[asterisk-users] Call Xfer issue between DataCenter and User Site

[Peder]

I had the exact same issue and it was caused by a crappy firewall at the phone site.  Once they swapped it out with a box that did NAT correctly, the issue went away. I don't think you said if the phone site is being NAT'd or firewalled and when you mentioned the debugs below, you said "datacenter IP" but you didn't specify if that was the datacenter internal IP or the public NAT'd IP.  That info would help.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steven Davison
Sent: Wednesday, January 20, 2010 11:58 AM
To: asterisk-users at lists.digium.com
Cc: Alistair Mackenzie
Subject: [asterisk-users] Call Xfer issue between DataCenter and User Site

Hi, 

I am running a Asterisk 1.6 box in our Data Centre, and have a number of users connecting to that box, as their PBX.

Calls in and out work fine, as does voicemail.

The PBX at the Data Centre has an External IP, Nat’d to it by the firewall, and the relevant ports are open.

The Office users have a dedicated internet connection for the phone lines, and calls are seen to traverse this correctly. The handsets are Linksys SPA922

The issue we are getting is in transferring calls, which happens like this :-

1. Incoming call from pstn/viop provider
2. Call is answered by a user
3. Call needs to be transferred
4. Xfer button is pushed, other user is called, answered, and they speak about the call
4b. The incoming call is held, listening to MoH
5. Xfer is pushed again,
6. <SIP Debug Output>
7. MoH stops, 
8. Office user gets no audio
9. Incoming call is silent, and then call is dropped
10. Office user gets fed up of saying ‘hello??!?’ and hangs up.

Here is the sip debug output...

<------------>
[Jan 20 16:43:38] set_destination: Parsing <sip:172 at XXX.XXX.XXX.XXX:10036> for address/port to send to
[Jan 20 16:43:38] set_destination: set destination to XXX.XXX.XXX.XXX, port 10036
[Jan 20 16:43:38] Reliably Transmitting (NAT) to XXX.XXX.XXX.XXX:10016:
NOTIFY sip:172 at XXX.XXX.XXX.XXX:10036 SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK48cff632;rport
Max-Forwards: 70
From: "Steve (NetTech)" <sip:176 at YYY.YYY.YYY.YYY>;tag=as4f7c4d0c
To: <sip:172 at XXX.XXX.XXX.XXX:10036>;tag=726be2fb618280d0i0
Contact: <sip:176 at YYY.YYY.YYY.YYY>
Call-ID: 718a30a4572984a918b88dc64df642ea at YYY.YYY.YYY.YYY
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 1.6.1.1
Event: refer;id=102
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 49

SIP/2.0 481 Call leg/transaction does not exist

---
[Jan 20 16:43:38]     -- Stopped music on hold on SIP/176-09bf9630
[Jan 20 16:43:38]
<--- SIP read from UDP://XXX.XXX.XXX.XXX:10016 --->
SIP/2.0 200 OK
To: <sip:172 at XXX.XXX.XXX.XXX:10036>;tag=726be2fb618280d0i0
From: "Steve (NetTech)" <sip:176 at YYY.YYY.YYY.YYY>;tag=as4f7c4d0c
Call-ID: 718a30a4572984a918b88dc64df642ea at YYY.YYY.YYY.YYY
CSeq: 103 NOTIFY
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK48cff632
Server: Linksys/SPA922-4.1.18
Content-Length: 0


<------------->
YYY.YYY.YYY.YYY is the IP of the Datacenter
XXX.XXX.XXX.XXX is the IP of the Office

I have been going over and over the configs on the routers, sip.conf etc trying to work this out... we have also checked that the users are using the above sequence to transfer a call...

Thanks to anyone who may have ideas for this... ☺

Steven Davison - Network Engineer
t:   0845 0034567
f:   0845 0034543
w: www.ntsols.com


 
Net Technical Solutions | Suite 1 Wesley Chambers | Queens Road | Aldershot | Hampshire | GU11 3JD








   
 

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