- This topic has 2 replies, 1 voice, and was last updated 17 years ago by Mastaplan.
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22nd October 2007 at 11:13 #31574mastaplanGuest
Hello,
Im configuring tenor d2400 on 1 e1 connection on asterisk server.
I successfully configured the outbound routes. (I can call from my ATA phone then goes out to the e1 connection).
My problem is i can’t able to pass the incoming calls coming from my e1 to the sip server.
Can anybody help me out on configuring this?
i tried to get the logs on ev log and im getting an alarm on ev log “bad ip str”..
22nd October 2007 at 12:22 #31575MikeM to MastaplanGuestIf you are using SIP, you need to make sure you have the sip server configured in the quintum and that all incoming calls have a destination number. At that point the quintum will send a message to the sip server for routing instructions.
MikeM
mike_voip@hotmail.com22nd October 2007 at 12:54 #31576MastaplanGuestThanks for the reply.
I have configured my sip primary registrar and proxy in the quintum.
when i checked the sip stat command in the CLI it says registered.
Im debugging my quintum and the alarm shows:
CH : 25043553:Received ANI: 09228757478.
CH : 25043553:received info = 0973777111F
CH : 25043553:[2:27] sent message to cas: Call-Proc
CH : 25043553:Routing requested for: public(1) orig=0973777111F public(1) normalized=0973777111 route code= tg=0.
CH : 25043553:1 match(es) found: 2
CH : 25043553:chsip : bandwidth info: max=-1 cur=12600.
CH : 25043553:Route response(154): result=1 cause=0.
CH : 25043553:TBCSM[154]: Setup from peer=0xd08004 NP=0x1 NT=0x1.
CH : 25043553:OrigNum=0973777111F NormNum=0973777111 TranNum=0973777111 OrigDest=.
CH : 25043553:sip[154]: tsipcall:stackSendSetup, media type=16
CH : 25043553:udp connect: 2 11
CH : 25043553: cba763b7 10240 0 0
CH : 25043553:siptg: outgoing translation 0973777111F -> 639730973777111.
CH : 25043553:sent message to sip: msg=2; ua=1
EXCP : 25043561:Bad IP addr str= calling proc=0x498ba0 proc1=0x470bac
EXCP : 25043569:Bad IP addr str= calling proc=0x498ba0 proc1=0x470bac
CH : 25043573:sip[0/0]: SipReleaseComplete rcvd at SipTermCall
CH : 25043573:sip[154/0]: tsipcall:RcvRelComp, cause=111
CH : 25043573:udp disconnect: 2 11
CH : 25043573: cba763b7 10240 0 0
CH : 25043573:OBCSM[154]: Release from peer=0xd0680c cause=0x6f redir=.
CH : 25043573:TBCSM[154]: Release complete from peer=0xd08004.
CH : 25043573:sip[0/0]: sipTG::term releaseCall:remove call from listhow do you configure this?
TIA
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