In a PBX environment it'll be better that phone sends the star code in an INVITE especially for features like pickup.
For the forwarding why wouldn't you use the local forwarding on the phone itself? Vishal -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sathish Chandrasekaran Sent: Monday, March 27, 2006 5:42 AM To: Aswin Bhupalam; [email protected] Subject: Re: [Sip-implementors] Query on Subscribe for Call forward feature Aswin, One thing I want to clarify is , Iam not looking at the call flows. I need details about - how the feature enabling in POTS phone will be translated to SIP message ?. If the feature enabling is done thro' SUBSCRIBE message , what will be event package name for call transfer /call forward . ? If not thro' SUBSCRIBE what will be the message used to enable a feature like call transfer /call forward.? Thanks, sathish Aswin Bhupalam <[EMAIL PROTECTED]> wrote: Sathish, Find attached sipping services draft. Hope this helps. Regards, Aswin Bhupalam. ----- Original Message ----- From: "Sathish Chandrasekaran" To: Sent: Monday, March 27, 2006 2:52 PM Subject: [Sip-implementors] Query on Subscribe for Call forward feature > All, > I have query on "SUBSCRIBE" regarding call features like Call forward unconditional. > Say for example , > Scenario is > - POTS phone connected to CPE( which Sends out as SIP message towards the SIP Server ) > > For example : > -If the end user needs to activate call forward busy / unconditional from POTS phone.he will be just dialing *xx followed by respective phone number. > 1) what would be the SIP message needs to sends out to server for intimating server to enable the feature ? > I feel it would be the "SUBSCRIBE" Message .If this is Ok , then what would be the package name ? > Can any body give me a pointer to solve this ? > Thanks , > sathish > > > > > > --------------------------------- > Jiyo cricket on Yahoo! India cricket > Yahoo! Messenger Mobile Stay in touch with your buddies all the time. > _______________________________________________ > Sip-implementors mailing list > [email protected] > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors SIPPING Working Group A. Johnston Internet-Draft MCI Expires: January 18, 2006 R. Sparks C. Cunningham S. Donovan Estacado Systems K. Summers Sonus July 17, 2005 Session Initiation Protocol Service Examples draft-ietf-sipping-service-examples-09 Status of this Memo By submitting this Internet-Draft, each author represents that any applicable patent or other IPR claims of which he or she is aware have been or will be disclosed, and any of which he or she becomes aware will be disclosed, in accordance with Section 6 of BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt. The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. This Internet-Draft will expire on January 18, 2006. Copyright Notice Copyright (C) The Internet Society (2005). Abstract This document gives examples of Session Initiation Protocol (SIP) services. This covers most features offered in so-called IP Centrex offerings from local exchange carriers and PBX (Private Branch Johnston, et al. Expires January 18, 2006 [Page 1] Internet-Draft SIP Service Examples July 2005 Exchange) features. Most of the services shown in this document are implemented in the SIP User Agents, although some require the assistance of a SIP Proxy. Some require some extensions to SIP including the REFER, SUBSCRIBE, and NOTIFY methods and the Replaces and Join headers. These features are not intended to be an exhaustive set, but rather show implementations of common features likely to be implemented on SIP IP telephones in a business environment. Table of Contents 1. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Service Examples . . . . . . . . . . . . . . . . . . . . . . 4 2.1 Call Hold . . . . . . . . . . . . . . . . . . . . . . . . 5 2.2 Consultation Hold . . . . . . . . . . . . . . . . . . . . 17 2.3 Music On Hold . . . . . . . . . . . . . . . . . . . . . . 34 2.4 Transfer - Unattended . . . . . . . . . . . . . . . . . . 42 2.5 Transfer - Attended . . . . . . . . . . . . . . . . . . . 49 2.6 Transfer - Instant Messaging . . . . . . . . . . . . . . . 61 2.7 Call Forwarding Unconditional . . . . . . . . . . . . . . 67 2.8 Call Forwarding - Busy . . . . . . . . . . . . . . . . . . 73 2.9 Call Forwarding - No Answer . . . . . . . . . . . . . . . 80 2.10 3-way Conference - Third Party is Added . . . . . . . . 88 2.11 3-way Conference - Third Party Joins . . . . . . . . . . 94 2.12 Single Line Extension . . . . . . . . . . . . . . . . . 99 2.13 Find-Me . . . . . . . . . . . . . . . . . . . . . . . . 117 2.14 Call Management (Incoming Call Screening) . . . . . . . 128 2.15 Call Management (Outgoing Call Screening) . . . . . . . 135 2.16 Call Park . . . . . . . . . . . . . . . . . . . . . . . 138 2.17 Call Pickup . . . . . . . . . . . . . . . . . . . . . . 148 2.18 Automatic Redial . . . . . . . . . . . . . . . . . . . . 156 2.19 Click to Dial . . . . . . . . . . . . . . . . . . . . . 161 3. Security Considerations . . . . . . . . . . . . . . . . . . 165 4. IANA Considerations . . . . . . . . . . . . . . . . . . . . 165 5. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 165 6. Document History . . . . . . . . . . . . . . . . . . . . . . 166 6.1 Changes since -07 . . . . . . . . . . . . . . . . . . . . 166 7. References . . . . . . . . . . . . . . . . . . . . . . . . . 166 7.1 Normative References . . . . . . . . . . . . . . . . . . . 166 7.2 Informative References . . . . . . . . . . . . . . . . . . 167 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . 168 Intellectual Property and Copyright Statements . . . . . . . 169 Johnston, et al. Expires January 18, 2006 [Page 2] Internet-Draft SIP Service Examples July 2005 1. Overview This document provides example call flows detailing a SIP implementation of the following traditional telephony services: Call Hold Music on Hold Unattended Transfer Consultation Hold Unconditional Call Forwarding Attended Transfer No Answer Call Forwarding Busy Call Forwarding Single-Line Extension 3-way Call Incoming Call Screening Find-Me Call Pickup Call Park Outgoing Call Screening Automatic Redial Click to Dial The call flows shown in this document were developed in the design of a SIP IP communications network. They represent an example set of so-called IP Centrex services or PBX services. It is the hope of the authors that this document will be useful for SIP implementers, designers, and protocol researchers alike and will help further the goal of a standard implementation of RFC 3261 [2] These flows represent carefully checked and working group reviewed scenarios of SIP service examples as a companion to the specifications. These call flows are based on the current version 2.0 of SIP in RFC 3261 [2] with SDP usage described in RFC 3264 [5] Other RFCs also comprise the SIP standard and are used and references in these call flows. The SIP specification and the other referenced documents are definitive as far as protocol issues are concerned. Also, these flows do not represent the only way to implement these services - other approaches such as 3pcc (Third Party Call Control) [17] or Back-to-Back User Agents (B2BUA) may be more appropriate in some circumstances. The peer-to-peer design and principles of these service examples are described in the Multiparty Framework document [12]. These flows assume the functionality described in the SIP Call Flow Examples document [16], which explores basic SIP behavior. Some of the scenarios described herein make use of the SIP method extension REFER [3] and the SIP header extension Replaces [4], the SIP header extension Join [9], and some of the concepts in the 3pcc (third party call control) [17] document. The SIP Events document [6] describes the use of SUBSCRIBE and NOTIFY while the SIP Dialog Event Package Johnston, et al. Expires January 18, 2006 [Page 3] Internet-Draft SIP Service Examples July 2005 [8] document describes the dialog event package. Some examples make use the GRUU (Globally Routable User Agent URI) [20] extension. These flows were prepared assuming a network of proxies, registrars, PSTN gateways, and other SIP servers. The use of Secure SIP URIs (sips) is shown throughout this document with assumed certificate validation for security. However, other security approaches such as Digest challenges can be used. Each call flow is presented with a textual description of the scenario, a message flow diagram showing the messages exchanged between separate network elements, and the detailed contents of each message shown in the diagram. 1.1 Legend for Message Flows Dashed lines (---) represent control messages that are mandatory to the call scenario. These control messages can be SIP signaling. Double dashed lines (===) represent media paths between network elements. Messages with parenthesis around name represent optional control messages. Messages are identified in the Figures as F1, F2, etc. This references the message details in the table that follows the Figure. Comments in the message details are shown in the following form: /* Comments. */ 2. Service Examples Johnston, et al. Expires January 18, 2006 [Page 4] Internet-Draft SIP Service Examples July 2005 2.1 Call Hold Alice Proxy Bob | | | | INVITE F1 | | |--------------->| | | | INVITE F2 | |(100 Trying) F3 |------------->| |<---------------| | | |180 Ringing F4| | 180 Ringing F5 |<-------------| |<---------------| | | | 200 OK F6 | | 200 OK F7 |<-------------| |<---------------| | | ACK F8 | | |--------------->| ACK F9 | | |------------->| | Both way RTP Established | |<=============================>| | |INVITE(hold) F10 |INVITE(hold) F11|<-------------| |<---------------| | | 200 OK F12 | | |--------------->| 200 OK F13 | | |------------->| | | ACK F14 | | ACK F15 |<-------------| |<---------------| | | No RTP Sent! | | | INVITE F16 | | INVITE F17 |<-------------| |<---------------| | | 200 OK F18 | | |--------------->| 200 OK F19 | | |------------->| | | ACK F20 | | ACK F21 |<-------------| |<---------------| | | Both way RTP Established | |<=============================>| | BYE F22 | | |--------------->| BYE F23 | | |------------->| | | 200 OK F24 | | 200 OK F25 |<-------------| |<---------------| | | | | Johnston, et al. Expires January 18, 2006 [Page 5] Internet-Draft SIP Service Examples July 2005 In this scenario, Alice calls Bob, then Bob places the call on hold. Bob then takes call off hold. Alice hangs up call. Note that hold is unidirectional in nature. However, a UA that places the other party on hold will generally also stop sending media, resulting in no media exchange between the UAs. Older UAs may set the connection address to 0.0.0.0 when initiating hold. However, this behavior has been deprecated in favor of using the a=sendonly SDP attribute. Message Details F1 INVITE Alice -> Proxy 1 INVITE sips:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf9 Max-Forwards: 70 From: Alice ;tag=1234567 To: Bob Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: ... v=0 o=alice 2890844526 2890844526 IN IP4 client.atlanta.example.com s=Session SDP c=IN IP4 client.atlanta.example.com t=3034423619 0 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F2 INVITE Proxy 1 -> Bob INVITE sips:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/TLS ss1.example.com:5061;branch=z9hG4bK83749.1 Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf9 ;received=192.0.2.103 Record-Route: Max-Forwards: 69 From: Alice ;tag=1234567 To: Bob Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: Johnston, et al. Expires January 18, 2006 [Page 6] Internet-Draft SIP Service Examples July 2005 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: ... v=0 o=alice 2890844526 2890844526 IN IP4 client.atlanta.example.com s=Session SDP c=IN IP4 client.atlanta.example.com t=3034423619 0 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F3 (100 Trying) Proxy 1 -> Alice SIP/2.0 100 Trying Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf9 ;received=192.0.2.103 From: Alice ;tag=1234567 To: Bob Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Content-Length: 0 F4 180 Ringing Bob -> Proxy 1 SIP/2.0 180 Ringing Via: SIP/2.0/TLS ss1.example.com:5061;branch=z9hG4bK83749.1 ;received=192.0.2.54 Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf9 ;received=192.0.2.103 Record-Route: From: Alice ;tag=1234567 To: Bob ;tag=314159 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: Content Length:0 F5 180 Ringing Proxy 1 -> Alice SIP/2.0 180 Ringing Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf9 ;received=192.0.2.103 Record-Route: Johnston, et al. Expires January 18, 2006 [Page 7] Internet-Draft SIP Service Examples July 2005 From: Alice ;tag=1234567 To: Bob ;tag=314159 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: Content Length: 0 F6 200 OK Bob -> Proxy 1 SIP/2.0 200 OK Via: SIP/2.0/TLS ss1.example.com:5061;branch=z9hG4bK83749.1 ;received=192.0.2.54 Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf9 ;received=192.0.2.103 Record-Route: From: Alice ;tag=1234567 To: Bob ;tag=314159 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: ... v=0 o=bob 2890844527 2890844527 IN IP4 client.biloxi.example.com s=Session SDP c=IN IP4 client.biloxi.example.com t=3034423619 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F7 200 OK Proxy 1 -> Alice SIP/2.0 200 OK Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf9 ;received=192.0.2.103 Record-Route: From: Alice ;tag=1234567 To: Bob ;tag=314159 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Johnston, et al. Expires January 18, 2006 [Page 8] Internet-Draft SIP Service Examples July 2005 Content-Type: application/sdp Content-Length: ... v=0 o=bob 2890844527 2890844527 IN IP4 client.biloxi.example.com s=Session SDP c=IN IP4 client.biloxi.example.com t=3034423619 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 F8 ACK Alice -> Proxy 1 ACK sips:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf92 Route: Max-Forwards: 70 From: Alice ;tag=1234567 To: Bob ;tag=314159 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Length: 0 F9 ACK Proxy 1 -> Bob ACK sips:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/TLS ss1.example.com:5061;branch=z9hG4bK837492.1 Via: SIP/2.0/TLS client.atlanta.example.com:5061;branch=z9hG4bK74bf92 ;received=192.0.2.103 Max-Forwards: 69 From: Alice ;tag=1234567 To: Bob ;tag=314159 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Length: 0 /* Bob places Alice on hold. Note that the version is incremented in the o= field of the SDP */ F10 INVITE Bob -> Proxy 1 Johnston, et al. Expires January 18, 2006 [Page 9] Internet-Draft SIP Service Examples July 2005 INVITE sips:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/TLS client.biloxi.example.com:5061;branch=z9hG4bKnashds7 Route: Max-Forwards: 70 From: Bob ;tag=314159 To: Alice ;tag=1234567 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: ... v=0 o=bob 2890844527 2890844528 IN IP4 client.biloxi.example.com s=Session SDP c=IN IP4 client.biloxi.example.com t=3034423619 0 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendonly F11 INVITE Proxy 1 -> Alice INVITE sips:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/TLS ss1.example.com:5061;branch=z9hG4bK83749.1 Via: SIP/2.0/TLS client.biloxi.example.com:5061;branch=z9hG4bKnashds7 ;received=192.0.2.105 Record-Route: Max-Forwards: 69 From: Bob ;tag=314159 To: Alice ;tag=1234567 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: ... v=0 o=bob 2890844527 2890844528 IN IP4 client.biloxi.example.com s=Session SDP c=IN IP4 client.biloxi.example.com === message truncated === --------------------------------- Jiyo cricket on Yahoo! India cricket Yahoo! Messenger Mobile Stay in touch with your buddies all the time. _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
