You can try sending 405 method not allowed. But acc to RFC3261, as soon as UAC sends a BYE, it assumes that the session is terminated. You may not have explicit control over by at IP Phone end. Also see the RFC snippet here: ----------------------------------- 15.1.1 UAC Behavior
A BYE request is constructed as would any other request within a dialog, as described in Section 12.
Once the BYE is constructed, the UAC core creates a new non-INVITE client transaction, and passes it the BYE request. The UAC MUST consider the session terminated (and therefore stop sending or listening for media) as soon as the BYE request is passed to the client transaction. If the response for the BYE is a 481 (Call/Transaction Does Not Exist) or a 408 (Request Timeout) or no response at all is received for the BYE (that is, a timeout is returned by the client transaction), the UAC MUST consider the session and the dialog terminated. --------------------------------------- -Ranga --- Neill Wilkinson neill.wilkinson@btinternet.com wrote:
All,
just started using SER (great experience so far!). I'm trying to us it for a contact centre application where I have a specific requirement that calls can only be terminated in one direction, from a Cisco AS53xx gateway, not a Cisco phone.
Here's the set-up and desired result:
User logs into the contact centre systems which initiates a call on the PSTN side of the AS53xx. This then creates a VOIP (SIP) call leg from a dial-peer on the gateway, via the SER proxy.
The agents all have Cisco 7960 IP phones which register with the SER registrar. The INVITE is proxied via a simple location lookup in the SER route[0] config file to the appropriate handset. The agent answers the call and is "logged" in to the system.
Now's the fun part. I want to prevent the SIP leg being torn-down by the agent either accidentally or on purpose hanging up the call from the handset.
Here's the logic (pseudo code) I have though out: Trap the BYE from the handset using a statement something like:
if (method=="BYE" and uri=~"sip:[0-9]+@*") { # Agent phone initiated the BYE so dosomething route(1); } else { #do something else - pass on the message I guess };
route[1] { # use exec to call an external function exec_msg ('script.file'); }
The external script needs to OK the BYE message - I assume use the FIFO function to construct the OK message.
Now the tricky part: how do I stop the called phone from hanging up, (I tried a simple 403 Forbidden response) but that didn't work. So I thought about the REFER method with the URL of the phone sending the BYE message- can I send a refer to the Gateway (AS5300) that still thinks the dialogue is open to the phone and "hopefully" get it to send a new INVITE dialogue to the phone? Or would I have to generate a new INVITE sequence from the SER proxy exec code?
If its the latter, then I need access to the RTP port number on the AS53xx from the original INVITE message - and this implies access to the SDP payload - which I can't see from the SER script (unless anyone can tell me how?)
Here's a picture of what I hope to achieve!
[AS5300} [SER Proxy} [Cisco 7960]
----------------------------------------------->INVITE
<----------------------------------------------TRYING
INVITE ------------------------------->
<---------------------------------TRYING
<---------------------------------RINGING
<----------------------------------------------RINGING
<-----------------------------------OK
<------------------------------------------------OK
<==================RTP exchange=================>
<-------------------------------- BYE
OK ---------------------------------->
<----------------------------------------------REFER (with URI of Cisco 7960)
-------------------------------------------------->INVITE
<-------------------------------------------------TRYING
INVITE ----------------------------->
<-------------------------------TRYING
<-------------------------------RINGING
<--------------------------------------------------RINGING
<-------------------------------OK
<-------------------------------------------------- OK <=================RTP Exchange ===================>
Thanks in advance!!! All ideas Gratefully received!
Neill....:o)
Neill Wilkinson
Senior Consultant
Quortex Consultants Ltd.
e: neill.wilkinson@quortex.com
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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