Tek-Tips is the largest IT community on the Internet today!

Members share and learn making Tek-Tips Forums the best source of peer-reviewed technical information on the Internet!

Register Log in
  • Congratulations TouchToneTommy on being selected by the Tek-Tips community for having the most helpful posts in the forums last week. Way to Go!

Incoming sip problem!

Status
Not open for further replies.
johnnybrian

johnnybrian

IS-IT--Management
Sep 11, 2007
233
GB
Hi!

I have setup an ipo connected via SIP to an audiocodes FXO 114. Works fine for outgoing calls from Avaya to Audiocodes via line group 9, but something is wrong the other way.

I have configured an incoming call route on my Avaya:
Line group 9 >Destination: 2006 (an ip phone on my desk)

The audiocodes gateway (192.168.13.10) has been set to dial 2006@192.168.33.240 (the ip office) but no calls ever reach my phone. I get ring back tone, but the 2006 phone remains silent.

Any ideas?

Here is the log:
********** Warning: Missed 1 packet(s) **********
9856588mS SipDebugInfo: 9.1428.1 -1 SIPTrunk Endpoint(ffcf8d70) SendSIPResponse, Number of Tag Count, 0
9856590mS SIP Trunk: 9:Tx
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.13.10;branch=z9hG4bKac788781461
From: <sip:2003@192.168.13.10>;tag=1c788777666
To: <sip:2006@192.168.33.240;user=phone>;tag=2b75f3902b700b5d
Call-ID: 7887772512120009028@192.168.13.10
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

9856592mS SipDebugInfo: 9.1428.1 -1 SIPTrunk Endpoint(ffcf8d70) INVITE Received ep 9.1428.1 -1 SIPTrunk Endpoint(ffcf9994), dialog ffcf8d70
9856593mS SipDebugInfo: 9.1428.1 -1 SIPTrunk Endpoint(ffcf8d70) UpdateSIPCallState SIPDialog::INITIAL(0) -> SIPDialog::INVITE_RCVD(10)
9856594mS SipDebugInfo: 9.1428.1 -1 SIPTrunk Endpoint(ffcf8d70) Present Call, no match (2006) from URI in To header.
9856594mS SipDebugInfo: 9.1428.1 -1 SIPTrunk Endpoint(ffcf8d70) Present Call, no match from URI in Request Line
9856595mS SipDebugInfo: 9.1428.1 -1 SIPTrunk Endpoint(ffcf8d70) SendSIPResponse: INVITE SENT TO 192.168.13.10 5060
9856596mS SipDebugInfo: 9.1428.1 -1 SIPTrunk Endpoint(ffcf8d70) Sending code 404 to method INVITE
9856597mS SipDebugInfo: 9.1428.1 -1 SIPTrunk Endpoint(ffcf8d70) SendSIPResponse, Number of Tag Count, 1
9856599mS SIP Trunk: 9:Tx
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.13.10;branch=z9hG4bKac788781461
From: <sip:2003@192.168.13.10>;tag=1c788777666
To: <sip:2006@192.168.33.240;user=phone>;tag=2b75f3902b700b5d
Call-ID: 7887772512120009028@192.168.13.10
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

9856601mS SipDebugInfo: 9.1428.1 -1 SIPTrunk Endpoint(ffcf8d70) UpdateSIPCallState SIPDialog::INVITE_RCVD(10) -> SIPDialog::FINAL(40)
9856601mS SipDebugInfo: SIP Line (9): Freed Txn Key 2016
9856720mS SIP Trunk: 9:Rx
ACK sip:2006@192.168.33.240;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.13.10;branch=z9hG4bKac788781461
Max-Forwards: 70
From: <sip:2003@192.168.13.10>;tag=1c788777666
To: <sip:2006@192.168.33.240;user=phone>;tag=2b75f3902b700b5d
Call-ID: 7887772512120009028@192.168.13.10
CSeq: 1 ACK
Contact: <sip:danosha@192.168.13.10>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-114 FXO/v.5.40A.033.005
Content-Length: 0
 
Sort by date Sort by votes
Did you add a SIP URI, instead of "Use Authentication Name" use "Use User Data". Goto the user, you'll find a new sip tab no the sip extension.

SIP Name:2006
SIP Display Name (Alias):2006
Contact:2006

Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
Upvote 0 Downvote
Other thing to try, leave it the way it worked.

Add a New SIP URI on the SIP line, don't "Use Authentication Name" or "Use User Data"

type 2006 on all 3 pulldown menus.


Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
Upvote 0 Downvote
Bas: Exactly, thats what i did to make it work! :)

Now it calls on my phone, thanks alot...

How would I direct it to a voicemail destination? Should the Audiocodes box dial VM:XXX@192.168.33.240, or that wouldnt work, would it?
 
Upvote 0 Downvote
Make a shorcode;

1234/VM:XXXXXXXXXX/Voicemail Collect
(NO @192.168.33.240 NEEDED)

And use 1234 instead

Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
Upvote 0 Downvote
Status
Not open for further replies.

Similar threads

Jouik
  • Locked
  • Question
Incoming calls Caller ID Name not showing on SIP Phones although received in Monitor trace
Replies
1
Views
316
Mo. Abdullah
Mo. Abdullah
Imran555
  • Locked
  • Question
Incoming caller ID with SIP server IP over SIP Trunks 1
Replies
1
Views
879
ipohead
ipohead
AndoverService
  • Question
Echo on incoming calls analog lines 2
Replies
4
Views
500
ACSscott
ACSscott
rsv23
  • Locked
  • Question
SIP Trunk Routing
Replies
3
Views
436
rsv23
rsv23
tac84
  • Locked
  • Solved
Problem with Critical Memory Alarms possibly related to my SIP Trunk
Replies
8
Views
603
tac84
tac84

Part and Inventory Search

Sponsor

Back
Top