggeling on dtmf tones within my router. While the SCCP Phones are working well, the SIP phones do not generate any DTMF tones. Problem occurs external and internal (e.g. when the SIP Phone calls the mailbox)
Unfortunately the first on-hand solution (adding "pass-thru content sdp" in voice service voip / sip) didn't worked out. After adding this there is no more internal dial (SIP to SCCP) possible.
Configuration and debug attached.
Any ideas ?
Regards, Frank
****************** Configuration ************************
version 15.2
service timestamps debug datetime msec
service timestamps log datetime msec
!
!
boot-start-marker
boot-end-marker
!
!
no logging buffered
!
aaa new-model
!
!
aaa authentication login default local
aaa authorization exec default local
!
!
!
aaa session-id common
network-clock-participate wic 1
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
!
!
!
ip dhcp relay information trust-all
ip dhcp excluded-address 10.10.10.1 10.10.10.10
!
ip dhcp pool phone
network 10.10.10.0 255.255.255.0
default-router 10.10.10.1
dns-server 217.0.43.97 217.0.43.113
option 150 ip 10.10.10.1
!
!
ip name-server 217.0.43.97
ip cef
no ipv6 cef
!
multilink bundle-name authenticated
!
!
voice rtp send-recv
!
voice service voip
ip address trusted list
no ip address trusted authenticate
callmonitor
no callmonitor trace
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip handle-replaces
sip
registrar server expires max 3600 min 180
asymmetric payload full
video screening
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g723r53
codec preference 4 g729r8
codec preference 5 g729br8
codec preference 6 g726r32
codec preference 7 g728
!
!
voice register global
mode cme
source-address 10.10.10.1 port 5060
max-dn 10
max-pool 4
load 9971 sip9971.9-3-2SR1-1
authenticate register
authenticate realm all
timezone 21
time-format 24
date-format D/M/Y
voicemail 999
tftp-path flash:
file text
create profile sync 0000420080667251
network-locale DE
user-locale DE
camera
video
overlap-signal
!
voice register dn 1
number 91
allow watch
no-reg
mwi
!
voice register pool 1
registration-timer max 300 min 60
id mac 0008.3030.6AAA
session-transport tcp
type 9971
number 1 dn 1
template 1
voice-class codec 1
username cisco password cisco
no vad
camera
video
!
!
!
voice translation-rule 1
rule 2 /^\(\+..\)\(.*\)/ /0\2/
!
voice translation-rule 2
rule 1 /123456/ /91/
!
voice translation-rule 3
rule 1 /^.*/ /123456/
!
!
voice translation-profile IN
translate calling 1
translate called 2
!
voice translation-profile OUT
translate calling 3
!
!
voice-card 0
!
!
license udi pid C1861-UC-B/K9-MS sn FHK144871U3
file privilege 0
!
!
no ip ftp passive
!
!
interface FastEthernet0/0
description ***ISP Provider***
ip address dhcp
no ip proxy-arp
ip nat outside
ip virtual-reassembly in
duplex auto
speed auto
!
interface Integrated-Service-Engine0/0
ip unnumbered Vlan1
ip nat inside
ip virtual-reassembly in
service-module ip address 10.10.10.2 255.255.255.0
service-module ip default-gateway 10.10.10.1
!
!
interface Vlan1
ip address 10.10.10.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
!
ip forward-protocol nd
ip http server
ip http authentication aaa
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:
!
!
ip nat inside source list 1 interface FastEthernet0/0 overload
ip route 0.0.0.0 0.0.0.0 87.186.225.92 254
ip route 10.10.10.2 255.255.255.255 Integrated-Service-Engine0/0
!
access-list 1 permit 10.10.10.0 0.0.0.255
access-list 1 permit 192.168.2.0 0.0.0.255
!
!
!
control-plane
!
!
voice-port 0/0/0
cptone DE
station-id number 94
caller-id enable
!
voice-port 0/0/1
!
voice-port 0/0/2
!
voice-port 0/0/3
!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/4/0
auto-cut-through
signal immediate
input gain auto-control
description Music On Hold Port
!
!
mgcp profile default
!
!
dial-peer voice 4 voip
description **Outgoing VOIP**
translation-profile outgoing OUT
destination-pattern ...T
session protocol sipv2
session target dns:tel.t-online.de
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
codec g711alaw
no vad
!
dial-peer voice 10 voip
description **Incoming VOIP**
translation-profile incoming IN
session protocol sipv2
session target ipv4:10.10.10.1
incoming called-number 123456
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
!
dial-peer voice 20 voip
description **Mailbox**
destination-pattern 99.
session protocol sipv2
session target ipv4:10.10.10.2
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 19 pots
destination-pattern 94
port 0/0/0
no sip-register
!
!
presence
presence call-list
!
sip-ua
credentials username 123456 password ***password*** realm tel.t-online.de
authentication username ***user*** password ***password*** realm tel.t-online.de
nat symmetric role active
nat symmetric check-media-src
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry notify 6
retry register 5
retry options 10
timers connect 100
timers notify 100
timers register 250
mwi-server ipv4:10.10.10.2 expires 3600 port 5060 transport udp
registrar 1 dns:tel.t-online.de expires 3600
sip-server dns:tel.t-online.de
connection-reuse
host-registrar
presence enable
!
!
!
telephony-service
no auto-reg-ephone
max-ephones 12
max-dn 30
ip source-address 10.10.10.1 port 2000
max-redirect 5
timeouts interdigit 2
url services http://10.10.10.2/voiceview/common/login.do
cnf-file location flash:
user-locale U1 load CME-locale-de_DE-German-8.8.2.5.tar
load 7925 CP7925G-1.4.4.3.LOADS
time-zone 23
time-format 24
date-format dd-mm-yy
voicemail 999
max-conferences 4 gain -6
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Nov 11 2013 14:07:24
!
!
!
ephone-dn 1 dual-line
number 92 no-reg primary
label IP-Mobil-1
description IP-Mobil-1
name IP-Mobil-1
allow watch
!
!
ephone-dn 2 dual-line
number 93 no-reg primary
label IP-Mobil-2
description IP-Mobil-2
name IP-Mobil-2
allow watch
!
!
!
ephone 1
device-security-mode none
description IP-Mobil-1
mac-address E05F.B9BC.74CE
max-calls-per-button 4
type 7925
button 1:1
!
!
!
ephone 2
device-security-mode none
description IP-Mobil-2
mac-address E05F.B9BB.F4D9
max-calls-per-button 4
type 7925
button 1:2
!
!
!
line con 0
no modem enable
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
line vty 0 4
access-class 102 in
privilege level 15
transport input all
transport output all
line vty 5 15
access-class 101 in
exec-timeout 30 0
privilege level 15
transport input all
transport output all
!
!
end
***************************************************************************************
**********debug (SIP Phone calls Unity Express Mailbox System) ***********
*Nov 11 19:29:14.642: //-1/6226DD5D8211/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabledimageattr parse payload numtok not foundimageattr payload found, specific is 255
*Nov 11 19:29:14.642: //556/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:29:14.642: //556/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 2 = 10.10.10.1
*Nov 11 19:29:14.642: //556/6226DD5D8211/SIP/Media/sipSPIDoMediaNegotiation: DO_Media_Negotiation: video m-line detected.
*Nov 11 19:29:14.642: //556/6226DD5D8211/SIP/Media/sipSPIUpdCallWithSdpInfo:
Preferred Codec : g711alaw, bytes :160
Preferred DTMF relay : inband-voice
Preferred NTE payload : 101
Early Media : No
Delayed Media : No
Bridge Done : No
New Media : No
DSP DNLD Reqd : No
*Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPISelectCodecVersion: g729r8 flavor of g729 codec will be used
*Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPISelectCodecVersion: g722-64 flavor of G722 codec will be used
*Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPIDisplayStreamInfo:
Stream type : voice-only
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 556
Peer Callid : -1
RTP/SRTP Negotiated : 8
Negotiated Codec : g711alaw, bytes :160
Nego. Codec payload : 8 (tx), 8 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.10.10.1]:0
Media Dest Addr/Port : [10.10.10.12]:17166
*Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPIDisplayStreamInfo:
Stream type : video
Media line : 2
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 557
Peer Callid : -1
RTP/SRTP Negotiated : 8
Negotiated Codec : h264, bytes
Nego. Codec payload : 97 (tx), 119 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.10.10.1]:0
Media Dest Addr/Port : [10.10.10.12]:28736
*Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : voice-only
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 556
Negotiated Codec : g711alaw, bytes :160
Nego. Codec payload : 8 (tx), 8 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.10.10.1]:0
Media Dest Addr/Port : [10.10.10.12]:17166
*Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 2 = 10.10.10.1
*Nov 11 19:29:14.646: //556/6226DD5D8211/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : video
Media line : 2
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 557
Negotiated Codec : h264, bytes
Nego. Codec payload : 97 (tx), 119 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.10.10.1]:0
Media Dest Addr/Port : [10.10.10.12]:28736
*Nov 11 19:29:14.646: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16662 for stream 1
*Nov 11 19:29:14.646: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16664 for stream 2
*Nov 11 19:29:14.650: //556/6226DD5D8211/SIP/Media/sipSPIDisplayStreamInfo:
Stream type : voice-only
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 556
Peer Callid : -1
RTP/SRTP Negotiated : 8
Negotiated Codec : g711alaw, bytes :160
Nego. Codec payload : 8 (tx), 8 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.10.10.1]:16662
Media Dest Addr/Port : [10.10.10.12]:17166
*Nov 11 19:29:14.650: //556/6226DD5D8211/SIP/Media/sipSPIDisplayStreamInfo:
Stream type : video
Media line : 2
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 557
Peer Callid : -1
RTP/SRTP Negotiated : 8
Negotiated Codec : h264, bytes
Nego. Codec payload : 97 (tx), 119 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.10.10.1]:16664
Media Dest Addr/Port : [10.10.10.12]:28736
*Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
*Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:29:14.682: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16666 for stream 1
*Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101
*Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 2 = 10.10.10.1
*Nov 11 19:29:14.682: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16668 for stream 2
*Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPI_ipip_UpdateSDPVideoMediaPayload: update ccb->pld.negotiated_bandwidth with current_max_bit_rate (100bps):10000
*Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPI_ipip_UpdateSDPVideoMediaPayload: VIDEO m-line 2, set ccb negotiated_bandwidth (kbps):1000; current_max_bit_rate:10000, local_bandwidth_req (kbps):0, remote_bandwidth_req (kbps):0
*Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_IDLE
*Nov 11 19:29:14.682: //558/6226DD5D8211/SIP/Media/sipSPIProcessRtpSessions: No active streams.
*Nov 11 19:29:14.686: //558/6226DD5D8211/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING
*Nov 11 19:29:14.686: //558/6226DD5D8211/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 558) to the VOIP RTP library
*Nov 11 19:29:14.686: //558/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:29:14.686: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
*Nov 11 19:29:14.690: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 10.10.10.1, lport = 16666, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE
src_callid = 558, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY
media_ip_addr = - , vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0
*Nov 11 19:29:14.690: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
*Nov 11 19:29:14.690: //558/6226DD5D8211/SIP/Media/sipSPICreateRtpSession: stun is disabled
*Nov 11 19:29:14.690: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp passthru enabled
*Nov 11 19:29:14.690: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=558
*Nov 11 19:29:14.806: //558/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:29:14.810: //558/6226DD5D8211/SIP/Media/sipSPIReplaceSDP: Main stream got changed & it's Flow Around
*Nov 11 19:29:14.810: //558/6226DD5D8211/SIP/Media/sipSPIUpdCallWithSdpInfo:
Preferred Codec : g711alaw, 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
*Nov 11 19:29:14.810: //558/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:29:14.810: //558/6226DD5D8211/SIP/Media/sipSPIDisplayStreamInfo:
Stream type : voice+dtmf
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 558
Peer Callid : -1
RTP/SRTP Negotiated : 8
Negotiated Codec : g711ulaw, bytes :160
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 : [10.10.10.1]:16666
Media Dest Addr/Port : [10.10.10.2]:21046
*Nov 11 19:29:14.810: //556/6226DD5D8211/SIP/Media/sipSPISetStreamInfo: 0 Active Streams
*Nov 11 19:29:14.810: //556/6226DD5D8211/SIP/Media/sipSPISetStreamInfo: Number of active streams is zero (0)!
*Nov 11 19:29:14.810: //556/6226DD5D8211/SIP/Media/sipSPISetStreamInfo:
caps.stream_count=0,caps.stream[0].stream_type=0xFFFF, caps.stream_list.xmitFunc=
*Nov 11 19:29:14.810: //556/6226DD5D8211/SIP/Media/sipSPISetStreamInfo: ??unknown??, caps.stream_list.context=
*Nov 11 19:29:14.810: //556/6226DD5D8211/SIP/Media/sipSPISetStreamInfo: 0x0 (gccb)
*Nov 11 19:29:14.810: //558/6226DD5D8211/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : voice+dtmf
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 558
Negotiated Codec : g711ulaw, bytes :160
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 : [10.10.10.1]:16666
Media Dest Addr/Port : [10.10.10.2]:21046
*Nov 11 19:29:14.814: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8B7786D0
*Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING
*Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice-only (callid 556) to the VOIP RTP library
*Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
*Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 10.10.10.1, lport = 16662, raddr = 10.10.10.12, rport=17166, do_rtcp=TRUE
src_callid = 556, dest_callid = 558, stream type = voice-only, stream direction = SENDRECV
media_ip_addr = 10.10.10.12, vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0
*Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
*Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPICreateRtpSession: stun is disabled
*Nov 11 19:29:14.818: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp passthru enabled
*Nov 11 19:29:14.818: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=556
*Nov 11 19:29:14.818: //556/6226DD5D8211/SIP/Media/sipSPIGetNewLocalMediaDirection:
New Remote Media Direction = SENDRECV
Present Local Media Direction = SENDRECV
New Local Media Direction = SENDRECV
retVal = 0
*Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING
*Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 558) to the VOIP RTP library
*Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
*Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 10.10.10.1, lport = 16666, raddr = 10.10.10.2, rport=21046, do_rtcp=TRUE
src_callid = 558, dest_callid = 556, stream type = voice+dtmf, stream direction = SENDRECV
media_ip_addr = 10.10.10.2, vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0
*Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtcpSession: RTP session already created - update
*Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8B7786D0
*Nov 11 19:29:14.822: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=558
*Nov 11 19:29:14.822: //558/6226DD5D8211/SIP/Media/sipSPIGetNewLocalMediaDirection:
New Remote Media Direction = SENDRECV
Present Local Media Direction = SENDRECV
New Local Media Direction = SENDRECV
retVal = 0
*Nov 11 19:29:14.850: //558/6226DD5D8211/SIP/Media/act_update_nat_media: NEW: rtp session to be updated with the new src info, old addr:port [10.10.10.2]:21046 , newaddr:port [10.10.10.2]:21046
*Nov 11 19:29:14.850: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8B7786D0
*Nov 11 19:29:14.898: //556/6226DD5D8211/SIP/Media/act_update_nat_media: NEW: rtp session to be updated with the new src info, old addr:port [10.10.10.12]:17166 , newaddr:port [10.10.10.12]:17166
*Nov 11 19:29:14.902: //556/6226DD5D8211/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6F25F8
*Nov 11 19:29:34.322: //556/6226DD5D8211/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6F25F8
*Nov 11 19:29:34.322: //558/6226DD5D8211/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8B7786D0
*Nov 11 19:29:34.326: //556/6226DD5D8211/SIP/Media/sipSPIHandleDestroyRtpSession: stream:8D6F25F8
*Nov 11 19:29:34.338: //558/6226DD5D8211/SIP/Media/sipSPIHandleDestroyRtpSession: stream:8B7786D0
*Nov 11 19:31:36.042: //565/B61D0194821B/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
*Nov 11 19:31:36.042: //565/B61D0194821B/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:31:36.042: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16670 for stream 1
*Nov 11 19:31:36.042: //565/B61D0194821B/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101
*Nov 11 19:31:36.042: //565/B61D0194821B/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_IDLE
*Nov 11 19:31:36.042: //565/B61D0194821B/SIP/Media/sipSPIProcessRtpSessions: No active streams.
*Nov 11 19:31:36.046: //565/B61D0194821B/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING
*Nov 11 19:31:36.046: //565/B61D0194821B/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 565) to the VOIP RTP library
*Nov 11 19:31:36.046: //565/B61D0194821B/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:31:36.046: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
*Nov 11 19:31:36.046: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 10.10.10.1, lport = 16670, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE
src_callid = 565, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY
media_ip_addr = - , vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0
*Nov 11 19:31:36.046: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
*Nov 11 19:31:36.046: //565/B61D0194821B/SIP/Media/sipSPICreateRtpSession: stun is disabled
*Nov 11 19:31:36.046: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp passthru enabled
*Nov 11 19:31:36.046: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=565
*Nov 11 19:31:36.170: //565/B61D0194821B/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:31:36.170: //565/B61D0194821B/SIP/Media/sipSPIReplaceSDP: Main stream got changed & it's Flow Around
*Nov 11 19:31:36.170: //565/B61D0194821B/SIP/Media/sipSPIUpdCallWithSdpInfo:
Preferred Codec : g711alaw, 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
*Nov 11 19:31:36.174: //565/B61D0194821B/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:31:36.174: //565/B61D0194821B/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : voice+dtmf
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 565
Negotiated Codec : g711ulaw, bytes :160
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 : [10.10.10.1]:16670
Media Dest Addr/Port : [10.10.10.2]:21114
*Nov 11 19:31:36.174: //565/B61D0194821B/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING
*Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 565) to the VOIP RTP library
*Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
*Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 10.10.10.1, lport = 16670, raddr = 10.10.10.2, rport=21114, do_rtcp=TRUE
src_callid = 565, dest_callid = 564, stream type = voice+dtmf, stream direction = SENDRECV
media_ip_addr = 10.10.10.2, vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0
*Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtcpSession: RTP session already created - update
*Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6C478C
*Nov 11 19:31:36.178: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=565
*Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPIGetNewLocalMediaDirection:
New Remote Media Direction = SENDRECV
Present Local Media Direction = SENDRECV
New Local Media Direction = SENDRECV
retVal = 0
*Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPISetStreamInfo: 1 Active Streams
*Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPISetStreamInfo: Adding stream type (voice+dtmf) from media
line 1 codec g711ulaw
*Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPISetStreamInfo:
caps.stream_count=1,caps.stream[0].stream_type=0x3, caps.stream_list.xmitFunc=
*Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPISetStreamInfo: voip_rtp_xmit, caps.stream_list.context=
*Nov 11 19:31:36.178: //565/B61D0194821B/SIP/Media/sipSPISetStreamInfo: 0x8D6F1880 (gccb)
*Nov 11 19:31:36.218: //565/B61D0194821B/SIP/Media/act_update_nat_media: NEW: rtp session to be updated with the new src info, old addr:port [10.10.10.2]:21114 , newaddr:port [10.10.10.2]:21114
*Nov 11 19:31:36.218: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6C478C
*Nov 11 19:31:43.474: //565/B61D0194821B/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6C478C
*Nov 11 19:31:43.490: //565/B61D0194821B/SIP/Media/sipSPIHandleDestroyRtpSession: stream:8D6C478C
*Nov 11 19:45:41.206: //-1/AE3093EE8227/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabledimageattr parse payload numtok not foundimageattr payload found, specific is 255
*Nov 11 19:45:41.206: //589/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 2 = 10.10.10.1
*Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPIDoMediaNegotiation: DO_Media_Negotiation: video m-line detected.
*Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPIUpdCallWithSdpInfo:
Preferred Codec : g711alaw, bytes :160
Preferred DTMF relay : inband-voice
Preferred NTE payload : 101
Early Media : No
Delayed Media : No
Bridge Done : No
New Media : No
DSP DNLD Reqd : No
*Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPISelectCodecVersion: g729r8 flavor of g729 codec will be used
*Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPISelectCodecVersion: g722-64 flavor of G722 codec will be used
*Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPIDisplayStreamInfo:
Stream type : voice-only
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 589
Peer Callid : -1
RTP/SRTP Negotiated : 8
Negotiated Codec : g711alaw, bytes :160
Nego. Codec payload : 8 (tx), 8 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.10.10.1]:0
Media Dest Addr/Port : [10.10.10.12]:20200
*Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPIDisplayStreamInfo:
Stream type : video
Media line : 2
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 590
Peer Callid : -1
RTP/SRTP Negotiated : 8
Negotiated Codec : h264, bytes
Nego. Codec payload : 97 (tx), 119 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.10.10.1]:0
Media Dest Addr/Port : [10.10.10.12]:22496
*Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : voice-only
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 589
Negotiated Codec : g711alaw, bytes :160
Nego. Codec payload : 8 (tx), 8 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.10.10.1]:0
Media Dest Addr/Port : [10.10.10.12]:20200
*Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 2 = 10.10.10.1
*Nov 11 19:45:41.210: //589/AE3093EE8227/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : video
Media line : 2
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 590
Negotiated Codec : h264, bytes
Nego. Codec payload : 97 (tx), 119 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.10.10.1]:0
Media Dest Addr/Port : [10.10.10.12]:22496
*Nov 11 19:45:41.210: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16672 for stream 1
*Nov 11 19:45:41.210: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16674 for stream 2
*Nov 11 19:45:41.214: //589/AE3093EE8227/SIP/Media/sipSPIDisplayStreamInfo:
Stream type : voice-only
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 589
Peer Callid : -1
RTP/SRTP Negotiated : 8
Negotiated Codec : g711alaw, bytes :160
Nego. Codec payload : 8 (tx), 8 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.10.10.1]:16672
Media Dest Addr/Port : [10.10.10.12]:20200
*Nov 11 19:45:41.214: //589/AE3093EE8227/SIP/Media/sipSPIDisplayStreamInfo:
Stream type : video
Media line : 2
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 590
Peer Callid : -1
RTP/SRTP Negotiated : 8
Negotiated Codec : h264, bytes
Nego. Codec payload : 97 (tx), 119 (rx)
Negotiated DTMF relay : inband-voice
Negotiated NTE payload : 0 (tx), 0 (rx)
Negotiated CN payload : 0
Media Srce Addr/Port : [10.10.10.1]:16674
Media Dest Addr/Port : [10.10.10.12]:22496
*Nov 11 19:45:41.230: //591/AE3093EE8227/SIP/Media/sipSPICopyStunConfigFromPeerToCCB: Firewall traversal is not enabled
*Nov 11 19:45:41.230: //591/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:45:41.230: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16676 for stream 1
*Nov 11 19:45:41.230: //591/AE3093EE8227/SIP/Media/sipSPIAddSDPMediaPayload: Preferred method of dtmf relay is: 6, with payload: 101
*Nov 11 19:45:41.234: //591/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 2 = 10.10.10.1
*Nov 11 19:45:41.234: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIReserveRtpPort: reserved port 16678 for stream 2
*Nov 11 19:45:41.234: //591/AE3093EE8227/SIP/Media/sipSPI_ipip_UpdateSDPVideoMediaPayload: update ccb->pld.negotiated_bandwidth with current_max_bit_rate (100bps):10000
*Nov 11 19:45:41.234: //591/AE3093EE8227/SIP/Media/sipSPI_ipip_UpdateSDPVideoMediaPayload: VIDEO m-line 2, set ccb negotiated_bandwidth (kbps):1000; current_max_bit_rate:10000, local_bandwidth_req (kbps):0, remote_bandwidth_req (kbps):0
*Nov 11 19:45:41.234: //591/AE3093EE8227/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_IDLE
*Nov 11 19:45:41.234: //591/AE3093EE8227/SIP/Media/sipSPIProcessRtpSessions: No active streams.
*Nov 11 19:45:41.238: //591/AE3093EE8227/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING
*Nov 11 19:45:41.238: //591/AE3093EE8227/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 591) to the VOIP RTP library
*Nov 11 19:45:41.238: //591/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:45:41.238: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
*Nov 11 19:45:41.238: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 10.10.10.1, lport = 16676, raddr = 0.0.0.0, rport=0, do_rtcp=FALSE
src_callid = 591, dest_callid = -1, stream type = voice+dtmf, stream direction = RECVONLY
media_ip_addr = - , vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0
*Nov 11 19:45:41.238: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
*Nov 11 19:45:41.238: //591/AE3093EE8227/SIP/Media/sipSPICreateRtpSession: stun is disabled
*Nov 11 19:45:41.238: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp passthru enabled
*Nov 11 19:45:41.238: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=591
*Nov 11 19:45:41.350: //591/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:45:41.350: //591/AE3093EE8227/SIP/Media/sipSPIReplaceSDP: Main stream got changed & it's Flow Around
*Nov 11 19:45:41.350: //591/AE3093EE8227/SIP/Media/sipSPIUpdCallWithSdpInfo:
Preferred Codec : g711alaw, 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
*Nov 11 19:45:41.350: //591/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:45:41.350: //591/AE3093EE8227/SIP/Media/sipSPIDisplayStreamInfo:
Stream type : voice+dtmf
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 591
Peer Callid : -1
RTP/SRTP Negotiated : 8
Negotiated Codec : g711ulaw, bytes :160
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 : [10.10.10.1]:16676
Media Dest Addr/Port : [10.10.10.2]:20670
*Nov 11 19:45:41.354: //589/AE3093EE8227/SIP/Media/sipSPISetStreamInfo: 0 Active Streams
*Nov 11 19:45:41.354: //589/AE3093EE8227/SIP/Media/sipSPISetStreamInfo: Number of active streams is zero (0)!
*Nov 11 19:45:41.354: //589/AE3093EE8227/SIP/Media/sipSPISetStreamInfo:
caps.stream_count=0,caps.stream[0].stream_type=0xFFFF, caps.stream_list.xmitFunc=
*Nov 11 19:45:41.354: //589/AE3093EE8227/SIP/Media/sipSPISetStreamInfo: ??unknown??, caps.stream_list.context=
*Nov 11 19:45:41.354: //589/AE3093EE8227/SIP/Media/sipSPISetStreamInfo: 0x0 (gccb)
*Nov 11 19:45:41.354: //591/AE3093EE8227/SIP/Media/sipSPIUpdCallWithSdpInfo:
Stream type : voice+dtmf
Media line : 1
State : STREAM_ADDING (2)
Stream address type : 1
Callid : 591
Negotiated Codec : g711ulaw, bytes :160
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 : [10.10.10.1]:16676
Media Dest Addr/Port : [10.10.10.2]:20670
*Nov 11 19:45:41.354: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6E4A28
*Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING
*Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice-only (callid 589) to the VOIP RTP library
*Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
*Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 10.10.10.1, lport = 16672, raddr = 10.10.10.12, rport=20200, do_rtcp=TRUE
src_callid = 589, dest_callid = 591, stream type = voice-only, stream direction = SENDRECV
media_ip_addr = 10.10.10.12, vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0
*Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: No rtp session, creating a new one
*Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPICreateRtpSession: stun is disabled
*Nov 11 19:45:41.362: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: Voice quality monitoring is not enabled for this RTP session due to sdp passthru enabled
*Nov 11 19:45:41.362: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=589
*Nov 11 19:45:41.362: //589/AE3093EE8227/SIP/Media/sipSPIGetNewLocalMediaDirection:
New Remote Media Direction = SENDRECV
Present Local Media Direction = SENDRECV
New Local Media Direction = SENDRECV
retVal = 0
*Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPIProcessRtpSessions: Processing stream state = STREAM_ADDING
*Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPIAddStream: Adding stream 1 of type voice+dtmf (callid 591) to the VOIP RTP library
*Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 10.10.10.1
*Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: sipSPIUpdateRtcpSession for m-line 1
*Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: rtcp_session info
laddr = 10.10.10.1, lport = 16676, raddr = 10.10.10.2, rport=20670, do_rtcp=TRUE
src_callid = 591, dest_callid = 589, stream type = voice+dtmf, stream direction = SENDRECV
media_ip_addr = 10.10.10.2, vrf tableid = 0 media_addr_type = 1 negotiated_bandwidth (kbps) = 0
*Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtcpSession: RTP session already created - update
*Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6E4A28
*Nov 11 19:45:41.362: //-1/xxxxxxxxxxxx/SIP/Media/sipSPIUpdateRtcpSession: VQM: gccb=0x0, gccb->callId=0, ccb->ccCallID=591
*Nov 11 19:45:41.362: //591/AE3093EE8227/SIP/Media/sipSPIGetNewLocalMediaDirection:
New Remote Media Direction = SENDRECV
Present Local Media Direction = SENDRECV
New Local Media Direction = SENDRECV
retVal = 0
*Nov 11 19:45:41.386: //591/AE3093EE8227/SIP/Media/act_update_nat_media: NEW: rtp session to be updated with the new src info, old addr:port [10.10.10.2]:20670 , newaddr:port [10.10.10.2]:20670
*Nov 11 19:45:41.386: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6E4A28
*Nov 11 19:45:41.438: //589/AE3093EE8227/SIP/Media/act_update_nat_media: NEW: rtp session to be updated with the new src info, old addr:port [10.10.10.12]:20200 , newaddr:port [10.10.10.12]:20200
*Nov 11 19:45:41.438: //589/AE3093EE8227/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8EE45620
*Nov 11 19:45:47.322: //589/AE3093EE8227/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8EE45620
*Nov 11 19:45:47.322: //591/AE3093EE8227/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:8D6E4A28
*Nov 11 19:45:47.326: //589/AE3093EE8227/SIP/Media/sipSPIHandleDestroyRtpSession: stream:8EE45620
*Nov 11 19:45:47.338: //591/AE3093EE8227/SIP/Media/sipSPIHandleDestroyRtpSession: stream:8D6E4A28
Solved! Go to Solution.
11-11-2013 12:53 PM
can you try this..
voice register pool 1
dtmf-relay rtp-nte
If it doesnt work pls send debug ccsip messages.
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"
11-11-2013 12:53 PM
can you try this..
voice register pool 1
dtmf-relay rtp-nte
If it doesnt work pls send debug ccsip messages.
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"
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