Re: [Sip-implementors] URI value mandatory for Alert-Info?

2015-05-12 Thread Alex Balashov
‎Many thanks for that! -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ Sent from my BlackBerry.   Ori

Re: [Sip-implementors] URI value mandatory for Alert-Info?

2015-05-12 Thread Dale R. Worley
Alex Balashov writes: > My fundamental question is: is there any grammatically acceptable way to > have an Alert-Info header value consisting of no URI and just a > generic-param, e.g. > > Alert-Info: ;internal=xyz > Alert-Info: <>;internal=xyz The URI is not optional and there is no "e

Re: [Sip-implementors] SIP Call on hold - Reinvite Session Attribute Vs Media Attribute

2015-05-12 Thread Brett Tate
Hi, Paul answered your question. However if something is actually sending mixed case "SendOnly", it might have interoperability problems since RFC 4566 section 5 discusses values as being case-sensitive unless a specific field defines otherwise. > -Original Message- > From: sip-implemen

Re: [Sip-implementors] Q regarding call transfer and P-Asserted-Identity

2015-05-12 Thread Paul Kyzivat
ISTM that this is a value judgement on the part of the PBX - exactly how it thinks its transfer feature should work. In particular, does it *want* to disclose to the caller that a transfer has occurred. Thanks, Paul On 5/12/15 11:55 AM, ankur bansal wrote: Hi Roger Cisco PBX

Re: [Sip-implementors] SIP Call on hold - Reinvite Session Attribute Vs Media Attribute

2015-05-12 Thread Paul Kyzivat
On 5/12/15 9:35 AM, Sylvester, Prasanth (Nokia - IN/Mumbai) wrote: Hi Team, I've a small doubt. I have a working call where A party is talking to B party. B party putts the call on hold, however I see "Session Attribute" with a value "SendOnly". Mostly I've seen it coming in Media Attribute.

Re: [Sip-implementors] Q regarding call transfer and P-Asserted-Identity

2015-05-12 Thread ankur bansal
Hi Roger Cisco PBX behavior seems correct here .Also after transfer is complete in step 3 , Phone-A is out of picture after having BYE exchange with Phone-B and PBX. So during step 4, PBX should use PAI of Phone-B. Regards Ankur Bansal On Tue, May 12, 2015 at 4:55 PM, Roger Wiklund wrote: > Hi

Re: [Sip-implementors] URI value mandatory for Alert-Info?

2015-05-12 Thread Paul Kyzivat
On 5/12/15 10:27 AM, CALME, Jim (Jim)** CTR ** wrote: Hi Paul, Would it be feasible to use a "data" URL scheme as defined by RFC 2397? data:[][;base64], With this syntax, the “mediatype” and “base64” are optional and may be omitted leaving the “,” available for carrying information as an alter

[Sip-implementors] SIP Call on hold - Reinvite Session Attribute Vs Media Attribute

2015-05-12 Thread Sylvester, Prasanth (Nokia - IN/Mumbai)
Hi Team, I've a small doubt. I have a working call where A party is talking to B party. B party putts the call on hold, however I see "Session Attribute" with a value "SendOnly". Mostly I've seen it coming in Media Attribute. "Session Attribute (a): sendonly" Instead of "Media Attribute (a): Se

Re: [Sip-implementors] URI value mandatory for Alert-Info?

2015-05-12 Thread CALME, Jim (Jim)** CTR **
Hi Paul, Would it be feasible to use a "data" URL scheme as defined by RFC 2397? data:[][;base64], With this syntax, the "mediatype" and "base64" are optional and may be omitted leaving the "," available for carrying information as an alternative to the Alert-Info generic-param. Thoug

Re: [Sip-implementors] URI value mandatory for Alert-Info?

2015-05-12 Thread Paul Kyzivat
On 5/11/15 9:12 PM, Alex Balashov wrote: Hello, The ABNF grammar for Alert-Info is: Alert-Info = "Alert-Info" HCOLON alert-param *(COMMA alert-param) alert-param = LAQUOT absoluteURI RAQUOT *( SEMI generic-param ) I have an application that seems to utilise just a generic-param wi

[Sip-implementors] Q regarding call transfer and P-Asserted-Identity

2015-05-12 Thread Roger Wiklund
Hi folks! Scenario: ITSP--PBX--PHONE-A--PHONE-B 1. Call from PSTN via ITSP to PHONE-A (via B2BUA PBX) 2. PHONE-A answers the call 3. PHONE-A makes a supervised transfer to PHONE-B (REFER within the PBX) 4. PBX sends UPDATE/Re-INVITE to ITSP with updated SDP co