Hi there
I'm facing the following problem:
CME 8.6 (ipvoice-mz.151-4.M10.bin) with a SIP trunk box behind an ASA (8.2.5).
CME -> Patton-SN5300 SIP provider box -> Cisco ASA -> SIP-Provider.
Problem:
I have no audio when an external incoming call to an internal number with CFA to an external number through the SIP trunk. Signalling is fine, it's ringing on the final number, but no audio.
External IN/OUT SIP trunk calls works fine as well consult transfer from an external incoming call to an external destination.
What I have already tried:
With "media flow-through" I see the RTP stream (show voip rtp connection). With "media flow-around" ther is no RTP stream shown.
SIP inspection is disabled on the ASA. Tried fixing the codec -> still no audio.
Any ideas? NAT'ing on the ASA?
Kind regards,
Norbert
Here the config:
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
redirect ip2ip
fax-relay ans-disable
h323
h245 caps mode restricted
sip
asserted-id ppi
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g729r8
codec preference 3 g729br8
codec preference 4 g711ulaw
codec preference 5 g723r63
codec preference 6 g723ar63
voice translation-rule 40
rule 1 /\(.*\)/ /0\1/
!
voice translation-rule 190
rule 1 /^0\(.*\)/ /\1/
!
voice translation-rule 191 <- internal to external map
rule 1 /^201/ /xxxxxxxx19/
rule 2 /^202/ /xxxxxxxx18/
rule 15 /^0\(.*\)/ /\1/
!
voice translation-rule 192 <- external to internal map
rule 2 /^xxxxxxxx19/ /201/
rule 3 /^xxxxxxxx18/ /202/
voice translation-profile TP_IN_SIP
translate calling 40
translate called 192
!
voice translation-profile TP_OUT_SIP
translate calling 191
translate called 190
!
dial-peer voice 2001 voip
description *** SIP-TRUNK (OUT) ***
translation-profile incoming TP_IN_SIP
translation-profile outgoing TP_OUT_SIP
destination-pattern 0.T
session protocol sipv2
session target ipv4:192.168.1.10:5062
session transport udp
incoming called-number .
voice-class codec 1
dtmf-relay rtp-nte
no vad
sip-ua
credentials username xxxxxx password 7 xxxxxx realm 192.168.1.10:5062 <- IP Patton box
keepalive target ipv4:192.168.1.10:5062
authentication username xxxx password 7 xxx
retry invite 2
retry response 2
retry bye 2
retry register 2
retry options 1
registrar ipv4:192.168.1.10:5062 expires 60
show voip rtp connections:
No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 2201 2202 18182 6542 192.168.1.1 192.168.1.10
2 2202 2201 17354 6544 192.168.1.1 192.168.1.10
Found 2 active RTP connections
show sip-ua calls:
Total SIP call legs:2, User Agent Client:1, User Agent Server:1
SIP UAC CALL INFO
Call 1
SIP Call ID : B537765A-F4F311E4-83AAF7BA-53BC2227@192.168.1.1
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : xxxxxxxx22
Called Number : xxxxxxxx75
Bit Flags : 0xC04018 0x50000100 0x0
CC Call ID : 2207
Source IP Address (Sig ): 192.168.1.1
Destn SIP Req Addr:Port : [192.168.1.10]:5062
Destn SIP Resp Addr:Port: [192.168.1.10]:5062
Destination Name : 192.168.1.10
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 2207
Stream Type : voice+dtmf (1)
Stream Media Addr Type : 1
Negotiated Codec : g711alaw (160 bytes)
Codec Payload Type : 8
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 101
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [192.168.1.1]:16670
Media Dest IP Addr:Port : [192.168.1.10]:6552
Options-Ping ENABLED:NO ACTIVE:NO
Number of SIP User Agent Client(UAC) calls: 1
SIP UAS CALL INFO
Call 1
SIP Call ID : 7f04a2619ca33575
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : xxxxxxxx22
Called Number : xxxxxxxx18
Bit Flags : 0xC0401E 0x18000100 0x4
CC Call ID : 2206
Source IP Address (Sig ): 192.168.1.1
Destn SIP Req Addr:Port : [192.168.1.10]:5062
Destn SIP Resp Addr:Port: [192.168.1.10]:5062
Destination Name : 192.168.1.10
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 2206
Stream Type : voice+dtmf (0)
Stream Media Addr Type : 1
Negotiated Codec : g711alaw (160 bytes)
Codec Payload Type : 8
Negotiated Dtmf-relay : rtp-nte
Dtmf-relay Payload Type : 99
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [192.168.1.1]:19172
Media Dest IP Addr:Port : [192.168.1.10]:6550
