[cisco-voip] ccme sip trunk problem
Barış Gülten
barisgulten at gmail.com
Thu Mar 4 15:24:30 EST 2010
Hi all,
I coudnt success before cme and trixbox with sip trunk.
After that i try with h323 and success.
Configs below ;
CME and Trixbox2.8, h323 trunk. My purpose use trixbox as ivr.
CME ip address :10.102.19.10
Trixbox ip address :10.102.19.11
CME
voice service voip
clid network-provided
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
supplementary-service media-renegotiate
fax protocol cisco
h323
sip
!
dial-peer voice 10 voip
destination-pattern 7777 // i direct calls from
outside to trixbox ivr.
session target ipv4:10.102.19.11
dtmf-relay rtp-nte
codec g711ulaw
!
[trixbox1.localdomain asterisk]# yum install asterisk-addons
[trixbox1.localdomain asterisk]# cat /etc/asterisk/ooh323.conf
; [general] section defines global parameters
[general]
port=1720
bindaddr=10.102.19.11
;Default - no
;gateway=no
faststart=yes
h245tunneling=yes
;mediawaitforconnect=yes
;h323id=ObjSysAsterisk
;e164=100
callerid=ivr
;gatekeeper = DISCOVER
;gatekeeper = a.b.c.d
gatekeeper = DISABLE
;logfile=/var/log/asterisk/h323_log
;context=default
context=from-pstn
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity when
we're not on hold
tos=mincost ;Default - none (lowdelay, thoughput, reliability, mincost,
none)
amaflags = default
;accountcode=h3230101
disallow=all ;Note order of disallow/allow is important.
;allow=gsm
allow=ulaw
; h245alphanumeric, h245signal.
;Default - rfc 2833
;dtmfmode=rfc2833
dtmfmode = rfc2833, h245alphanumeric, h245signal
;Define users here
;Section header is extension
;[myuser1]
;type=user
;context=context1
;disallow=all
;allow=gsm
;allow=ulaw
[ccme]
type=peer
;context=default
context=from-pstn
ip=10.102.19.10 ; CME ip address
port=1720
allow=ulaw
disallow=all
;[myfriend1]
;type=friend
;context=default
;ip=10.0.0.82 ; UPDATE with appropriate ip address
;port=1820 ; UPDATE with appropriate port
;disallow=all
;allow=ulaw
;e164=12345
;rtptimeout=60
;dtmfmode=rfc2833
Freepbx, add other trunk ;
Custom Dial String = OOH323/$OUTNUM$@10.102.19.10:1720
Add same cme extensions to trixbox and add follow me settings, for example ;
1001 extensions 91001#
Default outside routes;
9|.
And select created h323 trunk.
Its working perfectly,
Br,
Baris.
-----Original Message-----
From: matthn at gmail.com [mailto:matthn at gmail.com] On Behalf Of Nick Matthews
Sent: Monday, March 01, 2010 5:55 PM
To: baris gulten
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] ccme sip trunk problem
This looks like a password mismatch.
CME tries to send invite, receives response it needs authentication
with 'asterisk' realm.
CME tries to send invite with authenication in 'asterisk' realm, is
rejected again.
-nick
On Mon, Mar 1, 2010 at 1:57 AM, baris gulten <barisgulten at gmail.com> wrote:
> 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>;tag=1A5CDC-2B5
> To: <sip:7777 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>;tag=1A5CDC-2B5
> To: <sip:7777 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>;tag=1A5CDC-2B5
> To: <sip:7777 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>;tag=1A5CDC-2B5
> To: <sip:7777 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",respons
e="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>;tag=1A5CDC-2B5
> To: <sip:7777 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>;tag=1A5CDC-2B5
> To: <sip:7777 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#
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