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From: Joegen E. B. <joe...@gm...> - 2006-12-16 03:05:26
|
This are just warnings and may be ignored. It only means the configure script didn't find dependency librararies for the particular class. For example, G.729 support would only be enabled if you have instlled the Voice Age G.729 library. Gustavo Curetti wrote: > > Hi, I got this warnings when compiling OSS 1.1.3 with Visual Visual > C++ 2005: > > XMPP.obj : warning LNK4221: no public symbols found; archive member > will be inaccessible > > vsdl.obj : warning LNK4221: no public symbols found; archive member > will be inaccessible > > shttpsvc.obj : warning LNK4221: no public symbols found; archive > member will be inaccessible > > pssl.obj : warning LNK4221: no public symbols found; archive member > will be inaccessible > > psasl.obj : warning LNK4221: no public symbols found; archive member > will be inaccessible > > pldap.obj : warning LNK4221: no public symbols found; archive member > will be inaccessible > > pils.obj : warning LNK4221: no public symbols found; archive member > will be inaccessible > > vpblid.obj : warning LNK4221: no public symbols found; archive member > will be inaccessible > > h263codec.obj : warning LNK4221: no public symbols found; archive > member will be inaccessible > > g729codec.obj : warning LNK4221: no public symbols found; archive > member will be inaccessible > > g7231codec.obj : warning LNK4221: no public symbols found; archive > member will be inaccessible > > Any advice? > > Thanks, > > Gustavo > > > ------------------------------------------------------------------------ > Sé uno de los primeros a testar el Windows Live Messenger beta. ¡Haz > click aquí! <http://g.msn.com/8HMBESAR/2749??PS=47575> > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
From: Gustavo C. <cur...@ho...> - 2006-12-15 18:56:08
|
<html><div style='background-color:'><DIV class=RTE> <DIV class=RTE> <P>Hi, I got the followings errors when compiling opensbc 1.1.1 with Visual Visual C++ 2005:</P><FONT size=1><FONT size=1> <P>c1xx : fatal error C1083: Cannot open source file: '.\RTBEURI.cxx': No such file or directory</P> <P>c1xx : fatal error C1083: Cannot open source file: '.\RTBESession.cxx': No such file or directory</P> <P>c1xx : fatal error C1083: Cannot open source file: '.\RTBERoute.cxx': No such file or directory</P> <P>c1xx : fatal error C1083: Cannot open source file: '.\RTBEPingHandler.cxx': No such file or directory</P> <P>c1xx : fatal error C1083: Cannot open source file: '.\RTBEPacket.cxx': No such file or directory</P> <P></FONT></FONT><FONT size=2>Any advice? </FONT></P> <P><FONT color=#000000>Thanks,</FONT></P> <P><FONT color=#000000> Gustavo</P></DIV> <P></FONT><BR><BR></P> <DIV></DIV><FONT color=#000000></DIV> <DIV></DIV></FONT></div><br clear=all><hr>Búsquedas mejores con MSN Search. <a href="http://g.msn.com/8HMBESAR/2740??PS=47575" target="_top">Pruébalo. </a> </html> |
From: Gustavo C. <cur...@ho...> - 2006-12-15 18:48:46
|
<html><div style='background-color:'><DIV class=RTE> <P>Hi, I got this warnings when compiling OSS 1.1.3 with Visual Visual C++ 2005:</P><FONT size=1> <P>XMPP.obj : warning LNK4221: no public symbols found; archive member will be inaccessible</P> <P>vsdl.obj : warning LNK4221: no public symbols found; archive member will be inaccessible</P> <P>shttpsvc.obj : warning LNK4221: no public symbols found; archive member will be inaccessible</P> <P>pssl.obj : warning LNK4221: no public symbols found; archive member will be inaccessible</P> <P>psasl.obj : warning LNK4221: no public symbols found; archive member will be inaccessible</P> <P>pldap.obj : warning LNK4221: no public symbols found; archive member will be inaccessible</P> <P>pils.obj : warning LNK4221: no public symbols found; archive member will be inaccessible</P> <P>vpblid.obj : warning LNK4221: no public symbols found; archive member will be inaccessible</P> <P>h263codec.obj : warning LNK4221: no public symbols found; archive member will be inaccessible</P> <P>g729codec.obj : warning LNK4221: no public symbols found; archive member will be inaccessible</P> <P>g7231codec.obj : warning LNK4221: no public symbols found; archive member will be inaccessible</P> <P></FONT><FONT size=2>Any advice? </FONT></P> <P><FONT color=#000000>Thanks,</FONT></P> <P><FONT color=#000000> Gustavo</P></DIV></FONT></div><br clear=all><hr>Sé uno de los primeros a testar el Windows Live Messenger beta. <a href="http://g.msn.com/8HMBESAR/2749??PS=47575" target="_top">¡Haz click aquí! </a> </html> |
From: Becker, C. <cb...@ma...> - 2006-12-15 00:11:26
|
Tested, it works. -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Joegen E. Baclor Sent: Thursday, December 14, 2006 3:32 PM To: ope...@li... Subject: Re: [OpenSIPStack] opensipstack-devel Digest, Vol 7, Issue 9 Hi Jonathon/Cody, Yes, OpenSBC can do failover ;-) Try this configuration. In the B2BUA Routes section of the admin page, put [sip:*] sip:asterisk01*, sip:asterisk02* Where asterisk01 and asterisk02 are your primary and secondary servers. OpenSBC should try to failover to the secondary server if the call to=20 the primary server fails. Let me know if you get it to work. Joegen Becker, Cody wrote: > I don't think the phone should successfully register if there is nothing > to register to, the SBC can't register the phones, just proxy the > REGISTER request and OK messages. > > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of > Jonathon Warren > Sent: Wednesday, December 13, 2006 2:15 PM > To: ope...@li... > Subject: Re: [OpenSIPStack] opensipstack-devel Digest, Vol 7, Issue 9 > > Number 1 is what we are trying to do the only problem we see is if the > primary asterisk server is down does the primary SBC have the ability > to redirect traffic to the secondary SBC. If it does not, even if a sip > phone supports two proxies if the primary BC is running and the primary > asterisk is not, the phones would still register with the primary SBC. > Well, atleast this is what i am trying to get to the bottom of. Thanks > > > Jonathon Warren > President Business First Solutions Inc. > General Manager CHL Marketing Inc. > 4076947376 > > > > > > =20 >> -------- Original Message -------- >> Subject: opensipstack-devel Digest, Vol 7, Issue 9 >> From: ope...@li... >> Date: Wed, December 13, 2006 3:16 pm >> To: ope...@li... >> >> Send opensipstack-devel mailing list submissions to >> ope...@li... >> >> To subscribe or unsubscribe via the World Wide Web, visit >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> or, via email, send a message with subject or body 'help' to >> ope...@li... >> >> You can reach the person managing the list at >> ope...@li... >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of opensipstack-devel digest..." >> >> >> Today's Topics: >> >> 1. Asterisk and OpenSBC (Jonathon Warren) >> 2. Re: Asterisk and OpenSBC (Becker, Cody) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 13 Dec 2006 09:35:14 -0700 >> From: Jonathon Warren <j.w...@bf...> >> Subject: [OpenSIPStack] Asterisk and OpenSBC >> To: ope...@li... >> Message-ID: >> >> =20 > <200...@em... > cureserver.net> > =20 >> Content-Type: TEXT/plain; CHARSET=3DUS-ASCII >> >> I am in the decision process for choosing a SBC/NAT Transversal >> =20 > solution > =20 >> to my existing asterisk based network. I have a primary and secondary >> asterisk server and would like to put in place a primary and secondary >> sbc for redundancy. Is it possible to set up opensbc to route traffic >> to a primary server and if the primary server is down/unavailable, >> route traffic to a secondary server. Any information in regards to a >> Asterisk / OpenSBC setup would be very helpful. Thanks in advance. >> >> >> Jonathon Warren >> President Business First Solutions Inc. >> General Manager CHL Marketing Inc. >> 4076947376 >> >> >> >> >> >> >> >> >> ------------------------------ >> >> Message: 2 >> Date: Wed, 13 Dec 2006 08:51:13 -0800 >> From: "Becker, Cody" <cb...@ma...> >> Subject: Re: [OpenSIPStack] Asterisk and OpenSBC >> To: <ope...@li...> >> Message-ID: >> <919...@ma...> >> Content-Type: text/plain; charset=3D"us-ascii" >> >> We are in the process of testing this as well, we have come up with 3 >> solutions so far. >> >> #1 - Primary / Backup Proxy Server on the phones (We use Cisco 7912 >> phones which support keying in a primary / backup proxy). We are >> testing 2 OpenSBC boxes, one pointing at the primary and one at the >> backup. >> >> #2 - Ingate SIParator, internally supports primary/backup. >> >> #3 - Aliases on the ASTERISK boxes with DNS Priority records. >> >> -----Original Message----- >> From: ope...@li... >> [mailto:ope...@li...] On Behalf Of >> Jonathon Warren >> Sent: Wednesday, December 13, 2006 8:35 AM >> To: ope...@li... >> Subject: [OpenSIPStack] Asterisk and OpenSBC >> >> I am in the decision process for choosing a SBC/NAT Transversal >> =20 > solution > =20 >> to my existing asterisk based network. I have a primary and secondary >> asterisk server and would like to put in place a primary and secondary >> sbc for redundancy. Is it possible to set up opensbc to route traffic >> to a primary server and if the primary server is down/unavailable, >> route traffic to a secondary server. Any information in regards to a >> Asterisk / OpenSBC setup would be very helpful. Thanks in advance. >> >> >> Jonathon Warren >> President Business First Solutions Inc. >> General Manager CHL Marketing Inc. >> 4076947376 >> >> >> >> >> >> >> >> =20 > ------------------------------------------------------------------------ > =20 >> - >> Take Surveys. Earn Cash. Influence the Future of IT >> Join SourceForge.net's Techsay panel and you'll get the chance to >> =20 > share > =20 >> your >> opinions on IT & business topics through brief surveys - and earn cash >> >> =20 > http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE > =20 >> V >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> ------------------------------ >> >> >> =20 > ------------------------------------------------------------------------ > - > =20 >> Take Surveys. Earn Cash. Influence the Future of IT >> Join SourceForge.net's Techsay panel and you'll get the chance to >> =20 > share your > =20 >> opinions on IT & business topics through brief surveys - and earn cash >> >> =20 > http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE > V > =20 >> ------------------------------ >> >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> End of opensipstack-devel Digest, Vol 7, Issue 9 >> ************************************************ >> =20 > > > ------------------------------------------------------------------------ > - > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share > your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE > V > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > ------------------------------------------------------------------------ - > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE V > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > =20 ------------------------------------------------------------------------ - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE V _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
From: Joegen E. B. <joe...@gm...> - 2006-12-14 23:36:49
|
Becker, Cody wrote: > I don't think the phone should successfully register if there is nothing > to register to, the SBC can't register the phones, just proxy the > REGISTER request and OK messages. > > OpenSBC can accept registration if the To: URI points to a domain that resolves to the IP address of OpenSBC. In this case, OpenSBC will not forward registration but process/authenticate it using the Domain Accounts configuration page entries. OpenSBC only do upper registration if the To: URI resolves to an external route. Joegen |
From: Joegen E. B. <joe...@gm...> - 2006-12-14 23:32:10
|
Hi Jonathon/Cody, Yes, OpenSBC can do failover ;-) Try this configuration. In the B2BUA Routes section of the admin page, put [sip:*] sip:asterisk01*, sip:asterisk02* Where asterisk01 and asterisk02 are your primary and secondary servers. OpenSBC should try to failover to the secondary server if the call to the primary server fails. Let me know if you get it to work. Joegen Becker, Cody wrote: > I don't think the phone should successfully register if there is nothing > to register to, the SBC can't register the phones, just proxy the > REGISTER request and OK messages. > > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of > Jonathon Warren > Sent: Wednesday, December 13, 2006 2:15 PM > To: ope...@li... > Subject: Re: [OpenSIPStack] opensipstack-devel Digest, Vol 7, Issue 9 > > Number 1 is what we are trying to do the only problem we see is if the > primary asterisk server is down does the primary SBC have the ability > to redirect traffic to the secondary SBC. If it does not, even if a sip > phone supports two proxies if the primary BC is running and the primary > asterisk is not, the phones would still register with the primary SBC. > Well, atleast this is what i am trying to get to the bottom of. Thanks > > > Jonathon Warren > President Business First Solutions Inc. > General Manager CHL Marketing Inc. > 4076947376 > > > > > > >> -------- Original Message -------- >> Subject: opensipstack-devel Digest, Vol 7, Issue 9 >> From: ope...@li... >> Date: Wed, December 13, 2006 3:16 pm >> To: ope...@li... >> >> Send opensipstack-devel mailing list submissions to >> ope...@li... >> >> To subscribe or unsubscribe via the World Wide Web, visit >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> or, via email, send a message with subject or body 'help' to >> ope...@li... >> >> You can reach the person managing the list at >> ope...@li... >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of opensipstack-devel digest..." >> >> >> Today's Topics: >> >> 1. Asterisk and OpenSBC (Jonathon Warren) >> 2. Re: Asterisk and OpenSBC (Becker, Cody) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 13 Dec 2006 09:35:14 -0700 >> From: Jonathon Warren <j.w...@bf...> >> Subject: [OpenSIPStack] Asterisk and OpenSBC >> To: ope...@li... >> Message-ID: >> >> > <200...@em... > cureserver.net> > >> Content-Type: TEXT/plain; CHARSET=US-ASCII >> >> I am in the decision process for choosing a SBC/NAT Transversal >> > solution > >> to my existing asterisk based network. I have a primary and secondary >> asterisk server and would like to put in place a primary and secondary >> sbc for redundancy. Is it possible to set up opensbc to route traffic >> to a primary server and if the primary server is down/unavailable, >> route traffic to a secondary server. Any information in regards to a >> Asterisk / OpenSBC setup would be very helpful. Thanks in advance. >> >> >> Jonathon Warren >> President Business First Solutions Inc. >> General Manager CHL Marketing Inc. >> 4076947376 >> >> >> >> >> >> >> >> >> ------------------------------ >> >> Message: 2 >> Date: Wed, 13 Dec 2006 08:51:13 -0800 >> From: "Becker, Cody" <cb...@ma...> >> Subject: Re: [OpenSIPStack] Asterisk and OpenSBC >> To: <ope...@li...> >> Message-ID: >> <919...@ma...> >> Content-Type: text/plain; charset="us-ascii" >> >> We are in the process of testing this as well, we have come up with 3 >> solutions so far. >> >> #1 - Primary / Backup Proxy Server on the phones (We use Cisco 7912 >> phones which support keying in a primary / backup proxy). We are >> testing 2 OpenSBC boxes, one pointing at the primary and one at the >> backup. >> >> #2 - Ingate SIParator, internally supports primary/backup. >> >> #3 - Aliases on the ASTERISK boxes with DNS Priority records. >> >> -----Original Message----- >> From: ope...@li... >> [mailto:ope...@li...] On Behalf Of >> Jonathon Warren >> Sent: Wednesday, December 13, 2006 8:35 AM >> To: ope...@li... >> Subject: [OpenSIPStack] Asterisk and OpenSBC >> >> I am in the decision process for choosing a SBC/NAT Transversal >> > solution > >> to my existing asterisk based network. I have a primary and secondary >> asterisk server and would like to put in place a primary and secondary >> sbc for redundancy. Is it possible to set up opensbc to route traffic >> to a primary server and if the primary server is down/unavailable, >> route traffic to a secondary server. Any information in regards to a >> Asterisk / OpenSBC setup would be very helpful. Thanks in advance. >> >> >> Jonathon Warren >> President Business First Solutions Inc. >> General Manager CHL Marketing Inc. >> 4076947376 >> >> >> >> >> >> >> >> > ------------------------------------------------------------------------ > >> - >> Take Surveys. Earn Cash. Influence the Future of IT >> Join SourceForge.net's Techsay panel and you'll get the chance to >> > share > >> your >> opinions on IT & business topics through brief surveys - and earn cash >> >> > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDE > >> V >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> ------------------------------ >> >> >> > ------------------------------------------------------------------------ > - > >> Take Surveys. Earn Cash. Influence the Future of IT >> Join SourceForge.net's Techsay panel and you'll get the chance to >> > share your > >> opinions on IT & business topics through brief surveys - and earn cash >> >> > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDE > V > >> ------------------------------ >> >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> End of opensipstack-devel Digest, Vol 7, Issue 9 >> ************************************************ >> > > > ------------------------------------------------------------------------ > - > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share > your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDE > V > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > |
From: Becker, C. <cb...@ma...> - 2006-12-14 16:04:50
|
Here's something I found that may be helpful to someone else out there that I wanted to post. We were having the classic 'early media' problem, where the client phone would not receive audio until it first sent audio if all ports were not forwarded to the phone. Our solution on the Cisco 79xx was simply to disable silence suppression in the firmware on the phone. You have to build a new gkdefault.cfg and push it out to the phone with TFTP, but after that was done life has been golden, we now are not required to do any port forwarding on the client router as the silence packets always make it there before someone starts talking. It wastes a little bandwidth but is a good solution for us. Regards, Cody -----Original Message----- From: Becker, Cody=20 Sent: Wednesday, December 13, 2006 2:24 PM To: ope...@li... Subject: RE: [OpenSIPStack] opensipstack-devel Digest, Vol 7, Issue 9 I don't think the phone should successfully register if there is nothing to register to, the SBC can't register the phones, just proxy the REGISTER request and OK messages. -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Jonathon Warren Sent: Wednesday, December 13, 2006 2:15 PM To: ope...@li... Subject: Re: [OpenSIPStack] opensipstack-devel Digest, Vol 7, Issue 9 Number 1 is what we are trying to do the only problem we see is if the primary asterisk server is down does the primary SBC have the ability to redirect traffic to the secondary SBC. If it does not, even if a sip phone supports two proxies if the primary BC is running and the primary asterisk is not, the phones would still register with the primary SBC. Well, atleast this is what i am trying to get to the bottom of. Thanks Jonathon Warren President Business First Solutions Inc. General Manager CHL Marketing Inc. 4076947376 > -------- Original Message -------- > Subject: opensipstack-devel Digest, Vol 7, Issue 9 > From: ope...@li... > Date: Wed, December 13, 2006 3:16 pm > To: ope...@li... > > Send opensipstack-devel mailing list submissions to > ope...@li... > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > or, via email, send a message with subject or body 'help' to > ope...@li... > > You can reach the person managing the list at > ope...@li... > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of opensipstack-devel digest..." > > > Today's Topics: > > 1. Asterisk and OpenSBC (Jonathon Warren) > 2. Re: Asterisk and OpenSBC (Becker, Cody) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 13 Dec 2006 09:35:14 -0700 > From: Jonathon Warren <j.w...@bf...> > Subject: [OpenSIPStack] Asterisk and OpenSBC > To: ope...@li... > Message-ID: > <200...@em... cureserver.net> > > Content-Type: TEXT/plain; CHARSET=3DUS-ASCII > > I am in the decision process for choosing a SBC/NAT Transversal solution > to my existing asterisk based network. I have a primary and secondary > asterisk server and would like to put in place a primary and secondary > sbc for redundancy. Is it possible to set up opensbc to route traffic > to a primary server and if the primary server is down/unavailable, > route traffic to a secondary server. Any information in regards to a > Asterisk / OpenSBC setup would be very helpful. Thanks in advance. > > > Jonathon Warren > President Business First Solutions Inc. > General Manager CHL Marketing Inc. > 4076947376 > > > > > > > > > ------------------------------ > > Message: 2 > Date: Wed, 13 Dec 2006 08:51:13 -0800 > From: "Becker, Cody" <cb...@ma...> > Subject: Re: [OpenSIPStack] Asterisk and OpenSBC > To: <ope...@li...> > Message-ID: > <919...@ma...> > Content-Type: text/plain; charset=3D"us-ascii" > > We are in the process of testing this as well, we have come up with 3 > solutions so far. > > #1 - Primary / Backup Proxy Server on the phones (We use Cisco 7912 > phones which support keying in a primary / backup proxy). We are > testing 2 OpenSBC boxes, one pointing at the primary and one at the > backup. > > #2 - Ingate SIParator, internally supports primary/backup. > > #3 - Aliases on the ASTERISK boxes with DNS Priority records. > > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of > Jonathon Warren > Sent: Wednesday, December 13, 2006 8:35 AM > To: ope...@li... > Subject: [OpenSIPStack] Asterisk and OpenSBC > > I am in the decision process for choosing a SBC/NAT Transversal solution > to my existing asterisk based network. I have a primary and secondary > asterisk server and would like to put in place a primary and secondary > sbc for redundancy. Is it possible to set up opensbc to route traffic > to a primary server and if the primary server is down/unavailable, > route traffic to a secondary server. Any information in regards to a > Asterisk / OpenSBC setup would be very helpful. Thanks in advance. > > > Jonathon Warren > President Business First Solutions Inc. > General Manager CHL Marketing Inc. > 4076947376 > > > > > > > ------------------------------------------------------------------------ > - > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share > your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE > V > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > ------------------------------ > > ------------------------------------------------------------------------ - > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE V > > ------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > End of opensipstack-devel Digest, Vol 7, Issue 9 > ************************************************ ------------------------------------------------------------------------ - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE V _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
From: Becker, C. <cb...@ma...> - 2006-12-13 22:23:39
|
I don't think the phone should successfully register if there is nothing to register to, the SBC can't register the phones, just proxy the REGISTER request and OK messages. -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Jonathon Warren Sent: Wednesday, December 13, 2006 2:15 PM To: ope...@li... Subject: Re: [OpenSIPStack] opensipstack-devel Digest, Vol 7, Issue 9 Number 1 is what we are trying to do the only problem we see is if the primary asterisk server is down does the primary SBC have the ability to redirect traffic to the secondary SBC. If it does not, even if a sip phone supports two proxies if the primary BC is running and the primary asterisk is not, the phones would still register with the primary SBC. Well, atleast this is what i am trying to get to the bottom of. Thanks Jonathon Warren President Business First Solutions Inc. General Manager CHL Marketing Inc. 4076947376 > -------- Original Message -------- > Subject: opensipstack-devel Digest, Vol 7, Issue 9 > From: ope...@li... > Date: Wed, December 13, 2006 3:16 pm > To: ope...@li... > > Send opensipstack-devel mailing list submissions to > ope...@li... > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > or, via email, send a message with subject or body 'help' to > ope...@li... > > You can reach the person managing the list at > ope...@li... > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of opensipstack-devel digest..." > > > Today's Topics: > > 1. Asterisk and OpenSBC (Jonathon Warren) > 2. Re: Asterisk and OpenSBC (Becker, Cody) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 13 Dec 2006 09:35:14 -0700 > From: Jonathon Warren <j.w...@bf...> > Subject: [OpenSIPStack] Asterisk and OpenSBC > To: ope...@li... > Message-ID: > <200...@em... cureserver.net> > > Content-Type: TEXT/plain; CHARSET=3DUS-ASCII > > I am in the decision process for choosing a SBC/NAT Transversal solution > to my existing asterisk based network. I have a primary and secondary > asterisk server and would like to put in place a primary and secondary > sbc for redundancy. Is it possible to set up opensbc to route traffic > to a primary server and if the primary server is down/unavailable, > route traffic to a secondary server. Any information in regards to a > Asterisk / OpenSBC setup would be very helpful. Thanks in advance. > > > Jonathon Warren > President Business First Solutions Inc. > General Manager CHL Marketing Inc. > 4076947376 > > > > > > > > > ------------------------------ > > Message: 2 > Date: Wed, 13 Dec 2006 08:51:13 -0800 > From: "Becker, Cody" <cb...@ma...> > Subject: Re: [OpenSIPStack] Asterisk and OpenSBC > To: <ope...@li...> > Message-ID: > <919...@ma...> > Content-Type: text/plain; charset=3D"us-ascii" > > We are in the process of testing this as well, we have come up with 3 > solutions so far. > > #1 - Primary / Backup Proxy Server on the phones (We use Cisco 7912 > phones which support keying in a primary / backup proxy). We are > testing 2 OpenSBC boxes, one pointing at the primary and one at the > backup. > > #2 - Ingate SIParator, internally supports primary/backup. > > #3 - Aliases on the ASTERISK boxes with DNS Priority records. > > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of > Jonathon Warren > Sent: Wednesday, December 13, 2006 8:35 AM > To: ope...@li... > Subject: [OpenSIPStack] Asterisk and OpenSBC > > I am in the decision process for choosing a SBC/NAT Transversal solution > to my existing asterisk based network. I have a primary and secondary > asterisk server and would like to put in place a primary and secondary > sbc for redundancy. Is it possible to set up opensbc to route traffic > to a primary server and if the primary server is down/unavailable, > route traffic to a secondary server. Any information in regards to a > Asterisk / OpenSBC setup would be very helpful. Thanks in advance. > > > Jonathon Warren > President Business First Solutions Inc. > General Manager CHL Marketing Inc. > 4076947376 > > > > > > > ------------------------------------------------------------------------ > - > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share > your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE > V > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > ------------------------------ > > ------------------------------------------------------------------------ - > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE V > > ------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > End of opensipstack-devel Digest, Vol 7, Issue 9 > ************************************************ ------------------------------------------------------------------------ - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE V _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
From: Jonathon W. <j.w...@bf...> - 2006-12-13 22:15:22
|
Number 1 is what we are trying to do the only problem we see is if the primary asterisk server is down does the primary SBC have the ability to redirect traffic to the secondary SBC. If it does not, even if a sip phone supports two proxies if the primary BC is running and the primary asterisk is not, the phones would still register with the primary SBC. Well, atleast this is what i am trying to get to the bottom of. Thanks Jonathon Warren President Business First Solutions Inc. General Manager CHL Marketing Inc. 4076947376 > -------- Original Message -------- > Subject: opensipstack-devel Digest, Vol 7, Issue 9 > From: ope...@li... > Date: Wed, December 13, 2006 3:16 pm > To: ope...@li... > > Send opensipstack-devel mailing list submissions to > ope...@li... > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > or, via email, send a message with subject or body 'help' to > ope...@li... > > You can reach the person managing the list at > ope...@li... > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of opensipstack-devel digest..." > > > Today's Topics: > > 1. Asterisk and OpenSBC (Jonathon Warren) > 2. Re: Asterisk and OpenSBC (Becker, Cody) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 13 Dec 2006 09:35:14 -0700 > From: Jonathon Warren <j.w...@bf...> > Subject: [OpenSIPStack] Asterisk and OpenSBC > To: ope...@li... > Message-ID: > <200...@em...> > > Content-Type: TEXT/plain; CHARSET=US-ASCII > > I am in the decision process for choosing a SBC/NAT Transversal solution > to my existing asterisk based network. I have a primary and secondary > asterisk server and would like to put in place a primary and secondary > sbc for redundancy. Is it possible to set up opensbc to route traffic > to a primary server and if the primary server is down/unavailable, > route traffic to a secondary server. Any information in regards to a > Asterisk / OpenSBC setup would be very helpful. Thanks in advance. > > > Jonathon Warren > President Business First Solutions Inc. > General Manager CHL Marketing Inc. > 4076947376 > > > > > > > > > ------------------------------ > > Message: 2 > Date: Wed, 13 Dec 2006 08:51:13 -0800 > From: "Becker, Cody" <cb...@ma...> > Subject: Re: [OpenSIPStack] Asterisk and OpenSBC > To: <ope...@li...> > Message-ID: > <919...@ma...> > Content-Type: text/plain; charset="us-ascii" > > We are in the process of testing this as well, we have come up with 3 > solutions so far. > > #1 - Primary / Backup Proxy Server on the phones (We use Cisco 7912 > phones which support keying in a primary / backup proxy). We are > testing 2 OpenSBC boxes, one pointing at the primary and one at the > backup. > > #2 - Ingate SIParator, internally supports primary/backup. > > #3 - Aliases on the ASTERISK boxes with DNS Priority records. > > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of > Jonathon Warren > Sent: Wednesday, December 13, 2006 8:35 AM > To: ope...@li... > Subject: [OpenSIPStack] Asterisk and OpenSBC > > I am in the decision process for choosing a SBC/NAT Transversal solution > to my existing asterisk based network. I have a primary and secondary > asterisk server and would like to put in place a primary and secondary > sbc for redundancy. Is it possible to set up opensbc to route traffic > to a primary server and if the primary server is down/unavailable, > route traffic to a secondary server. Any information in regards to a > Asterisk / OpenSBC setup would be very helpful. Thanks in advance. > > > Jonathon Warren > President Business First Solutions Inc. > General Manager CHL Marketing Inc. > 4076947376 > > > > > > > ------------------------------------------------------------------------ > - > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share > your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDE > V > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > ------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > > ------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > End of opensipstack-devel Digest, Vol 7, Issue 9 > ************************************************ |
From: Jonathon W. <j.w...@ch...> - 2006-12-13 22:07:46
|
Number 1 is what we are trying to do the only problem we see is if the primary asterisk server is down does the primary SBC have the ability to redirect traffic to the secondary SBC. If it does not, even if a sip phone supports two proxies if the primary BC is running and the primary asterisk is not, the phones would still register with the primary SBC. Well, atleast this is what i am trying to get to the bottom of. Thanks Jonathon Warren President Business First Solutions Inc. General Manager CHL Marketing Inc. 4076947376 > -------- Original Message -------- > Subject: opensipstack-devel Digest, Vol 7, Issue 9 > From: ope...@li... > Date: Wed, December 13, 2006 3:16 pm > To: ope...@li... > > Send opensipstack-devel mailing list submissions to > ope...@li... > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > or, via email, send a message with subject or body 'help' to > ope...@li... > > You can reach the person managing the list at > ope...@li... > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of opensipstack-devel digest..." > > > Today's Topics: > > 1. Asterisk and OpenSBC (Jonathon Warren) > 2. Re: Asterisk and OpenSBC (Becker, Cody) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 13 Dec 2006 09:35:14 -0700 > From: Jonathon Warren <j.w...@bf...> > Subject: [OpenSIPStack] Asterisk and OpenSBC > To: ope...@li... > Message-ID: > <200...@em...> > > Content-Type: TEXT/plain; CHARSET=US-ASCII > > I am in the decision process for choosing a SBC/NAT Transversal solution > to my existing asterisk based network. I have a primary and secondary > asterisk server and would like to put in place a primary and secondary > sbc for redundancy. Is it possible to set up opensbc to route traffic > to a primary server and if the primary server is down/unavailable, > route traffic to a secondary server. Any information in regards to a > Asterisk / OpenSBC setup would be very helpful. Thanks in advance. > > > Jonathon Warren > President Business First Solutions Inc. > General Manager CHL Marketing Inc. > 4076947376 > > > > > > > > > ------------------------------ > > Message: 2 > Date: Wed, 13 Dec 2006 08:51:13 -0800 > From: "Becker, Cody" <cb...@ma...> > Subject: Re: [OpenSIPStack] Asterisk and OpenSBC > To: <ope...@li...> > Message-ID: > <919...@ma...> > Content-Type: text/plain; charset="us-ascii" > > We are in the process of testing this as well, we have come up with 3 > solutions so far. > > #1 - Primary / Backup Proxy Server on the phones (We use Cisco 7912 > phones which support keying in a primary / backup proxy). We are > testing 2 OpenSBC boxes, one pointing at the primary and one at the > backup. > > #2 - Ingate SIParator, internally supports primary/backup. > > #3 - Aliases on the ASTERISK boxes with DNS Priority records. > > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of > Jonathon Warren > Sent: Wednesday, December 13, 2006 8:35 AM > To: ope...@li... > Subject: [OpenSIPStack] Asterisk and OpenSBC > > I am in the decision process for choosing a SBC/NAT Transversal solution > to my existing asterisk based network. I have a primary and secondary > asterisk server and would like to put in place a primary and secondary > sbc for redundancy. Is it possible to set up opensbc to route traffic > to a primary server and if the primary server is down/unavailable, > route traffic to a secondary server. Any information in regards to a > Asterisk / OpenSBC setup would be very helpful. Thanks in advance. > > > Jonathon Warren > President Business First Solutions Inc. > General Manager CHL Marketing Inc. > 4076947376 > > > > > > > ------------------------------------------------------------------------ > - > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share > your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDE > V > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > ------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > > ------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > End of opensipstack-devel Digest, Vol 7, Issue 9 > ************************************************ |
From: Becker, C. <cb...@ma...> - 2006-12-13 16:51:39
|
We are in the process of testing this as well, we have come up with 3 solutions so far. #1 - Primary / Backup Proxy Server on the phones (We use Cisco 7912 phones which support keying in a primary / backup proxy). We are testing 2 OpenSBC boxes, one pointing at the primary and one at the backup. #2 - Ingate SIParator, internally supports primary/backup. #3 - Aliases on the ASTERISK boxes with DNS Priority records. -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Jonathon Warren Sent: Wednesday, December 13, 2006 8:35 AM To: ope...@li... Subject: [OpenSIPStack] Asterisk and OpenSBC I am in the decision process for choosing a SBC/NAT Transversal solution to my existing asterisk based network. I have a primary and secondary asterisk server and would like to put in place a primary and secondary sbc for redundancy. Is it possible to set up opensbc to route traffic to a primary server and if the primary server is down/unavailable, route traffic to a secondary server. Any information in regards to a Asterisk / OpenSBC setup would be very helpful. Thanks in advance. Jonathon Warren President Business First Solutions Inc. General Manager CHL Marketing Inc. 4076947376 ------------------------------------------------------------------------ - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE V _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
From: Jonathon W. <j.w...@bf...> - 2006-12-13 16:35:23
|
I am in the decision process for choosing a SBC/NAT Transversal solution to my existing asterisk based network. I have a primary and secondary asterisk server and would like to put in place a primary and secondary sbc for redundancy. Is it possible to set up opensbc to route traffic to a primary server and if the primary server is down/unavailable, route traffic to a secondary server. Any information in regards to a Asterisk / OpenSBC setup would be very helpful. Thanks in advance. Jonathon Warren President Business First Solutions Inc. General Manager CHL Marketing Inc. 4076947376 |
From: Picher, M. <mp...@cm...> - 2006-12-07 00:33:54
|
Not saying I can't compile it... Just saying I don't have the tools handy (nor do I want to pollute my machine any worse than it is with them)... I'm one of those non-programmer / engineer types :-) Thanks, Mike -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Joegen E. Baclor Sent: Wednesday, December 06, 2006 6:26 PM To: ope...@li... Subject: Re: [OpenSIPStack] Documentation Hi Mike, I will be releasing new versions of the stack and opensbc within the=20 next couple of days. I'll make a windows binary available in the=20 download pages. Are you saying you can't compile OpenSBC in windows? =20 What sort of problems are you encountering. I would be happy to help=20 you fix them. Joegen Picher, Michael wrote: > Any chance of getting a .exe for Windows? No Visual Studio available... > > Thanks, > Mike > > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of > Joegen E. Baclor > Sent: Tuesday, December 05, 2006 8:37 PM > To: Becker, Cody; ope...@li... > Subject: Re: [OpenSIPStack] Documentation > > > > Hi Cody, > > Thanks for the docs! > > Joegen > > Becker, Cody wrote: > =20 >> Part 1 Done, the install process. >> >> http://www.opensipstack.org/wiki/doku.php?id=3Dinstallation >> >> >> >> =20 >> =20 > > > ------------------------------------------------------------------------ > - > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share > your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE > V > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > ------------------------------------------------------------------------ - > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE V > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > =20 ------------------------------------------------------------------------ - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE V _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
From: Joegen E. B. <joe...@gm...> - 2006-12-06 23:28:33
|
Hi Cody, Made adjustments in CVS that would allow "Upper Registrations" to be hijacked by requests going in through the backdoor. Please do a fresh check out. Becker, Cody wrote: > Anyone know if it's possible to hijack the registration using IP > routing... ie.. > > Upper Registration Routes > [sip:*@publicip:*] sip:10.10.1.20:5060 > > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of > Joegen E. Baclor > Sent: Tuesday, December 05, 2006 8:37 PM > To: Becker, Cody; ope...@li... > Subject: Re: [OpenSIPStack] Documentation > > > > Hi Cody, > > Thanks for the docs! > > Joegen > > Becker, Cody wrote: > >> Part 1 Done, the install process. >> >> http://www.opensipstack.org/wiki/doku.php?id=installation >> >> >> >> >> > > > ------------------------------------------------------------------------ > - > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share > your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDE > V > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > ------------------------------------------------------------------------ > - > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share > your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDE > V > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > |
From: Joegen E. B. <joe...@gm...> - 2006-12-06 23:26:13
|
Hi Mike, I will be releasing new versions of the stack and opensbc within the next couple of days. I'll make a windows binary available in the download pages. Are you saying you can't compile OpenSBC in windows? What sort of problems are you encountering. I would be happy to help you fix them. Joegen Picher, Michael wrote: > Any chance of getting a .exe for Windows? No Visual Studio available... > > Thanks, > Mike > > -----Original Message----- > From: ope...@li... > [mailto:ope...@li...] On Behalf Of > Joegen E. Baclor > Sent: Tuesday, December 05, 2006 8:37 PM > To: Becker, Cody; ope...@li... > Subject: Re: [OpenSIPStack] Documentation > > > > Hi Cody, > > Thanks for the docs! > > Joegen > > Becker, Cody wrote: > >> Part 1 Done, the install process. >> >> http://www.opensipstack.org/wiki/doku.php?id=installation >> >> >> >> >> > > > ------------------------------------------------------------------------ > - > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share > your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDE > V > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > |
From: Joegen E. B. <joe...@gm...> - 2006-12-06 23:23:13
|
You may download ATLSIP from CVS and look at the OSSPhone projects. OSSPhone is a simple implementation of a SIP SoftPhone. Fatima Leyno wrote: > Hi, > We have just downloaded the opensipstack. Do you have any simple open > source telephony program that uses the sip stack? We wish to create an > application based on the available sample program. Thanks in advance. > > Fatima & Atchie > > ------------------------------------------------------------------------ > Cheap Talk? Check out > <http://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com> > Yahoo! Messenger's low PC-to-Phone call rates. > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
From: Becker, C. <cb...@ma...> - 2006-12-06 16:51:13
|
Anyone know if it's possible to hijack the registration using IP routing... ie.. Upper Registration Routes [sip:*@publicip:*] sip:10.10.1.20:5060 -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Joegen E. Baclor Sent: Tuesday, December 05, 2006 8:37 PM To: Becker, Cody; ope...@li... Subject: Re: [OpenSIPStack] Documentation Hi Cody, Thanks for the docs! Joegen Becker, Cody wrote: > Part 1 Done, the install process. > > http://www.opensipstack.org/wiki/doku.php?id=3Dinstallation > > > > =20 ------------------------------------------------------------------------ - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE V _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel ------------------------------------------------------------------------ - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE V _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
From: Jithendra <jit...@ce...> - 2006-12-06 08:38:56
|
Hello Joegen, Thank you very much for pointing me to the right direction. That helped. Thanks, Jithu -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Joegen E. Baclor Sent: Wednesday, December 06, 2006 6:54 AM To: ope...@li... Subject: Re: [OpenSIPStack] ATLSIP - Event handler problem Jithu, Check out http://msdn2.microsoft.com/en-us/library/system.windows.forms.control.checkf orillegalcrossthreadcalls.aspx It's because its now illegal in .net 2.0 to change the value of a control outside of the thread that created it. Setting the CheckForIllegalCrossThreadCalls property of your label to false would solve this. however, I strongly advice that you mutex the method properly to avoid overlapped critical section access. Joegen Jithendra wrote: > > Hello all, > > Iam trying to build a softphone using ATLSIP , using Winforms > (VC++.NET 2005). > > For sake of explaining the problem quickly, I reduce my GUI to only a > 'Label' control with shows me the status of SIP connectivity. ("Login > in success" or "Login Failed"). > > Now to display this status, I handle the OnLoginSuccessful event of > the ATLSIP control and modify the text of my label as below > > 25: System::Void mainForm::ATLSIPctl_OnLoginSucessful(System::Object^ > sender, System::EventArgs^ e){ > > 26: statusLbl->Text = "Login Successful"; > > 27: } > > But this does not update the label. I gave an add watch on the > statusLbl->Text variable (at line 26) and I see an error like "error: > cannot obtain value" . But if I change this statusLbl->Text from other > methods in the same Form class, it works with out problem. > > Why is this text field not accessible in the event handler of ATLSIP > control. > > Can some one help me out with this problem please. Cos I have already > got the control to login and successfully place calls and now Iam > concentrating on the cosmetics part and got stuck here. > > Thanks, > > Jithu > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > ------------------------------------------------------------------------- Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel -- -- |
From: Becker, C. <cb...@ma...> - 2006-12-06 03:49:02
|
After I work through my configuration on the web side and understand it = more, as well as test out the domain method of authenticating i'll = update the configuration page as well. Cody ________________________________ From: Joegen E. Baclor [mailto:joe...@gm...] Sent: Tue 12/5/2006 5:36 PM To: Becker, Cody; ope...@li... Subject: Re: Documentation Hi Cody, Thanks for the docs! Joegen Becker, Cody wrote: > Part 1 Done, the install process. > > http://www.opensipstack.org/wiki/doku.php?id=3Dinstallation > > > > =20 |
From: Fatima L. <fat...@ya...> - 2006-12-06 03:23:50
|
Hi,=0AWe have just downloaded the opensipstack. Do you have any simple open= source telephony program that uses the sip stack? We wish to create an app= lication based on the available sample program. Thanks in advance.=0A=0AFat= ima & Atchie=0A=0A=0A=0A=0A =0A____________________________________________= ________________________________________=0ADo you Yahoo!?=0AEveryone is rav= ing about the all-new Yahoo! Mail beta.=0Ahttp://new.mail.yahoo.com |
From: Picher, M. <mp...@cm...> - 2006-12-06 01:40:23
|
Any chance of getting a .exe for Windows? No Visual Studio available... Thanks, Mike -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Joegen E. Baclor Sent: Tuesday, December 05, 2006 8:37 PM To: Becker, Cody; ope...@li... Subject: Re: [OpenSIPStack] Documentation Hi Cody, Thanks for the docs! Joegen Becker, Cody wrote: > Part 1 Done, the install process. > > http://www.opensipstack.org/wiki/doku.php?id=3Dinstallation > > > > =20 ------------------------------------------------------------------------ - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3D= DEVDE V _______________________________________________ opensipstack-devel mailing list ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
From: Joegen E. B. <joe...@gm...> - 2006-12-06 01:36:41
|
Hi Cody, Thanks for the docs! Joegen Becker, Cody wrote: > Part 1 Done, the install process. > > http://www.opensipstack.org/wiki/doku.php?id=installation > > > > |
From: Joegen E. B. <joe...@gm...> - 2006-12-06 01:23:47
|
Jithu, Check out http://msdn2.microsoft.com/en-us/library/system.windows.forms.control.checkforillegalcrossthreadcalls.aspx It's because its now illegal in .net 2.0 to change the value of a control outside of the thread that created it. Setting the CheckForIllegalCrossThreadCalls property of your label to false would solve this. however, I strongly advice that you mutex the method properly to avoid overlapped critical section access. Joegen Jithendra wrote: > > Hello all, > > Iam trying to build a softphone using ATLSIP , using Winforms > (VC++.NET 2005). > > For sake of explaining the problem quickly, I reduce my GUI to only a > ‘Label’ control with shows me the status of SIP connectivity. (“Login > in success” or “Login Failed”). > > Now to display this status, I handle the OnLoginSuccessful event of > the ATLSIP control and modify the text of my label as below > > 25: System::Void mainForm::ATLSIPctl_OnLoginSucessful(System::Object^ > sender, System::EventArgs^ e){ > > 26: statusLbl->Text = “Login Successful”; > > 27: } > > But this does not update the label. I gave an add watch on the > statusLbl->Text variable (at line 26) and I see an error like “error: > cannot obtain value” . But if I change this statusLbl->Text from other > methods in the same Form class, it works with out problem. > > Why is this text field not accessible in the event handler of ATLSIP > control. > > Can some one help me out with this problem please. Cos I have already > got the control to login and successfully place calls and now Iam > concentrating on the cosmetics part and got stuck here. > > Thanks, > > Jithu > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
From: Jithendra <jit...@ce...> - 2006-12-05 12:39:24
|
Hello all, Iam trying to build a softphone using ATLSIP , using Winforms (VC++.NET 2005). For sake of explaining the problem quickly, I reduce my GUI to only a 'Label' control with shows me the status of SIP connectivity. ("Login in success" or "Login Failed"). Now to display this status, I handle the OnLoginSuccessful event of the ATLSIP control and modify the text of my label as below 25: System::Void mainForm::ATLSIPctl_OnLoginSucessful(System::Object^ sender, System::EventArgs^ e){ 26: statusLbl->Text = "Login Successful"; 27: } But this does not update the label. I gave an add watch on the statusLbl->Text variable (at line 26) and I see an error like "error: cannot obtain value" . But if I change this statusLbl->Text from other methods in the same Form class, it works with out problem. Why is this text field not accessible in the event handler of ATLSIP control. Can some one help me out with this problem please. Cos I have already got the control to login and successfully place calls and now Iam concentrating on the cosmetics part and got stuck here. Thanks, Jithu -- |
From: Becker, C. <cb...@ma...> - 2006-12-05 00:38:27
|
Thanks again for your help off list Joegen The :5080 backdoor trick for B2BUA worked like a champ. If we implement this we have to provide full internal documentation from our technical writers, I'll be sure to send you that for your files, it might save you some typing when you get around to documentation. Cody -----Original Message----- From: Becker, Cody=20 Sent: Monday, December 04, 2006 11:00 AM To: Becker, Cody; 'ope...@li...' Subject: RE: [OpenSIPStack] OpenSBC - Configuration Now that I have opensbc up and running, I am trying to configure it to speak to my SipX server and keep getting a core dump, most likely because I don't fully understand the webUI. What I am trying to do is the following: Place a opensbc server in our network border, say as below: OpenSBC box Private IP: 10.10.1.17 Public IP: x.x.x.99 SipX Box: 10.10.1.16 Phone outside LAN: 192.168.250.173 I get the call to the PBX but then I get a core dump in opensbc, what should my configuration routes look like in opensbc? Cody |