[cisco-voip] ccme sip trunk problem
malhatma
malhatma at gmx.net
Tue Mar 2 13:40:22 EST 2010
Hi,
here is a working solution I implemented. But it is without auth. So if you don't need that ....
http://uc-b.blogspot.com/2008/04/asterisk-cisco-callmanager-express-cme.html
Andre
Am 01.03.2010 16:55, schrieb Nick Matthews:
> 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",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>;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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>>
>>
>
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