Hi,
I am experiencing one-way audio from:
SIP Device ----> Asterisk -----> Cisco 7200
The Cisco 7200 has a VXE+ card that will allow you to do SIP. I can pass
audio from SIP Device to Asterisk through the Cisco 7200 to the other end,
but the Cisco 7200 does not return any audio back to the SIP Device or
Asterisk, it seems. I have tried upgrading to 12.3T IOS version, but no luck.
Has anyone else experienced this problem? My configuration and SIP debug is
posted below. Asterisk server in SIP debug is xxx.xxx.xxx.xxx and Cisco 7200
is yyy.yyy.yyy.yyy. Thanks!
IOS Config:
Building configuration...
Current configuration : 3362 bytes
!
! Last configuration change at 21:04:59 GMT Tue Nov 30 2004
!
version 12.2
service timestamps debug uptime
service timestamps log uptime
service password-encryption
!
hostname 7206_SIP
!
card type t1 1
card type t1 2
enable secret 5 $XXXXXXXXXXXXXXXXXXX
enable password 7 XXXXXXXXXXXXXXXXX
!
clock timezone GMT 0
dspint DSPfarm1/0
!
ip subnet-zero
no ip routing
!
!
ip name-server XXX.XXX.XXX.XXX
!
no ip cef
isdn switch-type primary-ni
call rsvp-sync
voice call send-alert
voice rtp send-recv
!
voice service voip
!
!
!
!
!
!
controller T1 1/0
framing esf
linecode b8zs
cablelength long 0db
pri-group timeslots 1-24
!
controller T1 1/1
framing esf
linecode b8zs
cablelength long 0db
pri-group timeslots 1-24
!
controller T1 2/0
framing esf
service-type ccs-voice
linecode b8zs
cablelength long 0db
pri-group timeslots 1-24
!
controller T1 2/1
framing esf
service-type ccs-voice
linecode b8zs
cablelength long 0db
pri-group timeslots 1-24
!
controller T1 2/2
framing esf
service-type ccs-voice
linecode b8zs
cablelength long 0db
pri-group timeslots 1-24
!
controller T1 2/3
framing esf
service-type ccs-voice
linecode b8zs
cablelength long 0db
!
controller T1 2/4
framing esf
service-type ccs-voice
linecode b8zs
cablelength long 0db
!
controller T1 2/5
framing esf
service-type ccs-voice
linecode b8zs
cablelength long 0db
!
controller T1 2/6
framing esf
service-type ccs-voice
linecode b8zs
cablelength long 0db
!
controller T1 2/7
framing esf
service-type ccs-voice
linecode b8zs
cablelength long 0db
!
!
!
interface FastEthernet0/0
ip address XXX.XXX.XXX.XXX
ip access-group 101 in
no ip route-cache
no ip mroute-cache
duplex full
!
interface Serial1/0:23
no ip address
no logging event link-status
isdn switch-type primary-ni
isdn incoming-voice voice
no cdp enable
!
interface Serial1/1:23
no ip address
no logging event link-status
isdn switch-type primary-ni
isdn incoming-voice voice
no cdp enable
!
interface Serial2/0:23
no ip address
no logging event link-status
isdn switch-type primary-ni
isdn incoming-voice voice
no cdp enable
!
interface Serial2/1:23
no ip address
no logging event link-status
isdn switch-type primary-ni
isdn incoming-voice voice
no cdp enable
!
interface Serial2/2:23
no ip address
no logging event link-status
isdn switch-type primary-ni
isdn incoming-voice voice
no cdp enable
!
ip classless
no ip http server
ip pim bidir-enable
!
access-list 101 permit ip any any
snmp-server community XXX.XXX.XXX.XXX
!
!
trunk group 1
hunt-scheme round-robin
!
voice-port 1/0:23
!
voice-port 1/1:23
!
voice-port 2/0:23
!
voice-port 2/1:23
!
voice-port 2/2:23
!
dial-peer cor custom
!
!
!
dial-peer voice 100 voip
destination-pattern .T
session protocol sipv2
session target ipv4: XXX.XXX.XXX.XXX
dtmf-relay h245-signal h245-alphanumeric
no vad
!
dial-peer voice 10 pots
destination-pattern .T
port 1/0:23
!
dial-peer voice 11 pots
destination-pattern .T
port 1/1:23
!
dial-peer voice 20 pots
destination-pattern .T
port 2/0:23
!
gateway
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4: XXX.XXX.XXX.XXX
!
!
gatekeeper
shutdown
!
!
line con 0
line aux 0
line vty 0 4
password 7 XXXXXXXXXXXXXX
login
!
ntp clock-period 17179879
ntp server XXX.XXX.XXX.XXX
end
SIP Debug:
Sip read:
INVITE sip:1001@xxx.xxx.xxx.xxx;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.200.169:5060
From: 3213084003
<sip:3213084003@xxx.xxx.xxx.xxx;user=phone>;tag=2785115378
To: <sip:1001@xxx.xxx.xxx.xxx;user=phone>
Call-ID: 1801199744@192.168.200.169
CSeq: 1 INVITE
Contact: 3213084003
<sip:3213084003@192.168.200.169:5060;user=phone;transport=udp>
User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A)
Expires: 300
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 280
Content-Type: application/sdp
v=0
o=3213084003 1311 1311 IN IP4 192.168.200.169
s=ATA186 Call
c=IN IP4 192.168.200.169
t=0 0
m=audio 16384 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
12 headers, 12 lines
Using latest request as basis request
Found user '3213084003'
Looking for 1001 in default
list_route: hop:
<sip:3213084003@192.168.200.169:5060;user=phone;transport=udp>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.200.169:5060;received=209.114.219.98;rport=5060
From: 3213084003
<sip:3213084003@xxx.xxx.xxx.xxx;user=phone>;tag=2785115378
To: <sip:1001@xxx.xxx.xxx.xxx;user=phone>;tag=as0f161add
Call-ID: 1801199744@192.168.200.169
CSeq: 1 INVITE
User-Agent: HCC Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1001@xxx.xxx.xxx.xxx>
Content-Length: 0
to 209.114.219.98:5060
-- Executing MYSQL("SIP/3213084003-cab8", "Connect connid
localhost
asterisk longpoint asterisk") in new stack
-- Executing MYSQL("SIP/3213084003-cab8", "Query resultid 7
SELECT
scriptname from mac2pin where userid=3213084003") in new stack
-- Executing MYSQL("SIP/3213084003-cab8", "Fetch fetchid 8
AGIScript") in
new stack
Jan 4 17:49:50 WARNING[27531]: app_addon_sql_mysql.c:318 aMYSQL_fetch:
ast_MYSQL_fetch: numFields=1
-- Executing GotoIf("SIP/3213084003-cab8", "0?5:7") in
new stack
-- Goto (default,1001,7)
-- Executing AGI("SIP/3213084003-cab8",
"HCC_TEST.agi|1001") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/HCC_TEST.agi
HCC_TEST.agi|1001: Dialing cisco SIP for Mark
-- AGI Script Executing Application: (Dial) Options:
(SIP/003214093773@3213084999)
-- Called 003214093773@3213084999
-- SIP/3213084999-71e7 is making progress passing it to
SIP/3213084003-cab8
We're at xxx.xxx.xxx.xxx port 17606
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with preferred capability 0x2 (gsm)
Answering with preferred capability 0x100 (g729)
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.169:5060;received=209.114.219.98;rport=5060
From: 3213084003
<sip:3213084003@xxx.xxx.xxx.xxx;user=phone>;tag=2785115378
To: <sip:1001@xxx.xxx.xxx.xxx;user=phone>;tag=as0f161add
Call-ID: 1801199744@192.168.200.169
CSeq: 1 INVITE
User-Agent: HCC Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1001@xxx.xxx.xxx.xxx>
Content-Type: application/sdp
Content-Length: 293
v=0
o=root 27531 27532 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 17606 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 209.114.219.98:5060
-- Attempting native bridge of SIP/3213084003-cab8 and SIP/3213084999-71e7
asterisk*CLI>
Sip read:
ACK sip:1001@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.200.169:5060
From: 3213084003
<sip:3213084003@xxx.xxx.xxx.xxx;user=phone>;tag=2785115378
To: <sip:1001@xxx.xxx.xxx.xxx;user=phone>;tag=as0f161add
Call-ID: 1801199744@192.168.200.169
CSeq: 1 ACK
User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A)
Content-Length: 0
8 headers, 0 lines
set_destination: Parsing
<sip:3213084003@192.168.200.169:5060;user=phone;transport=udp> for
address/port to send to
set_destination: set destination to 192.168.200.169, port 5060
We're at xxx.xxx.xxx.xxx port 17606
Answering with preferred capability 0x100 (g729)
Answering with non-codec capability 0x1 (telephone-event)
11 headers, 10 lines
Reliably Transmitting:
INVITE sip:3213084003@192.168.200.169:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK439dee7d;rport
From: <sip:1001@xxx.xxx.xxx.xxx;user=phone>;tag=as0f161add
To: 3213084003 <sip:3213084003@xxx.xxx.xxx.xxx;user=phone>;tag=2785115378
Contact: <sip:1001@xxx.xxx.xxx.xxx>
Call-ID: 1801199744@192.168.200.169
CSeq: 102 INVITE
User-Agent: HCC Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 222
v=0
o=root 27531 27533 IN IP4 yyy.yyy.yyy.yyy
s=session
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 16966 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(NAT) to 209.114.219.98:5060
asterisk*CLI>
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK439dee7d;rport
From: <sip:1001@xxx.xxx.xxx.xxx;user=phone>;tag=as0f161add
To: 3213084003 <sip:3213084003@xxx.xxx.xxx.xxx;user=phone>;tag=2785115378
Call-ID: 1801199744@192.168.200.169
CSeq: 102 INVITE
Contact: 3213084003
<sip:3213084003@192.168.200.169:5060;user=phone;transport=udp>
Server: Cisco ATA 186 v3.1.0 atasip (040211A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 228
Content-Type: application/sdp
v=0
o=3213084003 1380 1380 IN IP4 192.168.200.169
s=ATA186 Call
c=IN IP4 192.168.200.169
t=0 0
m=audio 16384 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
11 headers, 10 lines
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.200.169:16384
Found description format G729
Found description format telephone-event
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x100
(g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1
(g723)
list_route: hop:
<sip:3213084003@192.168.200.169:5060;user=phone;transport=udp>
set_destination: Parsing
<sip:3213084003@192.168.200.169:5060;user=phone;transport=udp> for
address/port to send to
set_destination: set destination to 192.168.200.169, port 5060
Transmitting:
ACK sip:3213084003@192.168.200.169:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK1de0edf4;rport
From: <sip:1001@xxx.xxx.xxx.xxx;user=phone>;tag=as0f161add
To: 3213084003 <sip:3213084003@xxx.xxx.xxx.xxx;user=phone>;tag=2785115378
Contact: <sip:1001@xxx.xxx.xxx.xxx>
Call-ID: 1801199744@192.168.200.169
CSeq: 102 ACK
User-Agent: HCC Asterisk PBX
Content-Length: 0
(NAT) to 209.114.219.98:5060
asterisk*CLI>
Sip read:
BYE sip:1001@xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 192.168.200.169:5060
From: 3213084003
<sip:3213084003@xxx.xxx.xxx.xxx;user=phone>;tag=2785115378
To: <sip:1001@xxx.xxx.xxx.xxx;user=phone>;tag=as0f161add
Call-ID: 1801199744@192.168.200.169
CSeq: 2 BYE
User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A)
Content-Length: 0
8 headers, 0 lines
Sending to 192.168.200.169 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.200.169:5060;received=209.114.219.98;rport=5060
From: 3213084003
<sip:3213084003@xxx.xxx.xxx.xxx;user=phone>;tag=2785115378
To: <sip:1001@xxx.xxx.xxx.xxx;user=phone>;tag=as0f161add
Call-ID: 1801199744@192.168.200.169
CSeq: 2 BYE
User-Agent: HCC Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1001@xxx.xxx.xxx.xxx>
Content-Length: 0
to 209.114.219.98:5060
-- AGI Script HCC_TEST.agi completed, returning 0
Destroying call '1801199744@192.168.200.169'
--
Brian Wilkins
Software Engineer
brian@hcc.net
Heritage Communications Corporation
Melbourne, FL USA 32935
321.308.4000 x33
http://www.hcc.net
