The information returned with the response depends on the method used in the request.ģxx responses give information about the user's new location, or about alternative services that might be able to satisfy the call. The Reason-Phrase, header fields, or message body MAY be used to convey more details about the call progress. The 183 (Session Progress) response is used to convey information about the progress of the call that is not otherwise classified.
Sip 2.0 404 not found update#
The server MAY issue several 182 (Queued) responses to update the caller about the status of the queued call. The reason phrase MAY give further details about the status of the call, for example, "5 calls queued expected waiting time is 15 minutes". When the callee becomes available, it will return the appropriate final status response. The called party is temporarily unavailable, but the server has decided to queue the call rather than reject it.
Sip 2.0 404 not found code#
This response MAY be used to initiate local ringback.Ī server MAY use this status code to indicate that the call is being forwarded to a different set of destinations. The UA receiving the INVITE is trying to alert the user. The 100 (Trying) response is different from other provisional responses, in that it is never forwarded upstream by a stateful proxy. This response, like all other provisional responses, stops retransmissions of an INVITE by a UAC. This response indicates that the request has been received by the next-hop server and that some unspecified action is being taken on behalf of this call (for example, a database is being consulted). Provisional (1xx) responses MAY contain message bodies, including session descriptions. They never cause the client to send an ACK. Note that 1xx responses are not transmitted reliably. A server sends a 1xx response if it expects to take more than 200 ms to obtain a final response. Provisional responses, also known as informational responses, indicate that the server contacted is performing some further action and does not yet have a definitive response. Other HTTP/1.1 response codes SHOULD NOT be used. Not all HTTP/1.1 response codes are appropriate, and only those that are appropriate are given here. Via: SIP/2.0/UDP 192.168.2.130:5060 branch=z9hG4bKa91iou30bof1dog296c1.The SIP response codes are consistent with, and extend to, HTTP/1.1 response codes. Via: SIP/2.0/UDP 192.168.2.130:5060 branch=z9hG4bK89kfkc30708g3oo2r1t1.1Īllow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISHĪccept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixedĬontent-Disposition: session handling=requiredĬisco-Gucid: 351192d001e367e198b6000000000000 IP address = 10.232.50.102 for SM100 (Sip Interface)ĭomain name for ECB sip peer = 192.168.2.110 there is no fqdn associated with this entity IP address = 192.168.2.110 for ECB User Agent Sip Peer Here is the traceSM call out put for the call I am having issues with.Ĭall flow is Gateway -> SBC -> SM -> Feature Server(CM) -> SM -> Phone.ĭestination is 78, a SIP 9641G provisioned on SM. RE: Avaya SM Call routing redphone (Programmer) 12 Dec 13 15:33 It walks one thru the steps to setup an inbound call adaptation. I have created a adaptation rule and just winged it the best I can (that is scarry), according to the Avaya app note "SM6ASBC_VzB_IPCC.pdf" (.
I know this app note is not to be followed verbatim, but it does speak of inbound call conversion / digit conversion for calls destined to SM. In this well written app note, it does speak of the different inbound and outbound call treatments.
I tried to follow the "SM6ASBC_VzB_IPCC.pdf" (. I do agree with you, I am pretty sure that I have a sip adaptation issue on my inbound calls on out bound calls, CM appends all the sip calls with a properly formed uri, E.164-> this is working perfectly to outbound pstn calls. I have my domain name as and that is working fine, in the routing / domain tab. As always thanks for your reply, You are great resource to this community.
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