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From: <jo...@op...> - 2007-11-03 06:24:51
|
Yes, Anonymous access is supported. Make sure your authentication is set = to pserver: voice wrote: > Hi Joe > > CVS asks' for a password? is it aonocvs? > > r > > ----- Original Message -----=20 > From: "jo...@op..." <joe...@gm...> > To: <ope...@li...> > Sent: Thursday, November 01, 2007 9:45 PM > Subject: Re: [OpenSIPStack] Dialog events > > > voice wrote: > =20 >> Hi Joe >> >> The CVS is a version control website. I have installed openSBC from >> =20 > source > =20 >> and compiled it aok. Is this CVS going to update that install? >> =20 > > CVS will only get the source files. You will need to recompile it after= > the download and change your installed binary with the newly created on= e > > > =20 >> Does it >> over write existing source or does it update complied libraries etc...= =2E >> >> =20 > > > It will update existing source not the binaries. > > =20 >> Does it update contributions from others on this list and is it tested= >> before it is added to the CVS? >> =20 > > You are assured that all codes commited to CVS are tested code. > > > =20 >> And finally is it selective in what gets >> updated? >> >> =20 > > Yes > > =20 >> r >> >> >> ----- Original Message -----=20 >> From: "Joegen E. Baclor" <joe...@gm...> >> To: <ope...@li...> >> Cc: <jo...@op...> >> Sent: Wednesday, October 31, 2007 10:32 PM >> Subject: Re: [OpenSIPStack] Dialog events >> >> >> Instructions how to download from CVS can be found at >> http://www.opensipstack.org/cvs.html >> >> voice wrote: >> >> =20 >>> Hi Joe >>> >>> Could you please also send me that elaboration explaining steps to >>> >>> =20 >> download >> >> =20 >>> latest versions of the codes >>> using CVS >>> >>> Robert. >>> >>> >>> ----- Original Message -----=20 >>> From: "jo...@op..." <joe...@gm...> >>> To: <ope...@li...> >>> Sent: Wednesday, October 17, 2007 5:54 PM >>> Subject: Re: [OpenSIPStack] Dialog events >>> >>> >>> Of course. See how RFC3265Agent and RFC3680Package is used in >>> RegisterSessionManager. >>> >>> >>> sebastian pastor wrote: >>> >>> >>> =20 >>>> Thanks, Joegen. >>>> >>>> >>>> >>>> Anyway, I'd like to know a bit more=85 >>>> >>>> My aim is, as developer, add a *new service* to the phone (for accee= ding >>>> >>>> >>>> =20 >>> to >>> >>> >>> =20 >>>> media contents) and this will be based on exchange of SUBs and NTFYs= >>>> =20 > with > =20 >>>> event=3Ddialog but with a diferent xml body than the standard. >>>> >>>> >>>> >>>> So, do you think it would be feasible or *realistic* trying to progr= am >>>> >>>> >>>> =20 >>> this >>> >>> >>> =20 >>>> whole behaviour based on existent OpenSipStack code ? It would be an= >>>> >>>> =20 >> state >> >> =20 >>>> machine regarding to SUBSCRIBEs, renews, NTFYs, response codes=85 th= at I >>>> >>>> >>>> =20 >>> could >>> >>> >>> =20 >>>> catch from the REGISTER one. >>>> >>>> Or it's better waiting for you to have this functionality implemente= d? >>>> >>>> >>>> >>>> How much *time* do you estimate that we are talking about? >>>> >>>> >>>> >>>> Best Regards, >>>> >>>> >>>> >>>> -Sebasti=E1n Pastor- >>>> >>>> Junior Telecomunications Engineer (Malaga, >>>> =20 > Spain) > =20 >>>> >>>> p.s BTW, *Lalith *you are a bit off-topic. But i attached a quick do= c >>>> >>>> =20 >> that >> >> =20 >>>> i've elaborated explaining steps to download latest versions of the >>>> =20 > codes > =20 >>>> using CVS. Hope to help you. >>>> >>>> >>>> 2007/10/17, Joegen E. Baclor <joe...@gm...>: >>>> >>>> >>>> >>>> =20 >>>>> Hi Sebastian, >>>>> >>>>> I am sorry but this feature has not made it yet to the mile stone >>>>> release because SIP Trunking was deemed a more popular need of user= s of >>>>> OpenSBC. >>>>> >>>>> >>>>> >>>>> -------------------------------------------------------------------= ---- >>>>> =20 > - > =20 >>>>> -------------------------------------------------------------------= ---- >>>>> =20 > - > =20 >> - >> >> =20 >>>>> This SF.net email is sponsored by: Splunk Inc. >>>>> Still grepping through log files to find problems? Stop. >>>>> Now Search log events and configuration files using AJAX and a brow= ser. >>>>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>>>> -------------------------------------------------------------------= ---- >>>>> =20 > - > =20 >>>>> _______________________________________________ >>>>> opensipstack-devel mailing list >>>>> ope...@li... >>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>>>> >>>>> -------------------------------------------------------------------= ---- >>>>> =20 > - > =20 >>>>> No virus found in this incoming message. >>>>> Checked by AVG Free Edition. >>>>> Version: 7.5.488 / Virus Database: 269.14.13/1074 - Release Date: >>>>> >>>>> >>>>> =20 >>> 10/16/2007 2:14 PM >>> >>> >>> >>> >>> ---------------------------------------------------------------------= ---- >>> This SF.net email is sponsored by: Splunk Inc. >>> Still grepping through log files to find problems? Stop. >>> Now Search log events and configuration files using AJAX and a browse= r. >>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> >>> ---------------------------------------------------------------------= ---- >>> This SF.net email is sponsored by: Splunk Inc. >>> Still grepping through log files to find problems? Stop. >>> Now Search log events and configuration files using AJAX and a browse= r. >>> Download your FREE copy of Splunk now >> http://get.splunk.com/ >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> >>> >>> =20 >> >> ----------------------------------------------------------------------= --- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser= =2E >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> ----------------------------------------------------------------------= --- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser= =2E >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> >> =20 > > > > -----------------------------------------------------------------------= -- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser.= > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > -----------------------------------------------------------------------= -- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser.= > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > =20 |
|
From: <jo...@op...> - 2007-11-03 06:23:48
|
Are you using ATLSIP or the SoftPhoneInterface? If so, you can try uusing the approach below. If not, send in more information how you are actually using OpenSIPStack SIPMessage myMsg; //// Populate your SIP Message here m_SoftPhone->GetSIPEndPoint()->GetUserAgent()->SendRequest(myMsg); Woo Chen wrote: > I would like to add a SIP request similar to MESSAGE. The new request differs from MESSAGE only in their request names. The signalings for them are the same. > > I've spent a few days tracing the code, but still can't get any ideas. Can you give me some guidelines on how to modify the UA part of OpenSIPStack to achieve my goal? > > Regards, > GCC > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > |
|
From: voice <vo...@ne...> - 2007-11-02 15:46:47
|
Hi Joe CVS asks' for a password? is it aonocvs? r ----- Original Message ----- From: "jo...@op..." <joe...@gm...> To: <ope...@li...> Sent: Thursday, November 01, 2007 9:45 PM Subject: Re: [OpenSIPStack] Dialog events voice wrote: > Hi Joe > > The CVS is a version control website. I have installed openSBC from source > and compiled it aok. Is this CVS going to update that install? CVS will only get the source files. You will need to recompile it after the download and change your installed binary with the newly created one > Does it > over write existing source or does it update complied libraries etc.... > It will update existing source not the binaries. > Does it update contributions from others on this list and is it tested > before it is added to the CVS? You are assured that all codes commited to CVS are tested code. > And finally is it selective in what gets > updated? > Yes > r > > > ----- Original Message ----- > From: "Joegen E. Baclor" <joe...@gm...> > To: <ope...@li...> > Cc: <jo...@op...> > Sent: Wednesday, October 31, 2007 10:32 PM > Subject: Re: [OpenSIPStack] Dialog events > > > Instructions how to download from CVS can be found at > http://www.opensipstack.org/cvs.html > > voice wrote: > >> Hi Joe >> >> Could you please also send me that elaboration explaining steps to >> > download > >> latest versions of the codes >> using CVS >> >> Robert. >> >> >> ----- Original Message ----- >> From: "jo...@op..." <joe...@gm...> >> To: <ope...@li...> >> Sent: Wednesday, October 17, 2007 5:54 PM >> Subject: Re: [OpenSIPStack] Dialog events >> >> >> Of course. See how RFC3265Agent and RFC3680Package is used in >> RegisterSessionManager. >> >> >> sebastian pastor wrote: >> >> >>> Thanks, Joegen. >>> >>> >>> >>> Anyway, I'd like to know a bit more… >>> >>> My aim is, as developer, add a *new service* to the phone (for acceeding >>> >>> >> to >> >> >>> media contents) and this will be based on exchange of SUBs and NTFYs with >>> event=dialog but with a diferent xml body than the standard. >>> >>> >>> >>> So, do you think it would be feasible or *realistic* trying to program >>> >>> >> this >> >> >>> whole behaviour based on existent OpenSipStack code ? It would be an >>> > state > >>> machine regarding to SUBSCRIBEs, renews, NTFYs, response codes… that I >>> >>> >> could >> >> >>> catch from the REGISTER one. >>> >>> Or it's better waiting for you to have this functionality implemented? >>> >>> >>> >>> How much *time* do you estimate that we are talking about? >>> >>> >>> >>> Best Regards, >>> >>> >>> >>> -Sebastián Pastor- >>> >>> Junior Telecomunications Engineer (Malaga, Spain) >>> >>> >>> >>> p.s BTW, *Lalith *you are a bit off-topic. But i attached a quick doc >>> > that > >>> i've elaborated explaining steps to download latest versions of the codes >>> using CVS. Hope to help you. >>> >>> >>> 2007/10/17, Joegen E. Baclor <joe...@gm...>: >>> >>> >>> >>>> Hi Sebastian, >>>> >>>> I am sorry but this feature has not made it yet to the mile stone >>>> release because SIP Trunking was deemed a more popular need of users of >>>> OpenSBC. >>>> >>>> >>>> >>>> ----------------------------------------------------------------------- - >>>> >>>> ----------------------------------------------------------------------- - >>>> > - > >>>> This SF.net email is sponsored by: Splunk Inc. >>>> Still grepping through log files to find problems? Stop. >>>> Now Search log events and configuration files using AJAX and a browser. >>>> Download your FREE copy of Splunk now >> http://get.splunk.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 Free Edition. >>>> Version: 7.5.488 / Virus Database: 269.14.13/1074 - Release Date: >>>> >>>> >> 10/16/2007 2:14 PM >> >> >> >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser. >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser. >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
|
From: Woo C. <vir...@in...> - 2007-11-02 10:29:10
|
I would like to add a SIP request similar to MESSAGE. The new request = differs from MESSAGE only in their request names. The signalings for them = are the same. I've spent a few days tracing the code, but still can't get any ideas. = Can you give me some guidelines on how to modify the UA part of = OpenSIPStack to achieve my goal? Regards, GCC =20 |
|
From: <jo...@op...> - 2007-11-02 03:45:41
|
voice wrote: > Hi Joe > > The CVS is a version control website. I have installed openSBC from so= urce > and compiled it aok. Is this CVS going to update that install? CVS will only get the source files. You will need to recompile it after=20 the download and change your installed binary with the newly created one > Does it > over write existing source or does it update complied libraries etc....= > =20 It will update existing source not the binaries. > Does it update contributions from others on this list and is it tested > before it is added to the CVS? You are assured that all codes commited to CVS are tested code. > And finally is it selective in what gets > updated? > =20 Yes > r > > > ----- Original Message -----=20 > From: "Joegen E. Baclor" <joe...@gm...> > To: <ope...@li...> > Cc: <jo...@op...> > Sent: Wednesday, October 31, 2007 10:32 PM > Subject: Re: [OpenSIPStack] Dialog events > > > Instructions how to download from CVS can be found at > http://www.opensipstack.org/cvs.html > > voice wrote: > =20 >> Hi Joe >> >> Could you please also send me that elaboration explaining steps to >> =20 > download > =20 >> latest versions of the codes >> using CVS >> >> Robert. >> >> >> ----- Original Message -----=20 >> From: "jo...@op..." <joe...@gm...> >> To: <ope...@li...> >> Sent: Wednesday, October 17, 2007 5:54 PM >> Subject: Re: [OpenSIPStack] Dialog events >> >> >> Of course. See how RFC3265Agent and RFC3680Package is used in >> RegisterSessionManager. >> >> >> sebastian pastor wrote: >> >> =20 >>> Thanks, Joegen. >>> >>> >>> >>> Anyway, I'd like to know a bit more=85 >>> >>> My aim is, as developer, add a *new service* to the phone (for acceed= ing >>> >>> =20 >> to >> >> =20 >>> media contents) and this will be based on exchange of SUBs and NTFYs = with >>> event=3Ddialog but with a diferent xml body than the standard. >>> >>> >>> >>> So, do you think it would be feasible or *realistic* trying to progra= m >>> >>> =20 >> this >> >> =20 >>> whole behaviour based on existent OpenSipStack code ? It would be an >>> =20 > state > =20 >>> machine regarding to SUBSCRIBEs, renews, NTFYs, response codes=85 tha= t I >>> >>> =20 >> could >> >> =20 >>> catch from the REGISTER one. >>> >>> Or it's better waiting for you to have this functionality implemented= ? >>> >>> >>> >>> How much *time* do you estimate that we are talking about? >>> >>> >>> >>> Best Regards, >>> >>> >>> >>> -Sebasti=E1n Pastor- >>> >>> Junior Telecomunications Engineer (Malaga, Sp= ain) >>> >>> >>> >>> p.s BTW, *Lalith *you are a bit off-topic. But i attached a quick doc= >>> =20 > that > =20 >>> i've elaborated explaining steps to download latest versions of the c= odes >>> using CVS. Hope to help you. >>> >>> >>> 2007/10/17, Joegen E. Baclor <joe...@gm...>: >>> >>> >>> =20 >>>> Hi Sebastian, >>>> >>>> I am sorry but this feature has not made it yet to the mile stone >>>> release because SIP Trunking was deemed a more popular need of users= of >>>> OpenSBC. >>>> >>>> >>>> >>>> --------------------------------------------------------------------= ---- >>>> >>>> --------------------------------------------------------------------= ---- >>>> =20 > - > =20 >>>> This SF.net email is sponsored by: Splunk Inc. >>>> Still grepping through log files to find problems? Stop. >>>> Now Search log events and configuration files using AJAX and a brows= er. >>>> Download your FREE copy of Splunk now >> http://get.splunk.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 Free Edition. >>>> Version: 7.5.488 / Virus Database: 269.14.13/1074 - Release Date: >>>> >>>> =20 >> 10/16/2007 2:14 PM >> >> >> >> >> ----------------------------------------------------------------------= --- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser= =2E >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> ----------------------------------------------------------------------= --- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser= =2E >> Download your FREE copy of Splunk now >> http://get.splunk.com/ >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> =20 > > > > -----------------------------------------------------------------------= -- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser.= > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > -----------------------------------------------------------------------= -- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser.= > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > =20 |
|
From: voice <vo...@ne...> - 2007-11-01 15:06:57
|
Hi Joe The CVS is a version control website. I have installed openSBC from source and compiled it aok. Is this CVS going to update that install? Does it over write existing source or does it update complied libraries etc.... Does it update contributions from others on this list and is it tested before it is added to the CVS? And finally is it selective in what gets updated? r ----- Original Message ----- From: "Joegen E. Baclor" <joe...@gm...> To: <ope...@li...> Cc: <jo...@op...> Sent: Wednesday, October 31, 2007 10:32 PM Subject: Re: [OpenSIPStack] Dialog events Instructions how to download from CVS can be found at http://www.opensipstack.org/cvs.html voice wrote: > Hi Joe > > Could you please also send me that elaboration explaining steps to download > latest versions of the codes > using CVS > > Robert. > > > ----- Original Message ----- > From: "jo...@op..." <joe...@gm...> > To: <ope...@li...> > Sent: Wednesday, October 17, 2007 5:54 PM > Subject: Re: [OpenSIPStack] Dialog events > > > Of course. See how RFC3265Agent and RFC3680Package is used in > RegisterSessionManager. > > > sebastian pastor wrote: > >> Thanks, Joegen. >> >> >> >> Anyway, I'd like to know a bit more… >> >> My aim is, as developer, add a *new service* to the phone (for acceeding >> > to > >> media contents) and this will be based on exchange of SUBs and NTFYs with >> event=dialog but with a diferent xml body than the standard. >> >> >> >> So, do you think it would be feasible or *realistic* trying to program >> > this > >> whole behaviour based on existent OpenSipStack code ? It would be an state >> machine regarding to SUBSCRIBEs, renews, NTFYs, response codes… that I >> > could > >> catch from the REGISTER one. >> >> Or it's better waiting for you to have this functionality implemented? >> >> >> >> How much *time* do you estimate that we are talking about? >> >> >> >> Best Regards, >> >> >> >> -Sebastián Pastor- >> >> Junior Telecomunications Engineer (Malaga, Spain) >> >> >> >> p.s BTW, *Lalith *you are a bit off-topic. But i attached a quick doc that >> i've elaborated explaining steps to download latest versions of the codes >> using CVS. Hope to help you. >> >> >> 2007/10/17, Joegen E. Baclor <joe...@gm...>: >> >> >>> Hi Sebastian, >>> >>> I am sorry but this feature has not made it yet to the mile stone >>> release because SIP Trunking was deemed a more popular need of users of >>> OpenSBC. >>> >>> >>> >>> ------------------------------------------------------------------------ >>> >>> ------------------------------------------------------------------------ - >>> This SF.net email is sponsored by: Splunk Inc. >>> Still grepping through log files to find problems? Stop. >>> Now Search log events and configuration files using AJAX and a browser. >>> Download your FREE copy of Splunk now >> http://get.splunk.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 Free Edition. >>> Version: 7.5.488 / Virus Database: 269.14.13/1074 - Release Date: >>> > 10/16/2007 2:14 PM > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
|
From: <jo...@op...> - 2007-11-01 06:43:00
|
See http://www.opensipstack.org/sbc_man_install.html#2.2 voice wrote: > Hi Joe > > Is it possible to run openSBC as a background Daemon? > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > |
|
From: Joegen E. B. <joe...@gm...> - 2007-11-01 04:36:33
|
Please direct this email to sa...@so.... Commercial support is not supposed to be discussed in the development mailing list. voice wrote: > Hi Joe > > So what am i buying if you purchase your > > "24x7 OpenSBC: Cost: $150 per instance/per month" > > Do i have to buy any of your equipment? > > Can we use our own? > > The devil is always in the details. > > "For a small monthly fee, the Solegy Service Control Center will remotely > manage an instance of OpenSBC running on your network. All configuration > issues, upgrades, etc. will be handled by the Solegy SCC staff. Get all of > the benefits of a scalable and resilient session border controller without > worrying about the application itself" > > Will you configure it to work with our sipXecs Proxy? > > r. > > > > > |
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From: Joegen E. B. <joe...@gm...> - 2007-11-01 04:32:19
|
Instructions how to download from CVS can be found at=20 http://www.opensipstack.org/cvs.html voice wrote: > Hi Joe > > Could you please also send me that elaboration explaining steps to down= load > latest versions of the codes > using CVS > > Robert. > > > ----- Original Message -----=20 > From: "jo...@op..." <joe...@gm...> > To: <ope...@li...> > Sent: Wednesday, October 17, 2007 5:54 PM > Subject: Re: [OpenSIPStack] Dialog events > > > Of course. See how RFC3265Agent and RFC3680Package is used in > RegisterSessionManager. > > > sebastian pastor wrote: > =20 >> Thanks, Joegen. >> >> >> >> Anyway, I'd like to know a bit more=85 >> >> My aim is, as developer, add a *new service* to the phone (for acceedi= ng >> =20 > to > =20 >> media contents) and this will be based on exchange of SUBs and NTFYs w= ith >> event=3Ddialog but with a diferent xml body than the standard. >> >> >> >> So, do you think it would be feasible or *realistic* trying to program= >> =20 > this > =20 >> whole behaviour based on existent OpenSipStack code ? It would be an s= tate >> machine regarding to SUBSCRIBEs, renews, NTFYs, response codes=85 that= I >> =20 > could > =20 >> catch from the REGISTER one. >> >> Or it's better waiting for you to have this functionality implemented?= >> >> >> >> How much *time* do you estimate that we are talking about? >> >> >> >> Best Regards, >> >> >> >> -Sebasti=E1n Pastor- >> >> Junior Telecomunications Engineer (Malaga, Spa= in) >> >> >> >> p.s BTW, *Lalith *you are a bit off-topic. But i attached a quick doc = that >> i've elaborated explaining steps to download latest versions of the co= des >> using CVS. Hope to help you. >> >> >> 2007/10/17, Joegen E. Baclor <joe...@gm...>: >> >> =20 >>> Hi Sebastian, >>> >>> I am sorry but this feature has not made it yet to the mile stone >>> release because SIP Trunking was deemed a more popular need of users = of >>> OpenSBC. >>> >>> >>> >>> ---------------------------------------------------------------------= --- >>> >>> ---------------------------------------------------------------------= ---- >>> This SF.net email is sponsored by: Splunk Inc. >>> Still grepping through log files to find problems? Stop. >>> Now Search log events and configuration files using AJAX and a browse= r. >>> Download your FREE copy of Splunk now >> http://get.splunk.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 Free Edition. >>> Version: 7.5.488 / Virus Database: 269.14.13/1074 - Release Date: >>> =20 > 10/16/2007 2:14 PM > =20 > > > > -----------------------------------------------------------------------= -- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser.= > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > -----------------------------------------------------------------------= -- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser.= > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > =20 |
|
From: Joegen E. B. <joe...@gm...> - 2007-11-01 04:30:19
|
The basic answer is to use ethereal and sniff the packets coming in and out of your public interface. If you are looking for a user interface where you can see traffic activity in a particular NIC in OpenSBC http admin, this functionality is not present in OpenSBC. voice wrote: > Hi Joe > > My openSBC is on a two (2) ethernet box. > > One NIC is a node on an intranet (dhcp) network running a sipx proxy. > > OpenSBC binds to both NIC's. > > I am using openSBC NIC (dhcp) as a gateway to the Internet. Is there some > way to monitor from openSBC sipX traffic as it goes through openSBC to the > second NIC facing the Internet? > > r > > > > > |
|
From: voice <vo...@ne...> - 2007-10-31 19:43:04
|
Hi Joe So what am i buying if you purchase your "24x7 OpenSBC: Cost: $150 per instance/per month" Do i have to buy any of your equipment? Can we use our own? The devil is always in the details. "For a small monthly fee, the Solegy Service Control Center will remotely manage an instance of OpenSBC running on your network. All configuration issues, upgrades, etc. will be handled by the Solegy SCC staff. Get all of the benefits of a scalable and resilient session border controller without worrying about the application itself" Will you configure it to work with our sipXecs Proxy? r. |
|
From: voice <vo...@ne...> - 2007-10-31 19:10:41
|
Hi Joe Could you please also send me that elaboration explaining steps to download latest versions of the codes using CVS Robert. ----- Original Message ----- From: "jo...@op..." <joe...@gm...> To: <ope...@li...> Sent: Wednesday, October 17, 2007 5:54 PM Subject: Re: [OpenSIPStack] Dialog events Of course. See how RFC3265Agent and RFC3680Package is used in RegisterSessionManager. sebastian pastor wrote: > Thanks, Joegen. > > > > Anyway, I'd like to know a bit more… > > My aim is, as developer, add a *new service* to the phone (for acceeding to > media contents) and this will be based on exchange of SUBs and NTFYs with > event=dialog but with a diferent xml body than the standard. > > > > So, do you think it would be feasible or *realistic* trying to program this > whole behaviour based on existent OpenSipStack code ? It would be an state > machine regarding to SUBSCRIBEs, renews, NTFYs, response codes… that I could > catch from the REGISTER one. > > Or it's better waiting for you to have this functionality implemented? > > > > How much *time* do you estimate that we are talking about? > > > > Best Regards, > > > > -Sebastián Pastor- > > Junior Telecomunications Engineer (Malaga, Spain) > > > > p.s BTW, *Lalith *you are a bit off-topic. But i attached a quick doc that > i've elaborated explaining steps to download latest versions of the codes > using CVS. Hope to help you. > > > 2007/10/17, Joegen E. Baclor <joe...@gm...>: > >> Hi Sebastian, >> >> I am sorry but this feature has not made it yet to the mile stone >> release because SIP Trunking was deemed a more popular need of users of >> OpenSBC. >> >> >> >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Splunk Inc. >> Still grepping through log files to find problems? Stop. >> Now Search log events and configuration files using AJAX and a browser. >> Download your FREE copy of Splunk now >> http://get.splunk.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 Free Edition. >> Version: 7.5.488 / Virus Database: 269.14.13/1074 - Release Date: 10/16/2007 2:14 PM >> ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
|
From: voice <vo...@ne...> - 2007-10-31 17:00:21
|
Hi Joe My openSBC is on a two (2) ethernet box. One NIC is a node on an intranet (dhcp) network running a sipx proxy. OpenSBC binds to both NIC's. I am using openSBC NIC (dhcp) as a gateway to the Internet. Is there some way to monitor from openSBC sipX traffic as it goes through openSBC to the second NIC facing the Internet? r |
|
From: voice <vo...@ne...> - 2007-10-31 16:51:59
|
Hi Joe Is it possible to run openSBC as a background Daemon? |
|
From: Ilian J. C. P. <ip...@so...> - 2007-10-30 06:12:50
|
Hi,
Whit Thiele wrote:
> Hey guys,
>
> Is there a way to automatically set the ATLSIP library to use a specific
> codec and get rid of all the other selections?
Currently, there is no clean way to get rid of other codecs. What you
are doing here:
> Do I just set AudioCodec0 =
> "codec_name" and that's it?
>
however should be enough. This sets the default highest priority codec
(0 being the highest priority).
> I've set the following in code:
>
> ATLSIP.AudioCodec0 = "G.711-uLaw-64k";
> ATLSIP.AudioCodec1 = "G.711-uLaw-64k";
>
>
> Is this the correct way to do this? I find ATLSIP is defaulting to gsm in
> some cases and I'm not sure why.
>
Yes this is correct. Defaulting GSM to may be happening because no
codecs were set prior to calling InitializeSIP(). This happens in
SoftPhoneInterface::Initialize():
...
for( PINDEX i = 0; i < 20; i++ )
{
PStringStream codecSlot;
codecSlot << "Codec" << i;
OString codec = m_Config->GetString( "Audio", codecSlot, "" );
if( !codec.IsEmpty() )
formatMask.AppendString( codec );
}
if( formatMask.GetSize() == 0 )
{
m_Config->SetString( "Audio", "Codec0", "iLBC-13k3" );
formatMask.AppendString( "iLBC-13k3" );
m_Config->SetString( "Audio", "Codec1", "GSM-06.10" );
formatMask.AppendString( "GSM-06.10" );
}
...
Make sure that you have set ATLSIP.AudioCodecX before calling
InitializeSIP().
Regards,
Ilian
> Whit
>
>
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems? Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >> http://get.splunk.com/
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>
|
|
From: voice <vo...@ne...> - 2007-10-30 05:44:46
|
Hi Joegen I have several questions but first let me discribe my network I have installed sipXecs box with Centos5 DHCP, DNS and NTP static192.168.1.2 . Gateway 192.168.1.1 netmask 255.255.0.0 I have installed openSBC box with two NIC's one with an 1st NIC static 192.168.1.3 c Gateway 192.168.1.1 netmask 255.255.0.0 2nd NIC static 69.x.x.241 public gateway 69.x.x.1 netmask 255.255.255.0 I am able to remotely config openSBC over the Internet and from the private netwrok from the openSBC box configure the sipXecs box I have set the openSBC mode to B2BUpperReg. Static RTP Media address to 192.168.1.2 Now openSBC should bind it's listeners to both NIC's. Additionally i have these settings in B2BUA i have [sip:*@:sipx.domain.net:*] sip:192.168.1.2:5060 which is pointing to a user account on the sipX Relay Routes [sip:*@:sipx.domain.net:*] sip:192.168.1.2:5060 Internal DNS Mapping [sip:*@:sipx.domain.net:*] sip:192.168.1.2:5060 Upper Registration Router [sip:sipx.domain.net*] sip:192.168.1.2:5060 domain.net is a placeholder for a read domain. The goal is to have openSBC handoff to sipX and vise-versa to handel the sipX proxy services. I want sipX to register external Users, phones and dialplans. Does the B2BUA, Relay routes, DNS Mapping and Upper registration Routes set properly to get the job done? Have I address everything needed? The remote phones TFTP are pointed to sipX as well as their OutBoundProxy plus there phone numbers username and password. Internally the sipX gateway setting is pointed to the openSBC box. Maybe the questions & answers could be posted to the wiki in the future. r |
|
From: Whit T. <de...@wh...> - 2007-10-29 13:49:29
|
I've done some more reading on the 491 issue. Looks like it's a problem with asterisk. http://bugs.digium.com/view.php?id=9431 Whit -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Joegen E. Baclor Sent: Thursday, October 25, 2007 9:25 PM To: ope...@li... Subject: Re: [OpenSIPStack] 491 Request Pending Request Pending Error is normally sent back by a server if it received a new INVITE transaction while there is still an existing INVITE transaction pending which has not been responded to with a final response. Whit Thiele wrote: > Hey Guys, > > I've been having some trouble with dialing to Asterisk (v.1.4.9) with my > ATLSIP Softphone. I am able to dial most of the time with no trouble > although I am getting "491 Request Pending" messages from the > ATLSIP_OnOutgoingCallRejected event randomly say 30% of calls launched. > > I've read that asterisk has had some trouble dealing with 491 errors but I > thought I'd ask the forum: > > 1. Has anyone else has had any problems with this? > > 2. Anyone know the best way to avoid this from happening. > > > It's a simple setup with the Softphone and Asterisk on the same local > network. > > ATLSIP --> Asterisk --> PRI --> PSTN > > > Thanks, > > Whit > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > ------------------------------------------------------------------------- This SF.net email is sponsored by: Splunk Inc. Still grepping through log files to find problems? Stop. Now Search log events and configuration files using AJAX and a browser. Download your FREE copy of Splunk now >> http://get.splunk.com/ _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
|
From: Whit T. <de...@wh...> - 2007-10-26 17:18:27
|
Hey guys, Is there a way to automatically set the ATLSIP library to use a specific codec and get rid of all the other selections? Do I just set AudioCodec0 = "codec_name" and that's it? I've set the following in code: ATLSIP.AudioCodec0 = "G.711-uLaw-64k"; ATLSIP.AudioCodec1 = "G.711-uLaw-64k"; Is this the correct way to do this? I find ATLSIP is defaulting to gsm in some cases and I'm not sure why. Whit |
|
From: Ilian J. C. P. <ip...@so...> - 2007-10-26 05:41:59
|
Hi Whit,
No problem with that. Just keep in mind that you will lose ringing
totally when calling UAs that don't generate their own ringing tones.
Thanks.
- Ilian
Whit Thiele wrote:
> Illian,
>
> Thanks for your information! In my particular instance, it was easier to
> temporarily disable the local ringing sound because I think my pbx
> (asterisk) isn't sending the correct packets. I'm investigating this further
> before I re-enable the local ringing sounds. I'll let you know what I find
> out!
>
> Whit
>
>
> -----Original Message-----
> From: ope...@li...
> [mailto:ope...@li...] On Behalf Of Ilian
> Jeri C. Pinzon
> Sent: Wednesday, October 24, 2007 6:35 AM
> To: jb...@so...; ope...@li...
> Subject: Re: [OpenSIPStack] [ATLSIP] Ringing Sounds
>
> RFC 3960 *suggests* the following policy:
>
> =========================================================
> 1. Unless a 180 (Ringing) response is received, never generate
> local ringing.
>
> 2. If a 180 (Ringing) has been received but there are no incoming
> media packets, generate local ringing.
>
> 3. If a 180 (Ringing) has been received and there are incoming
> media packets, play them and do not generate local ringing.
>
> ...
>
> That is, any UA should play incoming media packets (*and stop local
> ringing tone
> generation if it was being performed*) in order to avoid media clipping,
> even if the 200 (OK) response has not arrived.
> =========================================================
>
> I think interrupting the local ring tone, if it is playing, in favor of
> the early media
> is acceptable already.
>
> - Ilian
>
> Joegen E. Baclor wrote:
>
>> This may happen if your gateway sends a 180 without SDP followed by 180
>> or a 183 with SDP. This can be corrected by stopping the false ring
>> in OnProgress()
>>
>> void SoftPhoneSIPEndPoint::OnProgress(
>> CallSession & session,
>> const SIPMessage & alerting
>> )
>> {
>> OpalOSSEndPoint::OnProgress( session, alerting );
>> PString info = session.GetTargetURI().AsString();
>> if( session.GetType() == CallSession::Client )
>> m_Manager.GetSoftPhoneInterface()->Event_OutgoingCallRinging( (const
>> char *)info );
>> }
>>
>> Just insert m_Manager.GetSoftPhoneInterface()->StopRingBackTone(); in
>> this method as a quick hack. I think a cleaner way of doing this is to
>> not honor early media at all and retain the false ring if the call has
>> already received a no-media Alerting packet prior to the 183. perhaps
>> we can set this in the stack level. I am open to suggestions.
>>
>> What's your view Ilian?
>>
>> Joegen
>>
>> Whit Thiele wrote:
>>
>>
>>> Hey guys,
>>>
>>> Where and when should the ringing sounds be generated? I use Asterisk so
>>> when a call is launched, asterisk generates the ringing sound. If I don't
>>> disable the PlayRingingSound methods, I get "double" rings.
>>>
>>>
>>> Should this be a configurable setting in the ATLSIP library?
>>>
>>> Whit
>>>
>>>
>>>
>>> -------------------------------------------------------------------------
>>> This SF.net email is sponsored by: Splunk Inc.
>>> Still grepping through log files to find problems? Stop.
>>> Now Search log events and configuration files using AJAX and a browser.
>>> Download your FREE copy of Splunk now >> http://get.splunk.com/
>>> _______________________________________________
>>> opensipstack-devel mailing list
>>> ope...@li...
>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>>
>>>
>>>
>>>
>>>
>> -------------------------------------------------------------------------
>> This SF.net email is sponsored by: Splunk Inc.
>> Still grepping through log files to find problems? Stop.
>> Now Search log events and configuration files using AJAX and a browser.
>> Download your FREE copy of Splunk now >> http://get.splunk.com/
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>>
>>
>>
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems? Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >> http://get.splunk.com/
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems? Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >> http://get.splunk.com/
> _______________________________________________
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> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>
|
|
From: Joegen E. B. <joe...@gm...> - 2007-10-26 02:25:20
|
Request Pending Error is normally sent back by a server if it received a new INVITE transaction while there is still an existing INVITE transaction pending which has not been responded to with a final response. Whit Thiele wrote: > Hey Guys, > > I've been having some trouble with dialing to Asterisk (v.1.4.9) with my > ATLSIP Softphone. I am able to dial most of the time with no trouble > although I am getting "491 Request Pending" messages from the > ATLSIP_OnOutgoingCallRejected event randomly say 30% of calls launched. > > I've read that asterisk has had some trouble dealing with 491 errors but I > thought I'd ask the forum: > > 1. Has anyone else has had any problems with this? > > 2. Anyone know the best way to avoid this from happening. > > > It's a simple setup with the Softphone and Asterisk on the same local > network. > > ATLSIP --> Asterisk --> PRI --> PSTN > > > Thanks, > > Whit > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Splunk Inc. > Still grepping through log files to find problems? Stop. > Now Search log events and configuration files using AJAX and a browser. > Download your FREE copy of Splunk now >> http://get.splunk.com/ > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
|
From: Whit T. <de...@wh...> - 2007-10-25 19:05:16
|
Illian,
Thanks for your information! In my particular instance, it was easier to
temporarily disable the local ringing sound because I think my pbx
(asterisk) isn't sending the correct packets. I'm investigating this further
before I re-enable the local ringing sounds. I'll let you know what I find
out!
Whit
-----Original Message-----
From: ope...@li...
[mailto:ope...@li...] On Behalf Of Ilian
Jeri C. Pinzon
Sent: Wednesday, October 24, 2007 6:35 AM
To: jb...@so...; ope...@li...
Subject: Re: [OpenSIPStack] [ATLSIP] Ringing Sounds
RFC 3960 *suggests* the following policy:
=========================================================
1. Unless a 180 (Ringing) response is received, never generate
local ringing.
2. If a 180 (Ringing) has been received but there are no incoming
media packets, generate local ringing.
3. If a 180 (Ringing) has been received and there are incoming
media packets, play them and do not generate local ringing.
...
That is, any UA should play incoming media packets (*and stop local
ringing tone
generation if it was being performed*) in order to avoid media clipping,
even if the 200 (OK) response has not arrived.
=========================================================
I think interrupting the local ring tone, if it is playing, in favor of
the early media
is acceptable already.
- Ilian
Joegen E. Baclor wrote:
> This may happen if your gateway sends a 180 without SDP followed by 180
> or a 183 with SDP. This can be corrected by stopping the false ring
> in OnProgress()
>
> void SoftPhoneSIPEndPoint::OnProgress(
> CallSession & session,
> const SIPMessage & alerting
> )
> {
> OpalOSSEndPoint::OnProgress( session, alerting );
> PString info = session.GetTargetURI().AsString();
> if( session.GetType() == CallSession::Client )
> m_Manager.GetSoftPhoneInterface()->Event_OutgoingCallRinging( (const
> char *)info );
> }
>
> Just insert m_Manager.GetSoftPhoneInterface()->StopRingBackTone(); in
> this method as a quick hack. I think a cleaner way of doing this is to
> not honor early media at all and retain the false ring if the call has
> already received a no-media Alerting packet prior to the 183. perhaps
> we can set this in the stack level. I am open to suggestions.
>
> What's your view Ilian?
>
> Joegen
>
> Whit Thiele wrote:
>
>> Hey guys,
>>
>> Where and when should the ringing sounds be generated? I use Asterisk so
>> when a call is launched, asterisk generates the ringing sound. If I don't
>> disable the PlayRingingSound methods, I get "double" rings.
>>
>>
>> Should this be a configurable setting in the ATLSIP library?
>>
>> Whit
>>
>>
>>
>> -------------------------------------------------------------------------
>> This SF.net email is sponsored by: Splunk Inc.
>> Still grepping through log files to find problems? Stop.
>> Now Search log events and configuration files using AJAX and a browser.
>> Download your FREE copy of Splunk now >> http://get.splunk.com/
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>>
>>
>>
>
>
> -------------------------------------------------------------------------
> This SF.net email is sponsored by: Splunk Inc.
> Still grepping through log files to find problems? Stop.
> Now Search log events and configuration files using AJAX and a browser.
> Download your FREE copy of Splunk now >> http://get.splunk.com/
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
>
>
-------------------------------------------------------------------------
This SF.net email is sponsored by: Splunk Inc.
Still grepping through log files to find problems? Stop.
Now Search log events and configuration files using AJAX and a browser.
Download your FREE copy of Splunk now >> http://get.splunk.com/
_______________________________________________
opensipstack-devel mailing list
ope...@li...
https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
|
|
From: Whit T. <de...@wh...> - 2007-10-25 19:01:09
|
Hey Guys, I've been having some trouble with dialing to Asterisk (v.1.4.9) with my ATLSIP Softphone. I am able to dial most of the time with no trouble although I am getting "491 Request Pending" messages from the ATLSIP_OnOutgoingCallRejected event randomly say 30% of calls launched. I've read that asterisk has had some trouble dealing with 491 errors but I thought I'd ask the forum: 1. Has anyone else has had any problems with this? 2. Anyone know the best way to avoid this from happening. It's a simple setup with the Softphone and Asterisk on the same local network. ATLSIP --> Asterisk --> PRI --> PSTN Thanks, Whit |
|
From: Ilian J. C. P. <ip...@so...> - 2007-10-24 11:35:11
|
RFC 3960 *suggests* the following policy:
=========================================================
1. Unless a 180 (Ringing) response is received, never generate
local ringing.
2. If a 180 (Ringing) has been received but there are no incoming
media packets, generate local ringing.
3. If a 180 (Ringing) has been received and there are incoming
media packets, play them and do not generate local ringing.
...
That is, any UA should play incoming media packets (*and stop local
ringing tone
generation if it was being performed*) in order to avoid media clipping,
even if the 200 (OK) response has not arrived.
=========================================================
I think interrupting the local ring tone, if it is playing, in favor of
the early media
is acceptable already.
- Ilian
Joegen E. Baclor wrote:
> This may happen if your gateway sends a 180 without SDP followed by 180
> or a 183 with SDP. This can be corrected by stopping the false ring
> in OnProgress()
>
> void SoftPhoneSIPEndPoint::OnProgress(
> CallSession & session,
> const SIPMessage & alerting
> )
> {
> OpalOSSEndPoint::OnProgress( session, alerting );
> PString info = session.GetTargetURI().AsString();
> if( session.GetType() == CallSession::Client )
> m_Manager.GetSoftPhoneInterface()->Event_OutgoingCallRinging( (const
> char *)info );
> }
>
> Just insert m_Manager.GetSoftPhoneInterface()->StopRingBackTone(); in
> this method as a quick hack. I think a cleaner way of doing this is to
> not honor early media at all and retain the false ring if the call has
> already received a no-media Alerting packet prior to the 183. perhaps
> we can set this in the stack level. I am open to suggestions.
>
> What's your view Ilian?
>
> Joegen
>
> Whit Thiele wrote:
>
>> Hey guys,
>>
>> Where and when should the ringing sounds be generated? I use Asterisk so
>> when a call is launched, asterisk generates the ringing sound. If I don't
>> disable the PlayRingingSound methods, I get "double" rings.
>>
>>
>> Should this be a configurable setting in the ATLSIP library?
>>
>> Whit
>>
>>
>>
>> -------------------------------------------------------------------------
>> This SF.net email is sponsored by: Splunk Inc.
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From: Joegen E. B. <joe...@gm...> - 2007-10-24 10:22:52
|
This may happen if your gateway sends a 180 without SDP followed by 180
or a 183 with SDP. This can be corrected by stopping the false ring
in OnProgress()
void SoftPhoneSIPEndPoint::OnProgress(
CallSession & session,
const SIPMessage & alerting
)
{
OpalOSSEndPoint::OnProgress( session, alerting );
PString info = session.GetTargetURI().AsString();
if( session.GetType() == CallSession::Client )
m_Manager.GetSoftPhoneInterface()->Event_OutgoingCallRinging( (const
char *)info );
}
Just insert m_Manager.GetSoftPhoneInterface()->StopRingBackTone(); in
this method as a quick hack. I think a cleaner way of doing this is to
not honor early media at all and retain the false ring if the call has
already received a no-media Alerting packet prior to the 183. perhaps
we can set this in the stack level. I am open to suggestions.
What's your view Ilian?
Joegen
Whit Thiele wrote:
> Hey guys,
>
> Where and when should the ringing sounds be generated? I use Asterisk so
> when a call is launched, asterisk generates the ringing sound. If I don't
> disable the PlayRingingSound methods, I get "double" rings.
>
>
> Should this be a configurable setting in the ATLSIP library?
>
> Whit
>
>
>
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From: Joegen E. B. <joe...@gm...> - 2007-10-24 10:17:33
|
OpenSBC should be handle this naturally if you bind it to all interfaces. OpenSBC will bridge your static public network with your DHCP based network which i assume is private voice wrote: > Hi Joegen > > OpenSBC is essessially a sipNAT router. Can i have openSBC with two > ethernet cards running on a Linux server urunning a DHCP service. One card > pointing to the Internet and the other pointing to the DHCP address? Or do > i need to a router and a second Linux box providing DHCP. i.e. openSBD > <==> router <==> Linux DHCP. > > r > > > > > |
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