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From: Joegen B. <joe...@gm...> - 2009-08-07 01:54:49
|
There is a documentation about Upper Registration here http://www.opensourcesip.org:8080/clearspacex/docs/DOC-1010 -------------------------------------------------- From: "busybox busybox" <bu...@ya...> Sent: Friday, August 07, 2009 3:10 AM To: <ope...@li...> Subject: [OpenSIPStack] OpenSBC: passthrough SIP registration to externalregistrar > Hello overyone. > > I'm planning to use the OpenSBC as a transparent SIP proxy to another soft > switch. > > So, I've got a soft switch (SS), which contans user accounts and performs > users' calls processing. > > But between users' soft phones and SS there is OpenSBC. So, I need the > OpenSBC to passtrough transparently users' phones registrations to the SS > and the same about users' calls. > > Registration example: > > SoftPhone ---> OpenSBC ---> SS > > 1. SoftPhone sends SIP registration to the OpenSBC > 2. The OpenSBC sends this registration transparently to the the SS > 3. SS confirms the registration of the SoftPhone > > Call processing example: > > SoftPhone1 ---> OpenSBC ---> SS ---> OpenSBC ---> SoftPhone2 > > 1. SoftPhone sends SIP INVITE to the OpenSBC > 2. The OpenSBC sends this INVITE transparently to the the SS > 3. SS performs call processing and routing and sends SIP INVITE to the > SoftPhone2 through the OpenSBC transparently > > > Well. The question is HOW to configure OpenSBC to process registrations > and calls according to the schemes above??? > Is it actually possible or I need some extra coding of OpenSBC? > > Any info appreciated > > > > ------------------------------------------------------------------------------ > Let Crystal Reports handle the reporting - Free Crystal Reports 2008 > 30-Day > trial. Simplify your report design, integration and deployment - and focus > on > what you do best, core application coding. Discover what's new with > Crystal Reports now. http://p.sf.net/sfu/bobj-july > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
|
From: busybox b. <bu...@ya...> - 2009-08-06 19:10:47
|
Hello overyone. I'm planning to use the OpenSBC as a transparent SIP proxy to another soft switch. So, I've got a soft switch (SS), which contans user accounts and performs users' calls processing. But between users' soft phones and SS there is OpenSBC. So, I need the OpenSBC to passtrough transparently users' phones registrations to the SS and the same about users' calls. Registration example: SoftPhone ---> OpenSBC ---> SS 1. SoftPhone sends SIP registration to the OpenSBC 2. The OpenSBC sends this registration transparently to the the SS 3. SS confirms the registration of the SoftPhone Call processing example: SoftPhone1 ---> OpenSBC ---> SS ---> OpenSBC ---> SoftPhone2 1. SoftPhone sends SIP INVITE to the OpenSBC 2. The OpenSBC sends this INVITE transparently to the the SS 3. SS performs call processing and routing and sends SIP INVITE to the SoftPhone2 through the OpenSBC transparently Well. The question is HOW to configure OpenSBC to process registrations and calls according to the schemes above??? Is it actually possible or I need some extra coding of OpenSBC? Any info appreciated |
|
From: Jon S. <jsm...@gm...> - 2009-08-06 18:04:31
|
I am using opensbc 1.1.5rc5 on a linux machine with two interfaces, one public and one private. I am able to place and receive calls without issue. I have Enable-Local-Refer=True in my opensbc.ini file. When I try to refer a call however, the originator sends a Refer, gets back a 202 Accepted, and a new call is placed outbound. The call connects, however none of the SIP messages or RTP from leg 2 (Public side) are sent back to Leg 1 (Private side), and so the transfer fails. I have attached a packet capture, log file, and ,my opensbc.conf file. I can provide any additional info that you might need - just let me know. Regards, Jon |
|
From: Joegen B. <joe...@gm...> - 2009-07-25 01:03:53
|
Hi Alex,
Sorry no. You will have to modify
BOOL SBCCallHandler::DumpCATLog( B2BUAConnection & conn, BOOL rejected )
method and let it connect to a DB of your choice. It should be a trivial change though.
Joegen
From: Alexandre Keller
Sent: Saturday, July 25, 2009 8:54 AM
To: Joegen Baclor ; ope...@li...
Subject: Re: [OpenSIPStack] Develop some reports on OpenSBC traffic....
Hi.
Thanks for such a quick reply.
Any way to send CDR to a DataBase?!
Thanks in advance again.
--
Atenciosamente,
ALEXANDRE KELLER
"Dinheiro é a consequência de um trabalho bem feito
e não o motivo para se fazer um bom trabalho."
On 24/07/2009, at 20:24, Joegen Baclor wrote:
Hi Alex,
OpenSBC stores CDR's in "OpenSBC_data\CAT\static" folder.
--------------------------------------------------
From: "Alexandre Keller" <ale...@gm...>
Sent: Saturday, July 25, 2009 12:49 AM
To: <ope...@li...>
Subject: [OpenSIPStack] Develop some reports on OpenSBC traffic....
Hi there.
I wonder if is there any way of store the traffic passing through
OpenSBC, signalling I mean, to generate some reports and statistics
about it?!
Thanks for any advise or help.
Best regards.
--
Atenciosamente,
ALEXANDRE KELLER
"Dinheiro é a consequência de um trabalho bem feito
e não o motivo para se fazer um bom trabalho."
------------------------------------------------------------------------------
_______________________________________________
opensipstack-devel mailing list
ope...@li...
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
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_______________________________________________
opensipstack-devel mailing list
ope...@li...
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
|
|
From: Alexandre K. <ale...@gm...> - 2009-07-25 00:54:44
|
Hi. Thanks for such a quick reply. Any way to send CDR to a DataBase?! Thanks in advance again. -- Atenciosamente, ALEXANDRE KELLER "Dinheiro é a consequência de um trabalho bem feito e não o motivo para se fazer um bom trabalho." On 24/07/2009, at 20:24, Joegen Baclor wrote: > Hi Alex, > > OpenSBC stores CDR's in "OpenSBC_data\CAT\static" folder. > > -------------------------------------------------- > From: "Alexandre Keller" <ale...@gm...> > Sent: Saturday, July 25, 2009 12:49 AM > To: <ope...@li...> > Subject: [OpenSIPStack] Develop some reports on OpenSBC traffic.... > >> Hi there. >> >> I wonder if is there any way of store the traffic passing through >> OpenSBC, signalling I mean, to generate some reports and statistics >> about it?! >> >> Thanks for any advise or help. >> >> Best regards. >> -- >> Atenciosamente, >> >> ALEXANDRE KELLER >> >> "Dinheiro é a consequência de um trabalho bem feito >> e não o motivo para se fazer um bom trabalho." >> >> >> >> ------------------------------------------------------------------------------ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > > ------------------------------------------------------------------------------ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
|
From: Joegen B. <joe...@gm...> - 2009-07-24 23:25:27
|
Hi Alex, OpenSBC stores CDR's in "OpenSBC_data\CAT\static" folder. -------------------------------------------------- From: "Alexandre Keller" <ale...@gm...> Sent: Saturday, July 25, 2009 12:49 AM To: <ope...@li...> Subject: [OpenSIPStack] Develop some reports on OpenSBC traffic.... > Hi there. > > I wonder if is there any way of store the traffic passing through > OpenSBC, signalling I mean, to generate some reports and statistics > about it?! > > Thanks for any advise or help. > > Best regards. > -- > Atenciosamente, > > ALEXANDRE KELLER > > "Dinheiro é a consequência de um trabalho bem feito > e não o motivo para se fazer um bom trabalho." > > > > ------------------------------------------------------------------------------ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
|
From: Alexandre K. <ale...@gm...> - 2009-07-24 16:50:28
|
Hi there. I wonder if is there any way of store the traffic passing through OpenSBC, signalling I mean, to generate some reports and statistics about it?! Thanks for any advise or help. Best regards. -- Atenciosamente, ALEXANDRE KELLER "Dinheiro é a consequência de um trabalho bem feito e não o motivo para se fazer um bom trabalho." |
|
From: Meftah T. <tay...@gm...> - 2009-07-18 13:18:40
|
hi OpenSipStack Users, i created a new IRC Channel in Freenode.Net for the OSS Project please joint it: #OpenSipStack in freenode.net to discuce about OSS, ATLSip and OpenSBC thanks! Welcome __________ Information from ESET NOD32 Antivirus, version of virus signature database 4256 (20090718) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com |
|
From: Meftah T. <tay...@gm...> - 2009-07-16 21:51:35
|
hello, what i can do using OpenSBC? i can host IVR applications? have some media capability? Music on hold? thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4251 (20090716) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com |
|
From: Meftah T. <tay...@gm...> - 2009-07-16 10:44:26
|
hello, open SBC support SIP over TCP? if no, please cool anyone contribute it? i'm using OpenSBC realy very good bicose of the accessibility of there Web Admin Interface is realy accessible to my screen reader thank you __________ Information from ESET NOD32 Antivirus, version of virus signature database 4249 (20090716) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com |
|
From: Joegen B. <joe...@gm...> - 2009-06-29 01:18:29
|
Hi Everyone, The time has come to release a stable build for the 1.1.5 branch of OpenSBC. OpenSBC_1.1.5_RC5-Final branch has been tagged in CVS. This means this branch would only now be limited to critical bug fixes. This is the final curtain call for those of you who want to send in bug reports which you think should be part of this major release. Sources and windows installers may be downloaded from: http://sourceforge.net/project/showfiles.php?group_id=156710 I thank everyone who has contributed their time in sending bug reports and patches making this release possible. Long live open source. Joegen Baclor Founder - http://www.opensipstack.org |
|
From: Meftah T. <tay...@gm...> - 2009-06-25 07:03:46
|
hello, thank you for your reply atlsip is a very cool ActiveX component... you tel me that atlsip store settings automatikaly... cool i encript it for a secure softphone? also, please how i can use TLS? about freeswitch, i'm only a freeswitch consiltan for configuring / administering it, i'm no a developer but i will try to help you thanks Joegen Baclor wrote: > Hi Mefta, > > inline ... > > -------------------------------------------------- > From: "Meftah Tayeb" <tay...@gm...> > Sent: Wednesday, June 24, 2009 5:00 PM > To: <ope...@li...> > Subject: [OpenSIPStack] Developing a Powerfull Softphone using ATLSip > > >> hello, >> i'm developing a Softphone using atlsip >> is the only open source Licensed SIP/VoIp ActiveX Control >> but i need to know some informations... >> please i have 2 questions: >> >> >> 1. SIP Transport: >> >> the SIP Protocol Support 3 difere transports: >> >> * TCP >> * UDP >> * TLS (Secure) >> >> ATLSip support all this 3 transport or UDP only? >> if ATLSip support TLS, how i can get / use Security Certificate? >> >> > > As of now, only UDP is 100% working in the library. TCP is work in progress > but nobody is working on that side of the code currently. There is TSL > support just yet. > > > >> Profile Storage >> >> if i register to my Server and i close my softphone, i get difere ini >> files: default.ini, and other log files >> this files is required? >> > > Yes they are. ATLSIP uses those preserve its settings. > > >> how i can store my SIP Profile including Voice codecs, audio devices, >> registration settings? >> is stored automatikaly or i must store it manualy? >> > > > The should be stored automatically. > > > >> if anyone here want mor help about some PBX i'm here to help >> OpenSipUsers for free including Freeswitch >> > > I am contemplating in using freeswitch as a codec transcoder for opensbc. I > might need some advice soon. Thanks for offering your help. > > >> if anyone want to help me, please use this informations to contact me: >> >> >> My Informations: >> >> Full Name: Meftah Tayeb >> Click Here to Mail Me <ad...@ne...> >> Click here to drop me a SIP Call (SIP:Voi...@si...) >> <sip:voi...@si...> >> Skype: tayeb.meftah >> yahoo: vsdev2006 >> MSN: SQL...@ho... >> >> >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus >> signature database 4182 (20090624) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> ------------------------------------------------------------------------------ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > > > > >> No virus found in this incoming message. >> Checked by AVG - www.avg.com >> Version: 8.5.374 / Virus Database: 270.12.90/2198 - Release Date: 06/23/09 >> 17:54:00 >> >> > > ------------------------------------------------------------------------------ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4187 (20090625) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4187 (20090625) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com |
|
From: Meftah T. <tay...@gm...> - 2009-06-25 06:59:07
|
hello please cool you give me mor stuf about that? how i can transfer call? how i can auto answer and playback a audio file? thanksAndre Silo wrote: > Yes there is transfer from one sip account to another and hold. > Auto answer will depend on you application. > > > > > ________________________________ > From: Meftah Tayeb <tay...@gm...> > To: ope...@li... > Sent: Saturday, June 6, 2009 4:25:34 PM > Subject: [OpenSIPStack] ATLSip:call transfer / hold and Auto answering > > hello OpenSipStack users, > please cool you tel me if is it pocible to transfer calls using AtlSip? > also, is it pocible to hold the call? > ------------ > and is it pocible to auto answer the Call? > if yes, how i can Play audio File unstid of the Microphone device? > thanks > > ------------------------------------------------------------------------------ > OpenSolaris 2009.06 is a cutting edge operating system for enterprises > looking to deploy the next generation of Solaris that includes the latest > innovations from Sun and the OpenSource community. Download a copy and > enjoy capabilities such as Networking, Storage and Virtualization. > Go to: http://p.sf.net/sfu/opensolaris-get > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > > ------------------------------------------------------------------------------ > OpenSolaris 2009.06 is a cutting edge operating system for enterprises > looking to deploy the next generation of Solaris that includes the latest > innovations from Sun and the OpenSource community. Download a copy and > enjoy capabilities such as Networking, Storage and Virtualization. > Go to: http://p.sf.net/sfu/opensolaris-get > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4187 (20090625) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4187 (20090625) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com |
|
From: Joegen B. <joe...@gm...> - 2009-06-25 01:10:57
|
Hi Mefta, inline ... -------------------------------------------------- From: "Meftah Tayeb" <tay...@gm...> Sent: Wednesday, June 24, 2009 5:00 PM To: <ope...@li...> Subject: [OpenSIPStack] Developing a Powerfull Softphone using ATLSip > hello, > i'm developing a Softphone using atlsip > is the only open source Licensed SIP/VoIp ActiveX Control > but i need to know some informations... > please i have 2 questions: > > > 1. SIP Transport: > > the SIP Protocol Support 3 difere transports: > > * TCP > * UDP > * TLS (Secure) > > ATLSip support all this 3 transport or UDP only? > if ATLSip support TLS, how i can get / use Security Certificate? > As of now, only UDP is 100% working in the library. TCP is work in progress but nobody is working on that side of the code currently. There is TSL support just yet. > > Profile Storage > > if i register to my Server and i close my softphone, i get difere ini > files: default.ini, and other log files > this files is required? Yes they are. ATLSIP uses those preserve its settings. > how i can store my SIP Profile including Voice codecs, audio devices, > registration settings? > is stored automatikaly or i must store it manualy? The should be stored automatically. > if anyone here want mor help about some PBX i'm here to help > OpenSipUsers for free including Freeswitch I am contemplating in using freeswitch as a codec transcoder for opensbc. I might need some advice soon. Thanks for offering your help. > if anyone want to help me, please use this informations to contact me: > > > My Informations: > > Full Name: Meftah Tayeb > Click Here to Mail Me <ad...@ne...> > Click here to drop me a SIP Call (SIP:Voi...@si...) > <sip:voi...@si...> > Skype: tayeb.meftah > yahoo: vsdev2006 > MSN: SQL...@ho... > > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4182 (20090624) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > ------------------------------------------------------------------------------ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > No virus found in this incoming message. > Checked by AVG - www.avg.com > Version: 8.5.374 / Virus Database: 270.12.90/2198 - Release Date: 06/23/09 > 17:54:00 > |
|
From: Meftah T. <tay...@gm...> - 2009-06-24 09:02:18
|
hello,
i'm developing a Softphone using atlsip
is the only open source Licensed SIP/VoIp ActiveX Control
but i need to know some informations...
please i have 2 questions:
1. SIP Transport:
the SIP Protocol Support 3 difere transports:
* TCP
* UDP
* TLS (Secure)
ATLSip support all this 3 transport or UDP only?
if ATLSip support TLS, how i can get / use Security Certificate?
Profile Storage
if i register to my Server and i close my softphone, i get difere ini
files: default.ini, and other log files
this files is required?
how i can store my SIP Profile including Voice codecs, audio devices,
registration settings?
is stored automatikaly or i must store it manualy?
if anyone here want mor help about some PBX i'm here to help
OpenSipUsers for free including Freeswitch
if anyone want to help me, please use this informations to contact me:
My Informations:
Full Name: Meftah Tayeb
Click Here to Mail Me <ad...@ne...>
Click here to drop me a SIP Call (SIP:Voi...@si...)
<sip:voi...@si...>
Skype: tayeb.meftah
yahoo: vsdev2006
MSN: SQL...@ho...
__________ Information from ESET NOD32 Antivirus, version of virus signature database 4182 (20090624) __________
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
|
|
From: Andre S. <eds...@ya...> - 2009-06-07 05:02:17
|
Yes there is transfer from one sip account to another and hold.
Auto answer will depend on you application.
________________________________
From: Meftah Tayeb <tay...@gm...>
To: ope...@li...
Sent: Saturday, June 6, 2009 4:25:34 PM
Subject: [OpenSIPStack] ATLSip:call transfer / hold and Auto answering
hello OpenSipStack users,
please cool you tel me if is it pocible to transfer calls using AtlSip?
also, is it pocible to hold the call?
------------
and is it pocible to auto answer the Call?
if yes, how i can Play audio File unstid of the Microphone device?
thanks
------------------------------------------------------------------------------
OpenSolaris 2009.06 is a cutting edge operating system for enterprises
looking to deploy the next generation of Solaris that includes the latest
innovations from Sun and the OpenSource community. Download a copy and
enjoy capabilities such as Networking, Storage and Virtualization.
Go to: http://p.sf.net/sfu/opensolaris-get
_______________________________________________
opensipstack-devel mailing list
ope...@li...
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
|
|
From: Meftah T. <tay...@gm...> - 2009-06-06 08:24:59
|
hello OpenSipStack users,
please cool you tel me if is it pocible to transfer calls using AtlSip?
also, is it pocible to hold the call?
------------
and is it pocible to auto answer the Call?
if yes, how i can Play audio File unstid of the Microphone device?
thanks
|
|
From: voice <vo...@ne...> - 2009-04-28 23:58:43
|
Hi Joegen sipX 4.0 was released today. Now i will update our vmWare/Vyatta/sipX/OSBC and let you know.... r ----- Original Message ----- From: "Joegen Baclor" <joe...@gm...> To: <ope...@li...> Sent: Saturday, April 25, 2009 9:25 PM Subject: Re: [OpenSIPStack] Multicast Message > This same question was asked in the sip-implementors list. > https://lists.su.se/archive/public/sipforum-discussion/msg01044.html > Perhaps it is best for you to bring this question there if you want more > authoritative answers from RFC 3261 authors why SIP is not wired to handle > multicasting. > > -------------------------------------------------- > From: "amin mosayyebzadeh" <sir...@gm...> > Sent: Saturday, April 25, 2009 4:23 PM > To: <ope...@li...> > Subject: Re: [OpenSIPStack] Multicast Message > > > you mean I can't use SIP for multicast messaging? then how we can > > initialize a multicast call in VoIP? I mean before sending a multicast > > VoIP message, we need to set up a connection, so if SIP is not > > usefull, how we can do that? > > > > On Sat, Apr 25, 2009 at 10:14 AM, Joegen Baclor <joe...@gm...> > > wrote: > >> I haven't tried binding OpenSIPStack to a multicast address. How do you > >> plan to use multicast in SIP? AFAIK, multicast in SIP is originally > >> intended to support registrar discovery but not for calls. > >> > >> -------------------------------------------------- > >> From: "amin mosayyebzadeh" <sir...@gm...> > >> Sent: Saturday, April 25, 2009 1:18 PM > >> To: <ope...@li...> > >> Subject: [OpenSIPStack] Multicast Message > >> > >>> Hi anybody, > >>> > >>> can anybody tell me how I can send a multicast message? do we need > >>> proxy server? or just sending a message to multicast address is > >>> enough? > >>> thanks. > >>> > >>> ------------------------------------------------------------------------ ------ > >>> Crystal Reports - New Free Runtime and 30 Day Trial > >>> Check out the new simplified licensign option that enables unlimited > >>> royalty-free distribution of the report engine for externally facing > >>> server and web deployment. > >>> http://p.sf.net/sfu/businessobjects > >>> _______________________________________________ > >>> opensipstack-devel mailing list > >>> ope...@li... > >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > >>> > >> > >> > >> > >>> > >>> No virus found in this incoming message. > >>> Checked by AVG - www.avg.com > >>> Version: 8.0.238 / Virus Database: 270.12.4/2079 - Release Date: > >>> 04/24/09 > >>> 19:04:00 > >>> > >> > >> ------------------------------------------------------------------------- ----- > >> Crystal Reports - New Free Runtime and 30 Day Trial > >> Check out the new simplified licensign option that enables unlimited > >> royalty-free distribution of the report engine for externally facing > >> server and web deployment. > >> http://p.sf.net/sfu/businessobjects > >> _______________________________________________ > >> opensipstack-devel mailing list > >> ope...@li... > >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > >> > > > > > > > > -- > > siramin056.blogfa.com > > > > -------------------------------------------------------------------------- ---- > > Crystal Reports - New Free Runtime and 30 Day Trial > > Check out the new simplified licensign option that enables unlimited > > royalty-free distribution of the report engine for externally facing > > server and web deployment. > > http://p.sf.net/sfu/businessobjects > > _______________________________________________ > > opensipstack-devel mailing list > > ope...@li... > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > > > > > > No virus found in this incoming message. > > Checked by AVG - www.avg.com > > Version: 8.0.238 / Virus Database: 270.12.4/2079 - Release Date: 04/24/09 > > 19:04:00 > > > > -------------------------------------------------------------------------- ---- > Crystal Reports - New Free Runtime and 30 Day Trial > Check out the new simplified licensign option that enables unlimited > royalty-free distribution of the report engine for externally facing > server and web deployment. > http://p.sf.net/sfu/businessobjects > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
|
From: Joegen B. <joe...@gm...> - 2009-04-26 02:36:32
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This same question was asked in the sip-implementors list. https://lists.su.se/archive/public/sipforum-discussion/msg01044.html Perhaps it is best for you to bring this question there if you want more authoritative answers from RFC 3261 authors why SIP is not wired to handle multicasting. -------------------------------------------------- From: "amin mosayyebzadeh" <sir...@gm...> Sent: Saturday, April 25, 2009 4:23 PM To: <ope...@li...> Subject: Re: [OpenSIPStack] Multicast Message > you mean I can't use SIP for multicast messaging? then how we can > initialize a multicast call in VoIP? I mean before sending a multicast > VoIP message, we need to set up a connection, so if SIP is not > usefull, how we can do that? > > On Sat, Apr 25, 2009 at 10:14 AM, Joegen Baclor <joe...@gm...> > wrote: >> I haven't tried binding OpenSIPStack to a multicast address. How do you >> plan to use multicast in SIP? AFAIK, multicast in SIP is originally >> intended to support registrar discovery but not for calls. >> >> -------------------------------------------------- >> From: "amin mosayyebzadeh" <sir...@gm...> >> Sent: Saturday, April 25, 2009 1:18 PM >> To: <ope...@li...> >> Subject: [OpenSIPStack] Multicast Message >> >>> Hi anybody, >>> >>> can anybody tell me how I can send a multicast message? do we need >>> proxy server? or just sending a message to multicast address is >>> enough? >>> thanks. >>> >>> ------------------------------------------------------------------------------ >>> Crystal Reports - New Free Runtime and 30 Day Trial >>> Check out the new simplified licensign option that enables unlimited >>> royalty-free distribution of the report engine for externally facing >>> server and web deployment. >>> http://p.sf.net/sfu/businessobjects >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >> >> >> >>> >>> No virus found in this incoming message. >>> Checked by AVG - www.avg.com >>> Version: 8.0.238 / Virus Database: 270.12.4/2079 - Release Date: >>> 04/24/09 >>> 19:04:00 >>> >> >> ------------------------------------------------------------------------------ >> Crystal Reports - New Free Runtime and 30 Day Trial >> Check out the new simplified licensign option that enables unlimited >> royalty-free distribution of the report engine for externally facing >> server and web deployment. >> http://p.sf.net/sfu/businessobjects >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > > > > -- > siramin056.blogfa.com > > ------------------------------------------------------------------------------ > Crystal Reports - New Free Runtime and 30 Day Trial > Check out the new simplified licensign option that enables unlimited > royalty-free distribution of the report engine for externally facing > server and web deployment. > http://p.sf.net/sfu/businessobjects > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > No virus found in this incoming message. > Checked by AVG - www.avg.com > Version: 8.0.238 / Virus Database: 270.12.4/2079 - Release Date: 04/24/09 > 19:04:00 > |
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From: amin m. <sir...@gm...> - 2009-04-25 08:23:22
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you mean I can't use SIP for multicast messaging? then how we can initialize a multicast call in VoIP? I mean before sending a multicast VoIP message, we need to set up a connection, so if SIP is not usefull, how we can do that? On Sat, Apr 25, 2009 at 10:14 AM, Joegen Baclor <joe...@gm...> wrote: > I haven't tried binding OpenSIPStack to a multicast address. How do you > plan to use multicast in SIP? AFAIK, multicast in SIP is originally > intended to support registrar discovery but not for calls. > > -------------------------------------------------- > From: "amin mosayyebzadeh" <sir...@gm...> > Sent: Saturday, April 25, 2009 1:18 PM > To: <ope...@li...> > Subject: [OpenSIPStack] Multicast Message > >> Hi anybody, >> >> can anybody tell me how I can send a multicast message? do we need >> proxy server? or just sending a message to multicast address is >> enough? >> thanks. >> >> ------------------------------------------------------------------------------ >> Crystal Reports - New Free Runtime and 30 Day Trial >> Check out the new simplified licensign option that enables unlimited >> royalty-free distribution of the report engine for externally facing >> server and web deployment. >> http://p.sf.net/sfu/businessobjects >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> > > > >> >> No virus found in this incoming message. >> Checked by AVG - www.avg.com >> Version: 8.0.238 / Virus Database: 270.12.4/2079 - Release Date: 04/24/09 >> 19:04:00 >> > > ------------------------------------------------------------------------------ > Crystal Reports - New Free Runtime and 30 Day Trial > Check out the new simplified licensign option that enables unlimited > royalty-free distribution of the report engine for externally facing > server and web deployment. > http://p.sf.net/sfu/businessobjects > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > -- siramin056.blogfa.com |
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From: Joegen B. <joe...@gm...> - 2009-04-25 06:44:36
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I haven't tried binding OpenSIPStack to a multicast address. How do you plan to use multicast in SIP? AFAIK, multicast in SIP is originally intended to support registrar discovery but not for calls. -------------------------------------------------- From: "amin mosayyebzadeh" <sir...@gm...> Sent: Saturday, April 25, 2009 1:18 PM To: <ope...@li...> Subject: [OpenSIPStack] Multicast Message > Hi anybody, > > can anybody tell me how I can send a multicast message? do we need > proxy server? or just sending a message to multicast address is > enough? > thanks. > > ------------------------------------------------------------------------------ > Crystal Reports - New Free Runtime and 30 Day Trial > Check out the new simplified licensign option that enables unlimited > royalty-free distribution of the report engine for externally facing > server and web deployment. > http://p.sf.net/sfu/businessobjects > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > No virus found in this incoming message. > Checked by AVG - www.avg.com > Version: 8.0.238 / Virus Database: 270.12.4/2079 - Release Date: 04/24/09 > 19:04:00 > |
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From: amin m. <sir...@gm...> - 2009-04-25 05:18:45
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Hi anybody, can anybody tell me how I can send a multicast message? do we need proxy server? or just sending a message to multicast address is enough? thanks. |
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From: Joegen B. <joe...@gm...> - 2009-04-23 00:31:41
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We can post the tutorial as part of the library distribution tree perhaps under contrib directory. We can also post it in the forum. Regarding the retransmission, that is per specs. See text from 3261. 17.1.2.1 Overview of the non-INVITE Transaction Non-INVITE transactions do not make use of ACK. They are simple request-response interactions. For unreliable transports, requests are retransmitted at an interval which starts at T1 and doubles until it hits T2. If a provisional response is received, retransmissions continue for unreliable transports, but at an interval of T2. The server transaction retransmits the last response it sent, which can be a provisional or final response, only when a retransmission of the request is received. This is why request retransmissions need to continue even after a provisional response; they are to ensure reliable delivery of the final response. *** Take note of "If a provisional response is received, retransmissions continue for unreliable transports, but at an interval of T2." *** -------------------------------------------------- From: "Jamieson, William W" <jam...@RL...> Sent: Thursday, April 23, 2009 6:20 AM To: "'Joegen Baclor'" <jb...@so...>; <ope...@li...> Subject: RE: [OpenSIPStack] MWI Notify > I got this working pretty well now. The only problem left is that the > stack seems to resend the Notify message after a code 100 (Trying). It > takes a couple seconds for the Dialogic to send OK after a Notify. Is > there an easy way to change the timer value for how long it waits before a > resend or is this something I should worry about (i.e., it won't speed up > the client side by changing the timer values)? > > Here's what I get: > ---> xx1.527s Request: Notify > <--- xx1.541s Status: 100 Trying > ---> xx2.011s Request: Notify > <--- xx2.016s Status: 100 Trying > <--- xx3.995s Status: 200 OK > > I've send it go as long as 3 seconds but I've never seen more than one > resend. > > Joegen, > > Where is the Tutorial you requested I write supposed to go, on the forum? > > -Bill > > -----Original Message----- > From: Joegen Baclor [mailto:joe...@gm...] > Sent: Tuesday, April 21, 2009 3:46 PM > To: ope...@li... > Subject: Re: [OpenSIPStack] MWI Notify > > Sorry my bad! Yes those headers must be in the body. I'll change the > sample app when I get the chance. > > -------------------------------------------------- > From: "Jamieson, William W" <jam...@RL...> > Sent: Wednesday, April 22, 2009 3:44 AM > To: <ope...@li...> > Subject: Re: [OpenSIPStack] MWI Notify > >> Yeah, I got it working a while ago by moving the msg-status-line to >> the body. I didn't need the msg-account or msg-summary-line. You were >> right. >> >> Thanks again, >> -Bill >> >> -----Original Message----- >> From: Matthias Dreißig [mailto:mdr...@gm...] >> Sent: Tuesday, April 21, 2009 11:46 AM >> To: ope...@li... >> Subject: Re: [OpenSIPStack] MWI Notify >> >> RFC3842 sample: >> >> NOTIFY sip:al...@al... SIP/2.0 >> To: <sip:al...@ex...>;tag=78923 >> From: <sip:al...@ex...>;tag=4442 >> Date: Mon, 10 Jul 2000 03:55:07 GMT >> Call-Id: 13...@al... >> CSeq: 20 NOTIFY >> Contact: <sip:al...@vm...> >> Event: message-summary >> Subscription-State: active >> Content-Type: application/simple-message-summary >> Content-Length: 99 >> (body part size = 99) >> Messages-Waiting: yes >> Message-Account: sip:al...@vm... >> Voice-Message: 2/8 (0/2) >> >> Your part >> >> NOTIFY sip:3768720@130.97.10.5:5060 SIP/2.0 >> From: >> <sip:tes...@op...>;tag=5385369e3efe18109d428376cc92421d >> To: sip:3768720@130.97.10.5:5060 >> Via: SIP/2.0/UDP >> 130.97.35.171:5070;iid=417;branch=z9hG4bK5385369e3efe18109d438376cc92421d;uas-addr=130.97.10.5 >> CSeq: 1 NOTIFY >> Call-ID: 5385369e-3efe-1810-9603-8376cc92421d >> Contact: <sip:130.97.35.171:5070> >> Event: message-summary >> User-Agent: Sample UA >> Max-Forwards: 70 >> Messages-Waiting: yes >> Message-Account: sip:test_account@localhost >> Voice-Message: 1/1 >> Content-Length: 0 >> (empty body part size = 0) >> >> For me this looks not the same. After the content length there comes >> the body. >> >> And here is the definition of body part >> http://tools.ietf.org/html/rfc3842#section-5.2. >> >> Regards, >> Matthias >> >> Jamieson, William W schrieb: >>> The RFC looks the same as what I get on the wire - at least the way >>> Joegen laid it out with the message-waiting, etc as Custom Headers. >>> It looks like what the RFC shows for the packet - including that they >>> show a non-zero Content-Length (99 in their example). >>> http://tools.ietf.org/html/rfc3842#section-4.1 >>> >>> Maybe I'm not seeing how that a body is defined in the text document >>> I have from dialogic and from the RFC. They both show a blank line >>> after the last header and before the three MWI parts (Message-Waiting, >>> etc). >>> I'll try adding them to the body and see what happens. >>> >>> Thanks, >>> -Bill >>> >>> -----Original Message----- >>> From: Matthias Dreißig [mailto:mdr...@gm...] >>> Sent: Tuesday, April 21, 2009 10:56 AM >>> To: ope...@li... >>> Subject: Re: [OpenSIPStack] MWI Notify >>> >>> Hi Bill, >>> >>> what I can tell you is that the content length describes the length >>> of the body of the NOTIFY message. And because this NOTIFY doesn't >>> have a body the content length is 0. When I look at the RFC 3842 the >>> information Message-Waiting, Message-Account,.. has to be as text in >>> the body of the NOTIFY message and not as headers in the message >>> (http://tools.ietf.org/html/rfc3842#section-3.5). >>> >>> There are also sample messages >>> (http://tools.ietf.org/html/rfc3842#section-4). >>> >>> Regards, >>> Matthias >>> >>> >>> Jamieson, William W schrieb: >>> >>>> I have it working somewhat now but I notice on the wire that the >>>> content-length is zero. It's supposed to be the size of all the >>>> custom headers from what I can tell from the RFC. Is there a >>>> standard way of setting this? Also, I'm not getting the notification >>>> from ClientSession::OnIncomingSIPMessage when the server sends back >>>> it's status 100 "Trying" packet (see capture below). >>>> >>>> Here's what went on the wire. It shows an Internal Server Error but >>>> I'm not sure it is from the content length being zero. In the last >>>> version of the MWI service, I wrote my own SIP commands to do this, >>>> I found that I had to use Invite to get it to work even though >>>> Dialogic Support was telling me that they accept Notify and they >>>> reminded me that Invite may go away since it is not the standard. I >>>> just assumed that it was because I had written my own SIP commands >>>> to do it. I'll keep at this and maybe try it against our Cisco CM to >>>> see how it behaves. >>>> >>>> >>>> >>>> No. Time Source Destination >>>> Protocol Info >>>> 488 09:02:49.359600 130.97.35.171 130.97.10.5 SIP >>>> Request: NOTIFY sip:3768720@130.97.10.5:5060 >>>> >>>> Frame 488 (586 bytes on wire, 586 bytes captured) Ethernet II, Src: >>>> Dell_4e:64:f8 (00:21:9b:4e:64:f8), Dst: Cisco_f3:26:45 >>>> (00:18:ba:f3:26:45) Internet Protocol, Src: 130.97.35.171 >>>> (130.97.35.171), Dst: 130.97.10.5 (130.97.10.5) User Datagram >>>> Protocol, Src Port: vtsas (5070), Dst Port: sip (5060) >>>> Source port: vtsas (5070) >>>> Destination port: sip (5060) >>>> Length: 552 >>>> Checksum: 0xeb12 [correct] >>>> Session Initiation Protocol >>>> Request-Line: NOTIFY sip:3768720@130.97.10.5:5060 SIP/2.0 >>>> Message Header >>>> From: >>>> <sip:tes...@op...>;tag=5385369e3efe18109d428376cc92421d >>>> To: sip:3768720@130.97.10.5:5060 >>>> Via: SIP/2.0/UDP >>>> 130.97.35.171:5070;iid=417;branch=z9hG4bK5385369e3efe18109d438376cc92421d;uas-addr=130.97.10.5 >>>> CSeq: 1 NOTIFY >>>> Call-ID: 5385369e-3efe-1810-9603-8376cc92421d >>>> Contact: <sip:130.97.35.171:5070> >>>> Event: message-summary >>>> User-Agent: Sample UA >>>> Max-Forwards: 70 >>>> Messages-Waiting: yes >>>> Message-Account: sip:test_account@localhost >>>> Voice-Message: 1/1 >>>> Content-Length: 0 >>>> >>>> No. Time Source Destination >>>> Protocol Info >>>> 489 09:02:49.374563 130.97.10.5 130.97.35.171 SIP >>>> Status: 100 Trying >>>> >>>> Frame 489 (399 bytes on wire, 399 bytes captured) Ethernet II, Src: >>>> Cisco_f3:26:45 (00:18:ba:f3:26:45), Dst: Dell_4e:64:f8 >>>> (00:21:9b:4e:64:f8) Internet Protocol, Src: 130.97.10.5 >>>> (130.97.10.5), >>>> Dst: 130.97.35.171 (130.97.35.171) User Datagram Protocol, Src Port: >>>> ibm-pps (1376), Dst Port: vtsas (5070) >>>> Source port: ibm-pps (1376) >>>> Destination port: vtsas (5070) >>>> Length: 365 >>>> Checksum: 0x2aa2 [correct] >>>> Session Initiation Protocol >>>> Status-Line: SIP/2.0 100 Trying >>>> Message Header >>>> >>>> From:<sip:tes...@op...>;tag=5385369e3efe18109d428376cc92421d >>>> To:sip:3768720@130.97.10.5:5060 >>>> Call-ID:5385369e-3efe-1810-9603-8376cc92421d >>>> CSeq:1 NOTIFY >>>> Server:PBX-IP Media Gateway/2.1 >>>> Via:SIP/2.0/UDP >>>> 130.97.35.171:5070;iid=417;branch=z9hG4bK5385369e3efe18109d438376cc92421d;uas-addr=130.97.10.5 >>>> Content-Length:0 >>>> >>>> No. Time Source Destination >>>> Protocol Info >>>> 490 09:02:49.375037 130.97.10.5 130.97.35.171 SIP >>>> Status: 500 Internal Server Error >>>> >>>> Frame 490 (443 bytes on wire, 443 bytes captured) Ethernet II, Src: >>>> Cisco_f3:26:45 (00:18:ba:f3:26:45), Dst: Dell_4e:64:f8 >>>> (00:21:9b:4e:64:f8) Internet Protocol, Src: 130.97.10.5 >>>> (130.97.10.5), >>>> Dst: 130.97.35.171 (130.97.35.171) User Datagram Protocol, Src Port: >>>> ibm-pps (1376), Dst Port: vtsas (5070) >>>> Source port: ibm-pps (1376) >>>> Destination port: vtsas (5070) >>>> Length: 409 >>>> Checksum: 0x8abc [correct] >>>> Session Initiation Protocol >>>> Status-Line: SIP/2.0 500 Internal Server Error >>>> Message Header >>>> >>>> From:<sip:tes...@op...>;tag=5385369e3efe18109d428376cc92421d >>>> To:sip:3768720@130.97.10.5:5060;tag=27313246313536410067E556 >>>> Call-ID:5385369e-3efe-1810-9603-8376cc92421d >>>> CSeq:1 NOTIFY >>>> Server:PBX-IP Media Gateway/2.1 >>>> Via:SIP/2.0/UDP >>>> 130.97.35.171:5070;iid=417;branch=z9hG4bK5385369e3efe18109d438376cc92421d;uas-addr=130.97.10.5 >>>> Content-Length:0 >>>> >>>> -----Original Message----- >>>> From: Joegen Baclor [mailto:joe...@gm...] >>>> Sent: Thursday, April 16, 2009 11:11 PM >>>> To: ope...@li... >>>> Subject: Re: [OpenSIPStack] MWI Notify >>>> >>>> Hi Bill, >>>> >>>> Download the latest OpenSIPStack from CVS. There should now be a >>>> sample/client_server_demo directory. It contains the skeleton code >>>> for client and server UA. I've used unsolicited MWI notify in the >>>> client to >>>> better serve your purpose. Here is the snippet from the Main() >>>> function. >>>> It basically creates two user-agents bound to 5060 and 5070 >>>> respectively and >>>> let the client (5070) send MWI notify to the server(5060). You can >>>> get >>>> the >>>> response in OnIncomingSIPMessage of the client session. I also >>>> pasted the >>>> snippet below. Feel free to submit a tutorial based on your >>>> experience >>>> with this code. >>>> >>>> HTH, >>>> Joegen >>>> >>>> ----------------------------- >>>> >>>> void Process::Main() >>>> { >>>> #if WIN32 >>>> SetConsoleCtrlHandler(Win32SignalHandler, TRUE); >>>> #endif >>>> >>>> /// Create the server and and bind it to port 5060 >>>> m_UAS = new UA(); >>>> m_UAS->GetDefaultProfile().GetTransportProfile().EnableUDP( 0, >>>> 5060 ); >>>> m_UAS->Initialize(); >>>> m_UAS->StartTransportThreads(); >>>> m_UAS->m_SM = new SessionManager( *m_UAS ); >>>> >>>> /// Create the client and bind it to 5070 >>>> m_UAC = new UA(); >>>> m_UAC->GetDefaultProfile().GetTransportProfile().EnableUDP( 0, >>>> 5070 ); >>>> m_UAC->Initialize(); >>>> m_UAC->StartTransportThreads(); >>>> m_UAC->m_SM = new SessionManager( *m_UAC ); >>>> >>>> OString sesionId = ParserTools::GenGUID(); >>>> ClientSession * session = dynamic_cast<ClientSession >>>> *>(m_UAC->m_SM->CreateClientSession( m_UAC->GetDefaultProfile(), >>>> sesionId )); >>>> SIPURI target( "sip:test_account@localhost:5060" ); >>>> SIPURI localURI( "sip:tes...@op..." ); >>>> Via localVia; >>>> m_UAC->ConstructVia( target.GetAddress(), localVia, >>>> SIPTransport::UDP ); >>>> >>>> session->SetTargetURI( target ); >>>> session->SetRemoteURI( target ); >>>> session->SetLocalURI( localURI ); >>>> session->SetLocalVia( localVia ); >>>> >>>> /// Create a notify and send a test message to the server UA >>>> SIPMessage notify; >>>> if( session->CreateRequestOutOfDialog( SIPMessage::Method_NOTIFY, >>>> notify ) ) >>>> { >>>> Event event( "message-summary" ); >>>> notify.SetEvent( event ); >>>> >>>> /// Add Event specific headers headers >>>> SIPHeader messagesWaiting( "Messages-Waiting", "yes" ); >>>> SIPHeader messageAccount( "Message-Account", >>>> "sip:test_account@localhost" ); >>>> SIPHeader voiceMessage( "Voice-Message", "1/1" ); >>>> notify.AddCustomHeader( messagesWaiting ); >>>> notify.AddCustomHeader( messageAccount ); >>>> notify.AddCustomHeader( voiceMessage ); >>>> >>>> session->SendRequest( notify ); >>>> } >>>> >>>> Sleep( 5000 ); >>>> >>>> m_UAC->Terminate(); >>>> m_UAS->Terminate(); >>>> >>>> delete m_UAC; >>>> delete m_UAS; >>>> } >>>> >>>> >>>> >>>> void ClientSession::OnIncomingSIPMessage( >>>> SIPMessageArrival & messageEvent >>>> ) >>>> { >>>> SIPMessage message = messageEvent.GetMessage(); >>>> if( message.IsResponse() ) >>>> { >>>> LOG( LogInfo(), "Got SIP Reponse" ); >>>> }else >>>> { >>>> LOG( LogInfo(), "Got SIP Request" ) >>>> SIPMessage response; >>>> message.CreateResponse( response, SIPMessage::Code200_Ok ); >>>> SendRequest( response ); >>>> } >>>> Destroy(); >>>> } >>>> >>>> >>>> -------------------------------------------------- >>>> From: "Jamieson, William W" <jam...@RL...> >>>> Sent: Friday, April 17, 2009 11:24 AM >>>> To: "'Joegen Baclor'" <jb...@so...>; >>>> <ope...@li...> >>>> Subject: Re: [OpenSIPStack] MWI Notify >>>> >>>> >>>> >>>>> That's great! >>>>> >>>>> Thanks. >>>>> >>>>> -Bill >>>>> >>>>> -----Original Message----- >>>>> From: Joegen Baclor [mailto:joe...@gm...] >>>>> Sent: Thursday, April 16, 2009 6:25 PM >>>>> To: ope...@li... >>>>> Subject: Re: [OpenSIPStack] MWI Notify >>>>> >>>>> Hey Jaimieson, >>>>> >>>>> I'll try to crank up a sample application for MWI as your basis. It >>>>> quite >>>>> time we put some sample tutorials in opensipstack. Give it several >>>>> hours. >>>>> >>>>> Joegen >>>>> >>>>> -------------------------------------------------- >>>>> From: "Jamieson, William W" <jam...@RL...> >>>>> Sent: Thursday, April 16, 2009 9:35 PM >>>>> To: <ope...@li...> >>>>> Subject: [OpenSIPStack] MWI Notify >>>>> >>>>> >>>>> >>>>>> I'm really after sending an MWI Notify and I'm using the >>>>>> MWIResource and MWIPackage classes. It's an unsolicited Notify so >>>>>> I can't see why I need a subscription but that's the way the >>>>>> classes were written, I think. >>>>>> >>>>>> For now, I have given up on the two MWI classes. This is what I have: >>>>>> >>>>>> Make a SIPMessage called notify which I fill with all the fields I >>>>>> see defined in RFC 3842 (MWI) ... (code omitted) >>>>>> >>>>>> Make a sessionManager >>>>>> >>>>>> B2BUserAgent ua("MWI"); >>>>>> RFC3265ClientManager sessionMgr(ua, "MWI", 1, 1024 * 2); >>>>>> >>>>>> Make a SIPSession: >>>>>> >>>>>> RFC3265ClientManager::SubscriptionInfo info("reg", fromURI, >>>>>> "application/reginfo+xml"); // stole this so not sure what the >>>>>> args >>>>>> are >>>>>> OString sessionId = ParserTools::GenGUID(); SIPSession *session = >>>>>> sessionMgr.CreateClientSession(info, sessionId); // now I have the >>>>>> sessionId for the notify CallId which I add to the notify >>>>>> SIPMessage >>>>>> >>>>>> Send the request >>>>>> >>>>>> sessionMgr.SendRequest(invite); >>>>>> >>>>>> Though it compiles, I know it's not right since I'm not getting >>>>>> anything out on the wire. Am I even close? I thought of using this >>>>>> instead of the call to SendRequest but it doesn't seem right to >>>>>> call an event procedure directly... >>>>>> >>>>>> sessionMgr.OnUnsolicitedNotification(invite); >>>>>> >>>>>> It's very frustrating to work with no samples or docs, it's like >>>>>> trying to reverse engineer the whole thing. Can someone steer me >>>>>> to where I might find these? >>>>>> >>>>>> billjam >>>>>> ------------------------------------------------------------------ >>>>>> - >>>>>> - >>>>>> - >>>>>> - >>>>>> -------- Stay on top of everything new and different, both inside >>>>>> and around Java (TM) technology - register by April 22, and save >>>>>> $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. >>>>>> 300 plus technical and hands-on sessions. Register today. >>>>>> Use priority code J9JMT32. http://p.sf.net/sfu/p >>>>>> _______________________________________________ >>>>>> opensipstack-devel mailing list >>>>>> ope...@li... >>>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>>> >>>>>> >>>>> >>>>>> No virus found in this incoming message. >>>>>> Checked by AVG - www.avg.com >>>>>> Version: 8.0.238 / Virus Database: 270.11.58/2061 - Release Date: >>>>>> 04/15/09 19:52:00 >>>>>> >>>>>> >>>>>> >>>>> ------------------------------------------------------------------- >>>>> - >>>>> - >>>>> - >>>>> -------- Stay on top of everything new and different, both inside >>>>> and around Java >>>>> (TM) technology - register by April 22, and save $200 on the >>>>> JavaOne >>>>> (SM) conference, June 2-5, 2009, San Francisco. >>>>> 300 plus technical and hands-on sessions. Register today. >>>>> Use priority code J9JMT32. http://p.sf.net/sfu/p >>>>> _______________________________________________ >>>>> opensipstack-devel mailing list >>>>> ope...@li... >>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>> >>>>> ------------------------------------------------------------------- >>>>> - >>>>> - >>>>> - >>>>> -------- Stay on top of everything new and different, both inside >>>>> and around Java (TM) technology - register by April 22, and save >>>>> $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. >>>>> 300 plus technical and hands-on sessions. Register today. >>>>> Use priority code J9JMT32. http://p.sf.net/sfu/p >>>>> _______________________________________________ >>>>> opensipstack-devel mailing list >>>>> ope...@li... >>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>> >>>>> >>>>> >>>> >>>> >>>> >>>>> No virus found in this incoming message. >>>>> Checked by AVG - www.avg.com >>>>> Version: 8.0.238 / Virus Database: 270.11.59/2063 - Release Date: >>>>> 04/16/09 16:38:00 >>>>> >>>>> >>>>> >>>> -------------------------------------------------------------------- >>>> - >>>> - >>>> -------- Stay on top of everything new and different, both inside >>>> and around Java (TM) technology - register by April 22, and save >>>> $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. >>>> 300 plus technical and hands-on sessions. Register today. >>>> Use priority code J9JMT32. http://p.sf.net/sfu/p >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> -------------------------------------------------------------------- >>>> - >>>> - >>>> -------- Stay on top of everything new and different, both inside >>>> and around Java (TM) technology - register by April 22, and save >>>> $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. >>>> 300 plus technical and hands-on sessions. Register today. >>>> Use priority code J9JMT32. http://p.sf.net/sfu/p >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> >>>> >>> >>> --------------------------------------------------------------------- >>> - >>> -------- Stay on top of everything new and different, both inside and >>> around Java (TM) technology - register by April 22, and save $200 on >>> the JavaOne (SM) conference, June 2-5, 2009, San Francisco. >>> 300 plus technical and hands-on sessions. Register today. >>> Use priority code J9JMT32. http://p.sf.net/sfu/p >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> --------------------------------------------------------------------- >>> - >>> -------- Stay on top of everything new and different, both inside and >>> around Java (TM) technology - register by April 22, and save $200 on >>> the JavaOne (SM) conference, June 2-5, 2009, San Francisco. >>> 300 plus technical and hands-on sessions. Register today. >>> Use priority code J9JMT32. http://p.sf.net/sfu/p >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >> >> ---------------------------------------------------------------------- >> -------- Stay on top of everything new and different, both inside and >> around Java >> (TM) technology - register by April 22, and save $200 on the JavaOne >> (SM) conference, June 2-5, 2009, San Francisco. >> 300 plus technical and hands-on sessions. Register today. >> Use priority code J9JMT32. http://p.sf.net/sfu/p >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> ---------------------------------------------------------------------- >> -------- Stay on top of everything new and different, both inside and >> around Java (TM) technology - register by April 22, and save $200 on >> the JavaOne (SM) conference, June 2-5, 2009, San Francisco. >> 300 plus technical and hands-on sessions. Register today. >> Use priority code J9JMT32. http://p.sf.net/sfu/p >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > >> >> No virus found in this incoming message. >> Checked by AVG - www.avg.com >> Version: 8.0.238 / Virus Database: 270.12.1/2070 - Release Date: >> 04/20/09 17:56:00 >> > > ------------------------------------------------------------------------------ > Stay on top of everything new and different, both inside and around Java > (TM) technology - register by April 22, and save $200 on the JavaOne (SM) > conference, June 2-5, 2009, San Francisco. > 300 plus technical and hands-on sessions. Register today. > Use priority code J9JMT32. http://p.sf.net/sfu/p > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > No virus found in this incoming message. > Checked by AVG - www.avg.com > Version: 8.0.238 / Virus Database: 270.12.2/2072 - Release Date: 04/21/09 > 16:48:00 > |
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From: Jamieson, W. W <jam...@RL...> - 2009-04-22 22:21:29
|
I got this working pretty well now. The only problem left is that the stack seems to resend the Notify message after a code 100 (Trying). It takes a couple seconds for the Dialogic to send OK after a Notify. Is there an easy way to change the timer value for how long it waits before a resend or is this something I should worry about (i.e., it won't speed up the client side by changing the timer values)? Here's what I get: ---> xx1.527s Request: Notify <--- xx1.541s Status: 100 Trying ---> xx2.011s Request: Notify <--- xx2.016s Status: 100 Trying <--- xx3.995s Status: 200 OK I've send it go as long as 3 seconds but I've never seen more than one resend. Joegen, Where is the Tutorial you requested I write supposed to go, on the forum? -Bill -----Original Message----- From: Joegen Baclor [mailto:joe...@gm...] Sent: Tuesday, April 21, 2009 3:46 PM To: ope...@li... Subject: Re: [OpenSIPStack] MWI Notify Sorry my bad! Yes those headers must be in the body. I'll change the sample app when I get the chance. -------------------------------------------------- From: "Jamieson, William W" <jam...@RL...> Sent: Wednesday, April 22, 2009 3:44 AM To: <ope...@li...> Subject: Re: [OpenSIPStack] MWI Notify > Yeah, I got it working a while ago by moving the msg-status-line to > the body. I didn't need the msg-account or msg-summary-line. You were right. > > Thanks again, > -Bill > > -----Original Message----- > From: Matthias Dreißig [mailto:mdr...@gm...] > Sent: Tuesday, April 21, 2009 11:46 AM > To: ope...@li... > Subject: Re: [OpenSIPStack] MWI Notify > > RFC3842 sample: > > NOTIFY sip:al...@al... SIP/2.0 > To: <sip:al...@ex...>;tag=78923 > From: <sip:al...@ex...>;tag=4442 > Date: Mon, 10 Jul 2000 03:55:07 GMT > Call-Id: 13...@al... > CSeq: 20 NOTIFY > Contact: <sip:al...@vm...> > Event: message-summary > Subscription-State: active > Content-Type: application/simple-message-summary > Content-Length: 99 > (body part size = 99) > Messages-Waiting: yes > Message-Account: sip:al...@vm... > Voice-Message: 2/8 (0/2) > > Your part > > NOTIFY sip:3768720@130.97.10.5:5060 SIP/2.0 > From: > <sip:tes...@op...>;tag=5385369e3efe18109d428376cc92421d > To: sip:3768720@130.97.10.5:5060 > Via: SIP/2.0/UDP > 130.97.35.171:5070;iid=417;branch=z9hG4bK5385369e3efe18109d438376cc92421d;uas-addr=130.97.10.5 > CSeq: 1 NOTIFY > Call-ID: 5385369e-3efe-1810-9603-8376cc92421d > Contact: <sip:130.97.35.171:5070> > Event: message-summary > User-Agent: Sample UA > Max-Forwards: 70 > Messages-Waiting: yes > Message-Account: sip:test_account@localhost > Voice-Message: 1/1 > Content-Length: 0 > (empty body part size = 0) > > For me this looks not the same. After the content length there comes > the body. > > And here is the definition of body part > http://tools.ietf.org/html/rfc3842#section-5.2. > > Regards, > Matthias > > Jamieson, William W schrieb: >> The RFC looks the same as what I get on the wire - at least the way >> Joegen laid it out with the message-waiting, etc as Custom Headers. >> It looks like what the RFC shows for the packet - including that they >> show a non-zero Content-Length (99 in their example). >> http://tools.ietf.org/html/rfc3842#section-4.1 >> >> Maybe I'm not seeing how that a body is defined in the text document >> I have from dialogic and from the RFC. They both show a blank line >> after the last header and before the three MWI parts (Message-Waiting, etc). >> I'll try adding them to the body and see what happens. >> >> Thanks, >> -Bill >> >> -----Original Message----- >> From: Matthias Dreißig [mailto:mdr...@gm...] >> Sent: Tuesday, April 21, 2009 10:56 AM >> To: ope...@li... >> Subject: Re: [OpenSIPStack] MWI Notify >> >> Hi Bill, >> >> what I can tell you is that the content length describes the length >> of the body of the NOTIFY message. And because this NOTIFY doesn't >> have a body the content length is 0. When I look at the RFC 3842 the >> information Message-Waiting, Message-Account,.. has to be as text in >> the body of the NOTIFY message and not as headers in the message >> (http://tools.ietf.org/html/rfc3842#section-3.5). >> >> There are also sample messages >> (http://tools.ietf.org/html/rfc3842#section-4). >> >> Regards, >> Matthias >> >> >> Jamieson, William W schrieb: >> >>> I have it working somewhat now but I notice on the wire that the >>> content-length is zero. It's supposed to be the size of all the >>> custom headers from what I can tell from the RFC. Is there a >>> standard way of setting this? Also, I'm not getting the notification >>> from ClientSession::OnIncomingSIPMessage when the server sends back >>> it's status 100 "Trying" packet (see capture below). >>> >>> Here's what went on the wire. It shows an Internal Server Error but >>> I'm not sure it is from the content length being zero. In the last >>> version of the MWI service, I wrote my own SIP commands to do this, >>> I found that I had to use Invite to get it to work even though >>> Dialogic Support was telling me that they accept Notify and they >>> reminded me that Invite may go away since it is not the standard. I >>> just assumed that it was because I had written my own SIP commands >>> to do it. I'll keep at this and maybe try it against our Cisco CM to see how it behaves. >>> >>> >>> >>> No. Time Source Destination >>> Protocol Info >>> 488 09:02:49.359600 130.97.35.171 130.97.10.5 SIP >>> Request: NOTIFY sip:3768720@130.97.10.5:5060 >>> >>> Frame 488 (586 bytes on wire, 586 bytes captured) Ethernet II, Src: >>> Dell_4e:64:f8 (00:21:9b:4e:64:f8), Dst: Cisco_f3:26:45 >>> (00:18:ba:f3:26:45) Internet Protocol, Src: 130.97.35.171 >>> (130.97.35.171), Dst: 130.97.10.5 (130.97.10.5) User Datagram >>> Protocol, Src Port: vtsas (5070), Dst Port: sip (5060) >>> Source port: vtsas (5070) >>> Destination port: sip (5060) >>> Length: 552 >>> Checksum: 0xeb12 [correct] >>> Session Initiation Protocol >>> Request-Line: NOTIFY sip:3768720@130.97.10.5:5060 SIP/2.0 >>> Message Header >>> From: >>> <sip:tes...@op...>;tag=5385369e3efe18109d428376cc92421d >>> To: sip:3768720@130.97.10.5:5060 >>> Via: SIP/2.0/UDP >>> 130.97.35.171:5070;iid=417;branch=z9hG4bK5385369e3efe18109d438376cc92421d;uas-addr=130.97.10.5 >>> CSeq: 1 NOTIFY >>> Call-ID: 5385369e-3efe-1810-9603-8376cc92421d >>> Contact: <sip:130.97.35.171:5070> >>> Event: message-summary >>> User-Agent: Sample UA >>> Max-Forwards: 70 >>> Messages-Waiting: yes >>> Message-Account: sip:test_account@localhost >>> Voice-Message: 1/1 >>> Content-Length: 0 >>> >>> No. Time Source Destination >>> Protocol Info >>> 489 09:02:49.374563 130.97.10.5 130.97.35.171 SIP >>> Status: 100 Trying >>> >>> Frame 489 (399 bytes on wire, 399 bytes captured) Ethernet II, Src: >>> Cisco_f3:26:45 (00:18:ba:f3:26:45), Dst: Dell_4e:64:f8 >>> (00:21:9b:4e:64:f8) Internet Protocol, Src: 130.97.10.5 >>> (130.97.10.5), >>> Dst: 130.97.35.171 (130.97.35.171) User Datagram Protocol, Src Port: >>> ibm-pps (1376), Dst Port: vtsas (5070) >>> Source port: ibm-pps (1376) >>> Destination port: vtsas (5070) >>> Length: 365 >>> Checksum: 0x2aa2 [correct] >>> Session Initiation Protocol >>> Status-Line: SIP/2.0 100 Trying >>> Message Header >>> >>> From:<sip:tes...@op...>;tag=5385369e3efe18109d428376cc92421d >>> To:sip:3768720@130.97.10.5:5060 >>> Call-ID:5385369e-3efe-1810-9603-8376cc92421d >>> CSeq:1 NOTIFY >>> Server:PBX-IP Media Gateway/2.1 >>> Via:SIP/2.0/UDP >>> 130.97.35.171:5070;iid=417;branch=z9hG4bK5385369e3efe18109d438376cc92421d;uas-addr=130.97.10.5 >>> Content-Length:0 >>> >>> No. Time Source Destination >>> Protocol Info >>> 490 09:02:49.375037 130.97.10.5 130.97.35.171 SIP >>> Status: 500 Internal Server Error >>> >>> Frame 490 (443 bytes on wire, 443 bytes captured) Ethernet II, Src: >>> Cisco_f3:26:45 (00:18:ba:f3:26:45), Dst: Dell_4e:64:f8 >>> (00:21:9b:4e:64:f8) Internet Protocol, Src: 130.97.10.5 >>> (130.97.10.5), >>> Dst: 130.97.35.171 (130.97.35.171) User Datagram Protocol, Src Port: >>> ibm-pps (1376), Dst Port: vtsas (5070) >>> Source port: ibm-pps (1376) >>> Destination port: vtsas (5070) >>> Length: 409 >>> Checksum: 0x8abc [correct] >>> Session Initiation Protocol >>> Status-Line: SIP/2.0 500 Internal Server Error >>> Message Header >>> >>> From:<sip:tes...@op...>;tag=5385369e3efe18109d428376cc92421d >>> To:sip:3768720@130.97.10.5:5060;tag=27313246313536410067E556 >>> Call-ID:5385369e-3efe-1810-9603-8376cc92421d >>> CSeq:1 NOTIFY >>> Server:PBX-IP Media Gateway/2.1 >>> Via:SIP/2.0/UDP >>> 130.97.35.171:5070;iid=417;branch=z9hG4bK5385369e3efe18109d438376cc92421d;uas-addr=130.97.10.5 >>> Content-Length:0 >>> >>> -----Original Message----- >>> From: Joegen Baclor [mailto:joe...@gm...] >>> Sent: Thursday, April 16, 2009 11:11 PM >>> To: ope...@li... >>> Subject: Re: [OpenSIPStack] MWI Notify >>> >>> Hi Bill, >>> >>> Download the latest OpenSIPStack from CVS. There should now be a >>> sample/client_server_demo directory. It contains the skeleton code >>> for client and server UA. I've used unsolicited MWI notify in the >>> client to >>> better serve your purpose. Here is the snippet from the Main() >>> function. >>> It basically creates two user-agents bound to 5060 and 5070 >>> respectively and >>> let the client (5070) send MWI notify to the server(5060). You can get >>> the >>> response in OnIncomingSIPMessage of the client session. I also >>> pasted the >>> snippet below. Feel free to submit a tutorial based on your experience >>> with this code. >>> >>> HTH, >>> Joegen >>> >>> ----------------------------- >>> >>> void Process::Main() >>> { >>> #if WIN32 >>> SetConsoleCtrlHandler(Win32SignalHandler, TRUE); >>> #endif >>> >>> /// Create the server and and bind it to port 5060 >>> m_UAS = new UA(); >>> m_UAS->GetDefaultProfile().GetTransportProfile().EnableUDP( 0, 5060 ); >>> m_UAS->Initialize(); >>> m_UAS->StartTransportThreads(); >>> m_UAS->m_SM = new SessionManager( *m_UAS ); >>> >>> /// Create the client and bind it to 5070 >>> m_UAC = new UA(); >>> m_UAC->GetDefaultProfile().GetTransportProfile().EnableUDP( 0, 5070 ); >>> m_UAC->Initialize(); >>> m_UAC->StartTransportThreads(); >>> m_UAC->m_SM = new SessionManager( *m_UAC ); >>> >>> OString sesionId = ParserTools::GenGUID(); >>> ClientSession * session = dynamic_cast<ClientSession >>> *>(m_UAC->m_SM->CreateClientSession( m_UAC->GetDefaultProfile(), >>> sesionId )); >>> SIPURI target( "sip:test_account@localhost:5060" ); >>> SIPURI localURI( "sip:tes...@op..." ); >>> Via localVia; >>> m_UAC->ConstructVia( target.GetAddress(), localVia, >>> SIPTransport::UDP ); >>> >>> session->SetTargetURI( target ); >>> session->SetRemoteURI( target ); >>> session->SetLocalURI( localURI ); >>> session->SetLocalVia( localVia ); >>> >>> /// Create a notify and send a test message to the server UA >>> SIPMessage notify; >>> if( session->CreateRequestOutOfDialog( SIPMessage::Method_NOTIFY, >>> notify ) ) >>> { >>> Event event( "message-summary" ); >>> notify.SetEvent( event ); >>> >>> /// Add Event specific headers headers >>> SIPHeader messagesWaiting( "Messages-Waiting", "yes" ); >>> SIPHeader messageAccount( "Message-Account", >>> "sip:test_account@localhost" ); >>> SIPHeader voiceMessage( "Voice-Message", "1/1" ); >>> notify.AddCustomHeader( messagesWaiting ); >>> notify.AddCustomHeader( messageAccount ); >>> notify.AddCustomHeader( voiceMessage ); >>> >>> session->SendRequest( notify ); >>> } >>> >>> Sleep( 5000 ); >>> >>> m_UAC->Terminate(); >>> m_UAS->Terminate(); >>> >>> delete m_UAC; >>> delete m_UAS; >>> } >>> >>> >>> >>> void ClientSession::OnIncomingSIPMessage( >>> SIPMessageArrival & messageEvent >>> ) >>> { >>> SIPMessage message = messageEvent.GetMessage(); >>> if( message.IsResponse() ) >>> { >>> LOG( LogInfo(), "Got SIP Reponse" ); >>> }else >>> { >>> LOG( LogInfo(), "Got SIP Request" ) >>> SIPMessage response; >>> message.CreateResponse( response, SIPMessage::Code200_Ok ); >>> SendRequest( response ); >>> } >>> Destroy(); >>> } >>> >>> >>> -------------------------------------------------- >>> From: "Jamieson, William W" <jam...@RL...> >>> Sent: Friday, April 17, 2009 11:24 AM >>> To: "'Joegen Baclor'" <jb...@so...>; >>> <ope...@li...> >>> Subject: Re: [OpenSIPStack] MWI Notify >>> >>> >>> >>>> That's great! >>>> >>>> Thanks. >>>> >>>> -Bill >>>> >>>> -----Original Message----- >>>> From: Joegen Baclor [mailto:joe...@gm...] >>>> Sent: Thursday, April 16, 2009 6:25 PM >>>> To: ope...@li... >>>> Subject: Re: [OpenSIPStack] MWI Notify >>>> >>>> Hey Jaimieson, >>>> >>>> I'll try to crank up a sample application for MWI as your basis. It >>>> quite >>>> time we put some sample tutorials in opensipstack. Give it several >>>> hours. >>>> >>>> Joegen >>>> >>>> -------------------------------------------------- >>>> From: "Jamieson, William W" <jam...@RL...> >>>> Sent: Thursday, April 16, 2009 9:35 PM >>>> To: <ope...@li...> >>>> Subject: [OpenSIPStack] MWI Notify >>>> >>>> >>>> >>>>> I'm really after sending an MWI Notify and I'm using the >>>>> MWIResource and MWIPackage classes. It's an unsolicited Notify so >>>>> I can't see why I need a subscription but that's the way the >>>>> classes were written, I think. >>>>> >>>>> For now, I have given up on the two MWI classes. This is what I have: >>>>> >>>>> Make a SIPMessage called notify which I fill with all the fields I >>>>> see defined in RFC 3842 (MWI) ... (code omitted) >>>>> >>>>> Make a sessionManager >>>>> >>>>> B2BUserAgent ua("MWI"); >>>>> RFC3265ClientManager sessionMgr(ua, "MWI", 1, 1024 * 2); >>>>> >>>>> Make a SIPSession: >>>>> >>>>> RFC3265ClientManager::SubscriptionInfo info("reg", fromURI, >>>>> "application/reginfo+xml"); // stole this so not sure what the args >>>>> are >>>>> OString sessionId = ParserTools::GenGUID(); SIPSession *session = >>>>> sessionMgr.CreateClientSession(info, sessionId); // now I have the >>>>> sessionId for the notify CallId which I add to the notify >>>>> SIPMessage >>>>> >>>>> Send the request >>>>> >>>>> sessionMgr.SendRequest(invite); >>>>> >>>>> Though it compiles, I know it's not right since I'm not getting >>>>> anything out on the wire. Am I even close? I thought of using this >>>>> instead of the call to SendRequest but it doesn't seem right to >>>>> call an event procedure directly... >>>>> >>>>> sessionMgr.OnUnsolicitedNotification(invite); >>>>> >>>>> It's very frustrating to work with no samples or docs, it's like >>>>> trying to reverse engineer the whole thing. Can someone steer me >>>>> to where I might find these? >>>>> >>>>> billjam >>>>> ------------------------------------------------------------------ >>>>> - >>>>> - >>>>> - >>>>> - >>>>> -------- Stay on top of everything new and different, both inside >>>>> and around Java (TM) technology - register by April 22, and save >>>>> $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. >>>>> 300 plus technical and hands-on sessions. Register today. >>>>> Use priority code J9JMT32. http://p.sf.net/sfu/p >>>>> _______________________________________________ >>>>> opensipstack-devel mailing list >>>>> ope...@li... >>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>> >>>>> >>>> >>>>> No virus found in this incoming message. >>>>> Checked by AVG - www.avg.com >>>>> Version: 8.0.238 / Virus Database: 270.11.58/2061 - Release Date: >>>>> 04/15/09 19:52:00 >>>>> >>>>> >>>>> >>>> ------------------------------------------------------------------- >>>> - >>>> - >>>> - >>>> -------- Stay on top of everything new and different, both inside >>>> and around Java >>>> (TM) technology - register by April 22, and save $200 on the >>>> JavaOne >>>> (SM) conference, June 2-5, 2009, San Francisco. >>>> 300 plus technical and hands-on sessions. Register today. >>>> Use priority code J9JMT32. http://p.sf.net/sfu/p >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> ------------------------------------------------------------------- >>>> - >>>> - >>>> - >>>> -------- Stay on top of everything new and different, both inside >>>> and around Java (TM) technology - register by April 22, and save >>>> $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. >>>> 300 plus technical and hands-on sessions. Register today. >>>> Use priority code J9JMT32. http://p.sf.net/sfu/p >>>> _______________________________________________ >>>> opensipstack-devel mailing list >>>> ope...@li... >>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>> >>>> >>>> >>> >>> >>> >>>> No virus found in this incoming message. >>>> Checked by AVG - www.avg.com >>>> Version: 8.0.238 / Virus Database: 270.11.59/2063 - Release Date: >>>> 04/16/09 16:38:00 >>>> >>>> >>>> >>> -------------------------------------------------------------------- >>> - >>> - >>> -------- Stay on top of everything new and different, both inside >>> and around Java (TM) technology - register by April 22, and save >>> $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. >>> 300 plus technical and hands-on sessions. Register today. >>> Use priority code J9JMT32. http://p.sf.net/sfu/p >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> -------------------------------------------------------------------- >>> - >>> - >>> -------- Stay on top of everything new and different, both inside >>> and around Java (TM) technology - register by April 22, and save >>> $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. >>> 300 plus technical and hands-on sessions. Register today. >>> Use priority code J9JMT32. http://p.sf.net/sfu/p >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> >> >> --------------------------------------------------------------------- >> - >> -------- Stay on top of everything new and different, both inside and >> around Java (TM) technology - register by April 22, and save $200 on >> the JavaOne (SM) conference, June 2-5, 2009, San Francisco. >> 300 plus technical and hands-on sessions. Register today. >> Use priority code J9JMT32. http://p.sf.net/sfu/p >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> --------------------------------------------------------------------- >> - >> -------- Stay on top of everything new and different, both inside and >> around Java (TM) technology - register by April 22, and save $200 on >> the JavaOne (SM) conference, June 2-5, 2009, San Francisco. >> 300 plus technical and hands-on sessions. Register today. >> Use priority code J9JMT32. http://p.sf.net/sfu/p >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> > > ---------------------------------------------------------------------- > -------- Stay on top of everything new and different, both inside and > around Java > (TM) technology - register by April 22, and save $200 on the JavaOne > (SM) conference, June 2-5, 2009, San Francisco. > 300 plus technical and hands-on sessions. Register today. > Use priority code J9JMT32. http://p.sf.net/sfu/p > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > ---------------------------------------------------------------------- > -------- Stay on top of everything new and different, both inside and > around Java (TM) technology - register by April 22, and save $200 on > the JavaOne (SM) conference, June 2-5, 2009, San Francisco. > 300 plus technical and hands-on sessions. Register today. > Use priority code J9JMT32. http://p.sf.net/sfu/p > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > No virus found in this incoming message. > Checked by AVG - www.avg.com > Version: 8.0.238 / Virus Database: 270.12.1/2070 - Release Date: > 04/20/09 17:56:00 > ------------------------------------------------------------------------------ Stay on top of everything new and different, both inside and around Java (TM) technology - register by April 22, and save $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. 300 plus technical and hands-on sessions. Register today. Use priority code J9JMT32. http://p.sf.net/sfu/p _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
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From: Jamieson, W. W <jam...@RL...> - 2009-04-22 14:19:53
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Thanks. I put the PSyncPoint in the ClientSession class of the demo so it was accessible to Main (through the session variable) and to the ClientSession functions (OnTimerExpire and OnIncomingSIPMessage). The Process class already had a PSyncPoint, it's used to signal when the process is terminated. -Bill -----Original Message----- From: Joegen Baclor [mailto:joe...@gm...] Sent: Tuesday, April 21, 2009 9:12 PM To: 'Joegen Baclor'; ope...@li... Subject: Re: [OpenSIPStack] SUBSCRIBE and MWI You can use a PSyncPoint as a member variable of the Process class. PSyncPoint m_ExitSync; . . . m_ExitSync.Wait(); // <--- Just signal this anywhere you like in the code. m_UAC->Terminate(); m_UAS->Terminate(); delete m_UAC; delete m_UAS; -------------------------------------------------- From: "Jamieson, William W" <jam...@RL...> Sent: Wednesday, April 22, 2009 8:04 AM To: "'Joegen Baclor'" <jb...@so...>; <ope...@li...> Subject: Re: [OpenSIPStack] SUBSCRIBE and MWI > No, I haven't needed to do subscribes with my MWI implementation. > > With the new MWI code that joegen put in samples, it there an event > that can be used in a WaitForXXX in place of the Sleep that he has - > like maybe the two events that fire OnTimerExpire and the client > OnIncomingSIPMessage? I know I could do CreateEvent() \ SetEvent() for > both but don't want to if I could grab existing ones to wait on - just > not sure how it's done. > > -Bill > > -----Original Message----- > From: Joegen Baclor [mailto:joe...@gm...] > Sent: Tuesday, April 21, 2009 3:47 PM > To: ope...@li... > Subject: Re: [OpenSIPStack] SUBSCRIBE and MWI > > No not yet. Perhaps Bill and Mathias could help it see the light. > > -------------------------------------------------- > From: "Ramu" <con...@gm...> > Sent: Wednesday, April 22, 2009 1:36 AM > To: <ope...@li...> > Subject: [OpenSIPStack] SUBSCRIBE and MWI > >> Hi, >> >> Does OpenSBC support SUBCRIBE message for MWI notification? >> >> Please do let know how can setup SBC to forward SUBSCRIBE message. >> We setup SBC in UpperRegistration mode. >> >> Thanks, >> Ramu >> --------------------------------------------------------------------- >> - >> -------- Stay on top of everything new and different, both inside and >> around Java (TM) technology - register by April 22, and save $200 on >> the JavaOne (SM) conference, June 2-5, 2009, San Francisco. >> 300 plus technical and hands-on sessions. Register today. >> Use priority code J9JMT32. http://p.sf.net/sfu/p >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > >> >> No virus found in this incoming message. >> Checked by AVG - www.avg.com >> Version: 8.0.238 / Virus Database: 270.12.1/2070 - Release Date: >> 04/20/09 17:56:00 >> > > ---------------------------------------------------------------------- > -------- Stay on top of everything new and different, both inside and > around Java > (TM) technology - register by April 22, and save $200 on the JavaOne > (SM) conference, June 2-5, 2009, San Francisco. > 300 plus technical and hands-on sessions. Register today. > Use priority code J9JMT32. http://p.sf.net/sfu/p > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > ---------------------------------------------------------------------- > -------- Stay on top of everything new and different, both inside and > around Java (TM) technology - register by April 22, and save $200 on > the JavaOne (SM) conference, June 2-5, 2009, San Francisco. > 300 plus technical and hands-on sessions. Register today. > Use priority code J9JMT32. http://p.sf.net/sfu/p > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > No virus found in this incoming message. > Checked by AVG - www.avg.com > Version: 8.0.238 / Virus Database: 270.12.1/2070 - Release Date: > 04/20/09 17:56:00 > ------------------------------------------------------------------------------ Stay on top of everything new and different, both inside and around Java (TM) technology - register by April 22, and save $200 on the JavaOne (SM) conference, June 2-5, 2009, San Francisco. 300 plus technical and hands-on sessions. Register today. Use priority code J9JMT32. http://p.sf.net/sfu/p _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
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