[cisco-voip] ccme sip trunk problem
baris gulten
barisgulten at gmail.com
Sun Feb 28 15:17:58 EST 2010
Hi all,
I have 2801 ccme, c2801-ipvoicek9-mz.124-24.T2.bin
Trixbox to cme calls working but when i try cme to trixbox, i getting fast
busy signal and below error.
Is there anyone resolve this issue ?
Br,
Baris
*Trixbox configs: *Allow Anonymous Inbound SIP Calls? = Yes
[ccme]
host=192.168.100.200
secret=1234
username=1200
context=from-internal
disallow=all
allow=alaw&ulaw
dtmfmode=auto
insecure=very
type=friend
qualify=yes
trixbox1*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
ccme/1200 192.168.100.200 5060 OK (10 ms)
1200/1200 192.168.100.200 D A 5060 OK (10 ms)
1050/1050 192.168.100.102 D N A 5060 OK (8 ms)
*debug ccsip error:* *Feb 28 19:50:38.935:
//-1/xxxxxxxxxxxx/SIP/Error/rtpAvpCodec_to_voipCodec: Unexpected RTP
PayloadType :255 in SDP Body
*debug ccsip messages:*
*Feb 28 20:23:08.875: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:7777 at 192.168.100.205:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK7313B5
From: <sip:1001 at 192.168.100.205 <sip%3A1001 at 192.168.100.205>>;tag=1A5CDC-2B5
To: <sip:7777 at 192.168.100.205 <sip%3A7777 at 192.168.100.205>>
Date: Sun, 28 Feb 2010 20:23:08 GMT
Call-ID: EB69A5BC-23DD11DF-808DE5CE-7FEEA234 at 192.168.100.200
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3943409814-601690591-2156455374-2146345524
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1267388588
Contact: <sip:1001 at 192.168.100.200:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 314
v=0
o=CiscoSystemsSIP-GW-UserAgent 8631 870 IN IP4 192.168.100.200
s=SIP Call
c=IN IP4 192.168.100.200
t=0 0
m=audio 17076 RTP/AVP 0 8 18 101
c=IN IP4 192.168.100.200
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
*Feb 28 20:23:08.883: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.200:5060
;branch=z9hG4bK7313B5;received=192.168.100.200
From: <sip:1001 at 192.168.100.205 <sip%3A1001 at 192.168.100.205>>;tag=1A5CDC-2B5
To: <sip:7777 at 192.168.100.205 <sip%3A7777 at 192.168.100.205>>;tag=as030f4c80
Call-ID: EB69A5BC-23DD11DF-808DE5CE-7FEEA234 at 192.168.100.200
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2a79e099"
Content-Length: 0
*Feb 28 20:23:08.887: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:7777 at 192.168.100.205:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK7313B5
From: <sip:1001 at 192.168.100.205 <sip%3A1001 at 192.168.100.205>>;tag=1A5CDC-2B5
To: <sip:7777 at 192.168.100.205 <sip%3A7777 at 192.168.100.205>>;tag=as030f4c80
Date: Sun, 28 Feb 2010 20:23:08 GMT
Call-ID: EB69A5BC-23DD11DF-808DE5CE-7FEEA234 at 192.168.100.200
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
*Feb 28 20:23:08.891: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:7777 at 192.168.100.205:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK74D8F
From: <sip:1001 at 192.168.100.205 <sip%3A1001 at 192.168.100.205>>;tag=1A5CDC-2B5
To: <sip:7777 at 192.168.100.205 <sip%3A7777 at 192.168.100.205>>
Date: Sun, 28 Feb 2010 20:23:08 GMT
Call-ID: EB69A5BC-23DD11DF-808DE5CE-7FEEA234 at 192.168.100.200
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3943409814-601690591-2156455374-2146345524
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,
NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1267388588
Contact: <sip:1001 at 192.168.100.200:5060>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="1200",realm="asterisk",uri="
sip:7777 at 192.168.100.205:5060
",response="31ad48e84f740c1d40b40668edfdb9a9",nonce="2a79e099",algorithm=MD5
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 314
v=0
o=CiscoSystemsSIP-GW-UserAgent 8631 870 IN IP4 192.168.100.200
s=SIP Call
c=IN IP4 192.168.100.200
t=0 0
m=audio 17076 RTP/AVP 0 8 18 101
c=IN IP4 192.168.100.200
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
*Feb 28 20:23:08.895: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.100.200:5060
;branch=z9hG4bK74D8F;received=192.168.100.200
From: <sip:1001 at 192.168.100.205 <sip%3A1001 at 192.168.100.205>>;tag=1A5CDC-2B5
To: <sip:7777 at 192.168.100.205 <sip%3A7777 at 192.168.100.205>>;tag=as030f4c80
Call-ID: EB69A5BC-23DD11DF-808DE5CE-7FEEA234 at 192.168.100.200
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
*Feb 28 20:23:08.903: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:7777 at 192.168.100.205:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.200:5060;branch=z9hG4bK74D8F
From: <sip:1001 at 192.168.100.205 <sip%3A1001 at 192.168.100.205>>;tag=1A5CDC-2B5
To: <sip:7777 at 192.168.100.205 <sip%3A7777 at 192.168.100.205>>;tag=as030f4c80
Date: Sun, 28 Feb 2010 20:23:08 GMT
Call-ID: EB69A5BC-23DD11DF-808DE5CE-7FEEA234 at 192.168.100.200
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
*Config:*
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
!
logging message-counter syslog
!
no aaa new-model
dot11 syslog
ip source-route
!
ip dhcp excluded-address 192.168.100.1 192.168.100.100
ip dhcp excluded-address 192.168.100.150 192.168.100.254
!
ip dhcp pool data
network 192.168.100.0 255.255.255.0
default-router 192.168.100.254
dns-server 208.67.222.222
option 150 ip 192.168.100.200
!
ip cef
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
!
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily //i try with yes
no supplementary-service sip refer //i try
with yes
fax protocol pass-through g711ulaw
h323
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
registrar server expires max 3600 min 3600
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
voice-card 0
!
archive
log config
hidekeys
!
interface FastEthernet0/0
ip address 192.168.100.200 255.255.255.0
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 192.168.100.254
!
ip http server
no ip http secure-server
ip http path flash:
!
control-plane
!
dial-peer voice 10 voip
destination-pattern 7777
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.100.205
session transport udp
incoming called-number 1...
dtmf-relay rtp-nte
no vad
!
dial-peer voice 11 voip
destination-pattern 105.
session protocol sipv2
session target ipv4:192.168.100.205
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
!
sip-ua
credentials username 1200 password 7 135445415F realm asterisk
authentication username 1200 password 7 06575D7218
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 5
timers register 250
registrar ipv4:192.168.100.205 expires 3600
sip-server ipv4:192.168.100.205
!
telephony-service
em logout 0:0 0:0 0:0
max-ephones 5
max-dn 5
ip source-address 192.168.100.200 port 2000
auto assign 1 to 5
network-locale IT
network-locale 1 IT
network-locale 2 IT
network-locale 3 IT
network-locale 4 IT
max-conferences 4 gain -6
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!
ephone-dn 1 dual-line
number 1001
!
ephone 1
no phone-ui speeddial-fastdial
no phone-ui snr
no multicast-moh
mac-address 001E.BE90.xxxx
type 7970
button 1:1
!
line con 0
login local
line aux 0
line vty 0 4
login local
!
scheduler allocate 20000 1000
end
Router#
*
*
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20100228/ee6cef95/attachment.html>
More information about the cisco-voip
mailing list