Hello all!
I have PBX Aastra MX-ONE TSW which is connected to Cisco 2651xm via E1 PRI. So, Cisco2651xm is connected to Asterisk 11.2.1 via SIP trunk.
Cisco 2651xm has ip address 172.26.18.200
Asterisk has ip address 10.2.210.13
I can to make call from softphone 4092@10.2.210.13 in Asterisk to telephones 1312@172.26.18.200 Aastra MX-ONE successfully. But, I cannot to make call from Aastra telephones to Asterisk softphones. I recieve busy in telephone.
Cisco config:
Router#sho run
Building configuration...
Current configuration : 2126 bytes
!
! Last configuration change at 11:52:47 UTC Tue Jun 4 2013 by culture
!
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
!
card type e1 1 0
no logging console
!
no aaa new-model
no network-clock-participate slot 1
no network-clock-participate wic 0
ip cef
!
!
!
!
no ip domain lookup
ip auth-proxy max-nodata-conns 3
ip admission max-nodata-conns 3
!
!
!
isdn switch-type primary-qsig
voice-card 1
!
!
!
voice call carrier capacity active
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
redirect ip2ip
signaling forward rawmsg
sip
no call service stop
!
!
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
!
!
!
!
!
!
!
!
!
!
!
archive
log config
hidekeys
!
!
controller E1 1/0
pri-group timeslots 1-31
!
controller E1 1/1
!
!
!
!
!
!
interface FastEthernet0/0
no ip address
shutdown
duplex auto
speed auto
!
interface FastEthernet0/1
ip address 172.26.18.200 255.255.255.0
duplex auto
speed auto
!
interface Serial1/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn global-disconnect
isdn contiguous-bchan
isdn bchan-number-order ascending
no cdp enable
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 172.26.18.10
!
!
no ip http server
no ip http secure-server
!
!
!
!
control-plane
!
!
!
voice-port 1/0:15
!
!
!
!
!
dial-peer voice 1 pots
description *Aastra MX-ONE*
destination-pattern 1312
direct-inward-dial
port 1/0:15
forward-digits all
!
dial-peer voice 5 voip
description *to Asterisk*
destination-pattern 4092
voice-class codec 1
session protocol sipv2
session target ipv4:10.2.210.13
session transport udp
dtmf-relay rtp-nte
!
gateway
!
sip-ua
sip-server ipv4:10.2.210.13
!
!
!
!
line con 0
line aux 0
line vty 0 4
exec-timeout 30 0
logging synchronous
login local
transport input telnet
!
!
end
When I try to make call from telephone Aastra 1312@172.26.18.200 to softphone Asterisk 4092@10.2.210.13 i recieve these debug messages:
Jun 4 12:18:36.442: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:4092@10.2.210.13:5060 SIP/2.0
Via: SIP/2.0/UDP 172.26.18.200:5060;branch=z9hG4bK3733
From: <sip:1312@172.26.18.200>;tag=19D04532-1825
To: <sip:4092@10.2.210.13>
Date: Tue, 04 Jun 2013 12:18:36 GMT
Call-ID: B74918DF-CC4711E2-9EC98019-37A47AC2@172.26.18.200
Supported: 100rel,timer,replaces
Min-SE: 1800
Cisco-Guid: 3074740581-3427209698-2152333329-2481374176
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: <sip:1312@172.26.18.200>;party=calling;screen=yes;privacy=off
Timestamp: 1370348316
Contact: <sip:1312@172.26.18.200:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 286
v=0
o=CiscoSystemsSIP-GW-UserAgent 9478 9141 IN IP4 172.26.18.200
s=SIP Call
c=IN IP4 172.26.18.200
t=0 0
m=audio 17184 RTP/AVP 8 0 101 19
c=IN IP4 172.26.18.200
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
Jun 4 12:18:36.454: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.26.18.200:5060;branch=z9hG4bK3733;received=172.26.18.200;rport=55715
From: <sip:1312@172.26.18.200>;tag=19D04532-1825
To: <sip:4092@10.2.210.13>;tag=as00007b29
Call-ID: B74918DF-CC4711E2-9EC98019-37A47AC2@172.26.18.200
CSeq: 101 INVITE
Server: Asterisk PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="000024e0"
Content-Length: 0
Router#
Jun 4 12:18:36.458: //291/B744D165804A/SIP/Error/sipSPIHandleAuthChallenge: Error getting credentials
Jun 4 12:18:36.458: //291/B744D165804A/SIP/Error/act_sentinvite_new_message: Error handling AuthenticationChallenge
Jun 4 12:18:36.458: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:4092@10.2.210.13:5060 SIP/2.0
Via: SIP/2.0/UDP 172.26.18.200:5060;branch=z9hG4bK3733
From: <sip:1312@172.26.18.200>;tag=19D04532-1825
To: <sip:4092@10.2.210.13>;tag=as00007b29
Date: Tue, 04 Jun 2013 12:18:36 GMT
Call-ID: B74918DF-CC4711E2-9EC98019-37A47AC2@172.26.18.200
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0
Jun 4 12:18:37.928: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.26.18.200 SIP/2.0
Via: SIP/2.0/UDP 10.2.210.13:5060;branch=z9hG4bK00003cfd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.2.210.13>;tag=as00007a43
To: <sip:172.26.18.200>
Contact: <sip:asterisk@10.2.210.13:5060>
Call-ID: 00003dc200001ec4000010cc00002ebd@10.2.210.13:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.2.1
Date: Tue, 04 Jun 2013 08:18:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Content-Length: 0
Jun 4 12:18:37.944: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.210.13:5060;branch=z9hG4bK00003cfd
From: "asterisk" <sip:asterisk@10.2.210.13>;tag=as00007a43
To: <sip:172.26.18.200>;tag=19D04B0D-26CB
Date: Tue, 04 Jun 2013 12:18:37 GMT
Call-ID: 00003dc200001ec4000010cc00002ebd@10.2.210.13:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Accept: application/sdp
Allow-Events: telephone-e
Router#vent
Content-Length: 170
Content-Type: application/sdp
v=0
o=CiscoSystemsSIP-GW-UserAgent 7433 1229 IN IP4 172.26.18.200
s=SIP Call
c=IN IP4 172.26.18.200
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15 3
c=IN IP4 172.26.18.200
I don't understand these debug-messages:
Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.26.18.200:5060;branch=z9hG4bK3733;received=172.26.18.200;rport=55715
From: <sip:1312@172.26.18.200>;tag=19D04532-1825
To: <sip:4092@10.2.210.13>;tag=as00007b29
It seems to me number 1312 tries to register on Asterisk. What for? Can anyone to take me some advise? Thx.
