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From: Ashish K. <ash...@gm...> - 2007-09-05 05:02:57
|
Hi Joegen, I have two questions for RTP communication in openSBC. 1) clients have to use the same RTP ports to send the RTP packets on which they recieve RTP packets. So if the client is recieveing RTP packets on port 8000, Cient will use 8000 only to send the RTP packets? 2) When natted client initiated the call, for example Client A which is behind a NAT, calls B through openSBC, the first RTP packets need to be recieved from Client A, at openSBC, in order to send the RTP packets from Client B to Client A. this is because of NAT device, the RTP packets from A comes from different port than what was said in the SDP offfer and therefore RTP packets from B need to be send to this new port. is my understanding correct. Please clarify.? Please reply. Thanks. -Ashish |
From: Joegen E. B. <joe...@gm...> - 2007-09-05 04:10:44
|
Hi Julian, Got the logs. The reason for this apparent misbehavior is that your UAC is not sending a branch parameter in the VIA. Branch is vital in identifying transactions. Can you catch me in MSN Messenger? I want to try a possible hack. Joegen Joegen E. Baclor wrote: > Can I get a sample of the INVITE? Better yet, send me the entire level > 5 log offline ( jo...@op... ) > > Julian F. Tasis, III wrote: > >> Good day, >> >> In CallSession.cxx and OnIncomingSIPMessage >> >> OString fsm = tid.GetStateMachine(); >> >> When a call came from a sip client using RTC the variable fsm is null and that call will not be processed. >> >> For a workaround I just modified the else if part when fsm is equals to IST. >> >> I don't know if my workaround is the right way of handling such transaction and is it a must that fsm variable can't be null? >> >> Note: >> >> This only happens when a sip client is using RTC. >> >> Regards, >> Julian >> ------------------------------------------------------------------------- >> 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: Joegen E. B. <joe...@gm...> - 2007-09-05 02:14:54
|
Can I get a sample of the INVITE? Better yet, send me the entire level 5 log offline ( jo...@op... ) Julian F. Tasis, III wrote: > Good day, > > In CallSession.cxx and OnIncomingSIPMessage > > OString fsm = tid.GetStateMachine(); > > When a call came from a sip client using RTC the variable fsm is null and that call will not be processed. > > For a workaround I just modified the else if part when fsm is equals to IST. > > I don't know if my workaround is the right way of handling such transaction and is it a must that fsm variable can't be null? > > Note: > > This only happens when a sip client is using RTC. > > Regards, > Julian > ------------------------------------------------------------------------- > 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: Julian F. T. I. <jf...@id...> - 2007-09-05 01:54:09
|
Good day,=20 In CallSession.cxx and OnIncomingSIPMessage OString fsm =3D tid.GetStateMachine(); When a call came from a sip client using RTC the variable fsm is null = and that call will not be processed. For a workaround I just modified the else if part when fsm is equals to = IST. =20 I don't know if my workaround is the right way of handling such = transaction and is it a must that fsm variable can't be null? Note:=20 This only happens when a sip client is using RTC. Regards, Julian |
From: tomach <to...@dg...> - 2007-08-30 12:44:58
|
Hello! How are things going? I hope You are close? I really need this recording.... Please try to estimate when you can have it ready... |
From: <jo...@op...> - 2007-08-28 13:17:33
|
Yes. SIP allows multiple registrations of the same AOR pointing to=20 multiple bindings. The normal behavior of SIP proxies is to fork the=20 INVITEs if there are multiple bindings registered for an AOR. OpenSBC=20 doesn't fork currently (at least not yet) so it just sends the INVITE to = the latest binding. Kolneath SOMETH wrote: > > So is it normal that a sip account can register more than one time at=20 > the same time ? > (I mean *alice@domain* on UA1 n *alice@domain* on UA2 can register at=20 > the same time) > > Kolneath SOMETH > > > > > *"jo...@op..." <joegen.baclor* > > Envoy=E9 par : ope...@li... > > 28/08/2007 15:01 > Veuillez r=E9pondre =E0 joegen; Veuillez r=E9pondre =E0 opensipstack-de= vel > Remis le : 28/08/2007 15:01 > > =20 > Pour : ope...@li... > cc : (ccc : Kolneath SOMETH/DRD/SAGEM) > Objet : Re: [OpenSIPStack] R=E9f. : > > > > > You may modify the following code in opensbc.cxx > > > RegisterSession::AcceptRegistrationResponse OpenSBC::OnAcceptRegistrati= on( > RegisterSession &, > const SIPMessage & reg > ) > { > if( m_AcceptAllRegistration ) > return RegisterSession::AcceptByOk; > > /****************HERE IT IS*******************************/ > if( m_Registrar->GetRegistrationDB().FindRegistration( reg ) ) > { > PTRACE( 1, "Registration exists!" ); > } > /*********************************************************/ > > return RegisterSession::AcceptByChallenge; > } > > Kolneath SOMETH wrote: > > Hi Everyone, > > > > I want Registrar to check for the existence of SIP account (=20 > wheither he > > is already registered or not yet ). > > Could you pls tell me which file and what line number ? > > Thanks for your help in advance. > > > > Kolneath SOMETH > > > > > > > > > > " Ce courriel et les documents qui y sont attaches peuvent contenir=20 > des informations confidentielles. Si vous n'etes pas le destinataire=20 > escompte, merci d'en informer l'expediteur immediatement et de=20 > detruire ce courriel ainsi que tous les documents attaches de votre=20 > systeme informatique. Toute divulgation, distribution ou copie du=20 > present courriel et des documents attaches sans autorisation prealable = > de son emetteur est interdite." > > > > " This e-mail and any attached documents may contain confidential or = > proprietary information. If you are not the intended recipient, please = > advise the sender immediately and delete this e-mail and all attached=20 > documents from your computer system. Any unauthorised disclosure,=20 > distribution or copying hereof is prohibited." > >=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 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.= > 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 > > > > > " Ce courriel et les documents qui y sont attaches peuvent contenir=20 > des informations confidentielles. Si vous n'etes pas le destinataire=20 > escompte, merci d'en informer l'expediteur immediatement et de=20 > detruire ce courriel ainsi que tous les documents attaches de votre=20 > systeme informatique. Toute divulgation, distribution ou copie du=20 > present courriel et des documents attaches sans autorisation prealable = > de son emetteur est interdite." > > " This e-mail and any attached documents may contain confidential or=20 > proprietary information. If you are not the intended recipient, please = > advise the sender immediately and delete this e-mail and all attached=20 > documents from your computer system. Any unauthorised disclosure,=20 > distribution or copying hereof is prohibited." |
From: <jo...@op...> - 2007-08-28 13:01:32
|
You may modify the following code in opensbc.cxx RegisterSession::AcceptRegistrationResponse OpenSBC::OnAcceptRegistration( RegisterSession &, const SIPMessage & reg ) { if( m_AcceptAllRegistration ) return RegisterSession::AcceptByOk; /****************HERE IT IS*******************************/ if( m_Registrar->GetRegistrationDB().FindRegistration( reg ) ) { PTRACE( 1, "Registration exists!" ); } /*********************************************************/ return RegisterSession::AcceptByChallenge; } Kolneath SOMETH wrote: > Hi Everyone, > > I want Registrar to check for the existence of SIP account ( wheither he > is already registered or not yet ). > Could you pls tell me which file and what line number ? > Thanks for your help in advance. > > Kolneath SOMETH > > > > > " Ce courriel et les documents qui y sont attaches peuvent contenir des informations confidentielles. Si vous n'etes pas le destinataire escompte, merci d'en informer l'expediteur immediatement et de detruire ce courriel ainsi que tous les documents attaches de votre systeme informatique. Toute divulgation, distribution ou copie du present courriel et des documents attaches sans autorisation prealable de son emetteur est interdite." > > " This e-mail and any attached documents may contain confidential or proprietary information. If you are not the intended recipient, please advise the sender immediately and delete this e-mail and all attached documents from your computer system. Any unauthorised disclosure, distribution or copying hereof is prohibited." > ------------------------------------------------------------------------- > 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: Kolneath S. <kol...@sa...> - 2007-08-28 12:44:21
|
Hi Everyone, I want Registrar to check for the existence of SIP account ( wheither he=20 is already registered or not yet ). Could you pls tell me which file and what line number ? Thanks for your help in advance. Kolneath SOMETH " Ce courriel et les documents qui y sont attaches peuvent contenir des inf= ormations confidentielles. Si vous n'etes pas le destinataire escompte, me= rci d'en informer l'expediteur immediatement et de detruire ce courriel ai= nsi que tous les documents attaches de votre systeme informatique. Toute di= vulgation, distribution ou copie du present courriel et des documents attac= hes sans autorisation prealable de son emetteur est interdite."=20 " This e-mail and any attached documents may contain confidential or propri= etary information. If you are not the intended recipient, please advise the= sender immediately and delete this e-mail and all attached documents from = your computer system. Any unauthorised disclosure, distribution or copying = hereof is prohibited." |
From: GCC <vir...@in...> - 2007-08-28 04:34:31
|
Well... You can find many useful functions at SoftPhoneInterface.cxx. To develop an UA, maybe you can download MFC version of OSSPhone in ATLSIP and study the source. Hope my little suggestion helps. Regards, GCC |
From: <jo...@op...> - 2007-08-28 01:47:30
|
Hi Everyone, I have recently added a new trunk in OpenSBC called CALEA trunk (refer to SBCCALEATrunk.cxx). What this is, is a separate trunk that handles processing of packets for lawful intercep as mandated in CALEA (Communications Assistance for Law Enforcement Act). Take note that this code is in its infancy and is not complete yet. As of the moment, only the capability to save sip packets to pcap format is in place (rtp packets to follow). Slowly the CALEA trunk will evolve into a fully functional LEA collector. To test, look for the new section in http admin name "CALEA Calls". You will find a text box here name CALEA Filter. This should contain a set of comma delimited SIP URIs that you want to intercept (wild cards accepted). Example: sip:1111*, sip:2222* All calls beginning with 1111 and 2222 will be intercepted by the CALEA trunk. The PCAP dump will be saved in $(OPENSBCDIR)/calea/pcap directory. CAVEAT: There is still a bug in PCAFile.cxx that wrongly computes the ip checksum of the packets. This will show in ethereal. This error is considered benign. |
From: voice <vo...@ne...> - 2007-08-27 16:11:45
|
Hi with respect to 'Relay Routes' and 'Upper Registration Routes ' set to [sip:sipx.network.net*] sip:sipx.network.net:5060 Would this be the correct way to setup Relay Routes sip:200.sipx.network.net:5060 |
From: voice <vo...@ne...> - 2007-08-27 16:08:23
|
Hi With respect to 'B2BUA Routes' settings. if the ATA's DID is 7739050169 and the 'Upper Registration Routes' is set to [sip:sipx.network.net*] sip:sipx.network.net:5060 Would this be the correct way to set B2BUA Route? sip:7739050169.sipx.network.net:5060 |
From: voice <vo...@ne...> - 2007-08-27 16:02:40
|
Hi I can see from OpenSBC Registration Status this AOR Expires Call-ID 200@69.38.187.241 sip:200@10.10.10.2:5060 845 1343322131@10.10.10.2 20...@si... sip:200@10.10.10.2:5060 845 1343322131@10.10.10.2 where is the 10.10.10.2:5060 ipaddr in the expires set and why the DHCP addressing when the openSBC is sitting on a public IP? also where does the call-id 1343322131@10.10.10.2 number and 10.10.10.2 domain being set? It is reading some status value but from where are they set. r |
From: voice <vo...@ne...> - 2007-08-27 13:33:01
|
Hi joegen Thank you very much. joegen wrote > Solegy (Sponsor Company of OpenSBC) has a similar setup. In fact > B2BUpperReg mode came into life when we started to evaluate sipXecs. How do i contact Solegy? Do they still have those test boxes available? Do you know what version of sipX they are/were using? r ----- Original Message ----- From: "jo...@op..." <joe...@gm...> To: <ope...@li...> Sent: Sunday, August 26, 2007 10:27 PM Subject: Re: [OpenSIPStack] Upper Registration Routes > > voice wrote: > > Hi > > > > > > > > Can someone check my logic and setup? > > > > > > > > I am using a Cisco ATA-186 in atasip 3.1.0 firmware with a public IPaddr > > (69.x.x.x). > > > > I am setting the OutBoundProxy on the ATAs' to my openSBC name/ipaddr == > > sbc.netxxxxxx.net with a public IP Address 69.x.x.50. > > > > The ATA is registered in sipXecs 3.9.x also with a public IPaddr 69.x.x.40. > > > > > > > > I am also using but not in this test several Cisco 26xx routers run nat/PAT. > > These routers are in front several private networks. The public side of the > > NAT are connected to transceivers (69.x.x.x). For testing both boxes are > > hanging out on the Internet. No firewalls, Nat nothing to get in the way > > etc... > > > > > > > > I would like to also have remote Internet (outside my network) ATA/IPPhones > > register to the sipXecs. > > > > > > > > On the openSBC I have set 'Upper Registration Routes' to point to the > > sipxecs. This is my string: > > > > [sip:*@sbc.network.net*] sip:sipx.network.net:5060. > > > > Using this approach will local and possibly the remote ATA be able to > > register via the openSBC? > > > > This should be possible. The trick is, the domain sip:sbc.network.net > should not resolve as the local address of OpenSBC. If > sip:sbc.network.net is a local address, then OpenSBC would try to be > authoritative over the registrations. The correct format would be: > > [sip:sipx.network.net*] sip:sipx.network.net:5060 > > > > > > > > > Using openSBC in this way do should i set SBC Mode something other then > > 'Full Mode'? > > > > > > I am suggesting B2BUpperReg Mode. This would limit the work of OpenSBC > to b2bua and upper reg. Full mode includes the proxy functionality. > > > > > > When using openSBC using 'Upper registration Routes' is there a basic set > > of setting i should be concerned with at the exclusion of other settings? > > > > > > > > No. B2BUpperReg routes are just static routes for OpenSBC to follow in > case the remote domain is not resolvable via DNS. In B2BUpperReg mode, > you don't even have to put anything in the routes as long as the To-URI > resolves to a remote domain. > > > > > > > > > Are there any sipXecs folks on this list with similar issues' to mine? > > > > I would be willing to open my test sipxecs and opensbc box's to further > > development challenges in exchange for help setting up both boxes to > > intergrate sipXecs and openSBC.... > > > > > > > > > Solegy (Sponsor Company of OpenSBC) has a similar setup. In fact > B2BUpperReg mode came into life when we started to evaluate sipXecs. > > > > > r > > > > > > > > ------------------------------------------------------------------------- > > 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: voice <vo...@ne...> - 2007-08-27 13:20:17
|
Hi So your advice is to leave 'Internal DNS Mapping' blank if sipX domain is a 'FQDN' on the primary DNS? r ----- Original Message ----- From: "jo...@op..." <joe...@gm...> To: <ope...@li...> Sent: Sunday, August 26, 2007 10:31 PM Subject: Re: [OpenSIPStack] Internal DNS Mapping > voice wrote: > > Hi > > > > We run our own primary and secondary DNS. with a public add of 69.x.x.x. > > For security reasons we don't use the SRV feature so our pbx resolves to a > > full FQDN. i.e. @sipx.network.net and not @network.net. > > > > Will this setup interferre with openSBC resolving DNS. > > > > Should i set openSBCs' 'Internal DNS Mapping' to our primary DNS or should i > > leave it blank? > > > > > > Internal DNS Mapping is there so that you could fool OpenSBC that a > doamin is actually an FQDN even though it doesn't have a DNS record for > it. If you internal and external DNS only affects this particular > domain, I think you should use the internal DNS as with 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 |
From: <jo...@op...> - 2007-08-27 03:31:07
|
voice wrote: > Hi > > We run our own primary and secondary DNS. with a public add of 69.x.x.x. > For security reasons we don't use the SRV feature so our pbx resolves to a > full FQDN. i.e. @sipx.network.net and not @network.net. > > Will this setup interferre with openSBC resolving DNS. > > Should i set openSBCs' 'Internal DNS Mapping' to our primary DNS or should i > leave it blank? > > Internal DNS Mapping is there so that you could fool OpenSBC that a doamin is actually an FQDN even though it doesn't have a DNS record for it. If you internal and external DNS only affects this particular domain, I think you should use the internal DNS as with OpenSBC |
From: <jo...@op...> - 2007-08-27 03:28:06
|
voice wrote: > Hi > > > > Can someone check my logic and setup? > > > > I am using a Cisco ATA-186 in atasip 3.1.0 firmware with a public IPaddr > (69.x.x.x). > > I am setting the OutBoundProxy on the ATAs' to my openSBC name/ipaddr == > sbc.netxxxxxx.net with a public IP Address 69.x.x.50. > > The ATA is registered in sipXecs 3.9.x also with a public IPaddr 69.x.x.40. > > > > I am also using but not in this test several Cisco 26xx routers run nat/PAT. > These routers are in front several private networks. The public side of the > NAT are connected to transceivers (69.x.x.x). For testing both boxes are > hanging out on the Internet. No firewalls, Nat nothing to get in the way > etc... > > > > I would like to also have remote Internet (outside my network) ATA/IPPhones > register to the sipXecs. > > > > On the openSBC I have set 'Upper Registration Routes' to point to the > sipxecs. This is my string: > > [sip:*@sbc.network.net*] sip:sipx.network.net:5060. > > Using this approach will local and possibly the remote ATA be able to > register via the openSBC? > This should be possible. The trick is, the domain sip:sbc.network.net should not resolve as the local address of OpenSBC. If sip:sbc.network.net is a local address, then OpenSBC would try to be authoritative over the registrations. The correct format would be: [sip:sipx.network.net*] sip:sipx.network.net:5060 > > > Using openSBC in this way do should i set SBC Mode something other then > 'Full Mode'? > > I am suggesting B2BUpperReg Mode. This would limit the work of OpenSBC to b2bua and upper reg. Full mode includes the proxy functionality. > > When using openSBC using 'Upper registration Routes' is there a basic set > of setting i should be concerned with at the exclusion of other settings? > > > No. B2BUpperReg routes are just static routes for OpenSBC to follow in case the remote domain is not resolvable via DNS. In B2BUpperReg mode, you don't even have to put anything in the routes as long as the To-URI resolves to a remote domain. > > > Are there any sipXecs folks on this list with similar issues' to mine? > > I would be willing to open my test sipxecs and opensbc box's to further > development challenges in exchange for help setting up both boxes to > intergrate sipXecs and openSBC.... > > > Solegy (Sponsor Company of OpenSBC) has a similar setup. In fact B2BUpperReg mode came into life when we started to evaluate sipXecs. > r > > > > ------------------------------------------------------------------------- > 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...> - 2007-08-26 17:09:22
|
Hi We run our own primary and secondary DNS. with a public add of 69.x.x.x. For security reasons we don't use the SRV feature so our pbx resolves to a full FQDN. i.e. @sipx.network.net and not @network.net. Will this setup interferre with openSBC resolving DNS. Should i set openSBCs' 'Internal DNS Mapping' to our primary DNS or should i leave it blank? |
From: voice <vo...@ne...> - 2007-08-26 16:54:05
|
Hi Can someone check my logic and setup? I am using a Cisco ATA-186 in atasip 3.1.0 firmware with a public IPaddr (69.x.x.x). I am setting the OutBoundProxy on the ATAs' to my openSBC name/ipaddr == sbc.netxxxxxx.net with a public IP Address 69.x.x.50. The ATA is registered in sipXecs 3.9.x also with a public IPaddr 69.x.x.40. I am also using but not in this test several Cisco 26xx routers run nat/PAT. These routers are in front several private networks. The public side of the NAT are connected to transceivers (69.x.x.x). For testing both boxes are hanging out on the Internet. No firewalls, Nat nothing to get in the way etc... I would like to also have remote Internet (outside my network) ATA/IPPhones register to the sipXecs. On the openSBC I have set 'Upper Registration Routes' to point to the sipxecs. This is my string: [sip:*@sbc.network.net*] sip:sipx.network.net:5060. Using this approach will local and possibly the remote ATA be able to register via the openSBC? Using openSBC in this way do should i set SBC Mode something other then 'Full Mode'? When using openSBC using 'Upper registration Routes' is there a basic set of setting i should be concerned with at the exclusion of other settings? Are there any sipXecs folks on this list with similar issues' to mine? I would be willing to open my test sipxecs and opensbc box's to further development challenges in exchange for help setting up both boxes to intergrate sipXecs and openSBC.... r |
From: voice <vo...@ne...> - 2007-08-26 16:13:16
|
From: voice <vo...@ne...> - 2007-08-26 15:50:41
|
Hi Is there a how to for 'Upper Registration Routes' menu option? r |
From: voice <vo...@ne...> - 2007-08-25 20:44:59
|
Hi Joegen I have set up a public openSBC. Could this be helpful for you to use for testing? r ----- Original Message ----- From: "Joegen E. Baclor" <joe...@gm...> To: <ope...@li...> Sent: Wednesday, August 22, 2007 8:49 PM Subject: Re: [OpenSIPStack] last stable release? > Hi Last stable release is 1.1.4 (OpenSBC) 1.1.7 (OpenSIPStack). These > are available in the download sections. However CVS is worth a try. > We try our best to not to introduce new bugs in CVS that would affect > the previous released functionality. My advise is to go with CVS > > Joegen > > > voice wrote: > > Hi > > > > Which version is the last stable release of opensipstack for Linux/Centos5? > > > > ------------------------------------------------------------------------- > > 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: voice <vo...@ne...> - 2007-08-25 20:13:13
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Hi All Some where in your lit i found this 'SIP egress and ingress trunking among others'. Is there a doc somewhere that explains how to do this. r |
From: voice <vo...@ne...> - 2007-08-25 19:55:24
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Hi Joegen Terminology is reality. My PSTN provider provides an ipaddr on the internet. I am calling this a SIP Trunk gateway. It expects no SBC or encryption just plain everyday SIP from me. They use peer-to-peer IP addressing for security. So this setup may not fit neatly within the definition of 'SIP Trucking'. Sipx clearly can't help me. So i am turning to openSBC as a possible solution. I will point the sipX 'sip trunk' to my opensbc box and setup the openSBC to handle the sip protocol. Anyway i'm in the experimental business at the moment so i have nothing to loose. r ----- Original Message ----- From: "Joegen E. Baclor" <joe...@gm...> To: <ope...@li...> Sent: Wednesday, August 22, 2007 8:52 PM Subject: Re: [OpenSIPStack] sipXecs and openSBC > voice wrote: > > Hi > > > > I have setup a sipXecs box and a standalone openSBC box. > > > > i want to use the sipX 'SIP Trunk' Gateway to define and point to openSBC from sipX. > > > > My goal is to use sipX to register USERS and PHONES and to use openSBC to pass SiP traffic to remote SBC, which will handle DID < == > PSTN > > > > ATA <==> sipX <==> openSBC <==> SBC/PSTN > > > > Can this be done with openSBC in the middle? > > > > How do i set up openSBC to do 'SIP egress and ingress TRUNKing'? > > > > > If you are pertaining to the SIP Trunking being discussed in the sipX > user list, then this functionality is not yet complete. However, > OpenSBC can already be used for basic trunking needs like traversing NAT > and routing to your carrier as a B2BUA. If your carrier needs to do a > separate authentication, then you must wait till I announce the > trunking feature completion. This will happen soon so watch this space. > > > > ------------------------------------------------------------------------- > 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: Yacine A. <yac...@ms...> - 2007-08-25 10:28:09
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Hi, This is the code that answer the call and the one that make a call i'm not sending a makecall direcetly after selecting a number. if m_CallState=3D'CallingIn' then begin try OpenSIPStackCtl1.AnswerCall; exit; except end; end; if length(cboSIPURI.Text)=3D0 then exit; try m_CallState :=3D 'CallingOut'; status.Caption:=3D'Dialing '+cboSIPURI.Text+' ...'; with form_settings do begin OpenSIPStackCtl1.AccountName :=3D txtUserId.Text; OpenSIPStackCtl1.AuthenticationUser :=3D txtAccountId.Text; OpenSIPStackCtl1.AuthenticationPassword :=3D txtPassword.Text; OpenSIPStackCtl1.SIPServerAddress :=3D txtRegistrar.Text; OpenSIPStackCtl1.AccountAddress:=3DtxtUserId.Text+'@'+txtRegistrar.Text; OpenSIPStackCtl1.proxyAuthenticationUser:=3DtxtAccountId.Text; OpenSIPStackCtl1.proxyAuthenticationPassword:=3DtxtPassword.Text; OpenSIPStackCtl1.InitializeSIP; OpenSIPStackCtl1.DisableSilenceDetection; OpenSIPStackCtl1.MakeCall( 'sip:'+cboSIPURI.Text+'@'+txtRegistrar.Text ); end; except status.Caption:=3D'Can not make call'; > Date: Thu, 23 Aug 2007 20:25:27 +0800 > From: ip...@so... > To: ope...@li... > Subject: Re: [OpenSIPStack] ATLSIP crashes >=20 > Can you send me the code snippet that handles answering a call (the one=20 > that handles the button click)? I hope Delphi isn't too alien for me.=20 > Never really worked with it before. ;) >=20 > I'm also guessing that selecting a dialed number will automatically call= =20 > that number. Is this correct? >=20 > - Ilian >=20 > Yacine Auczone wrote: > > Hi, > > I'm not using OSSPhone .net > > i'm using a simple softphone developed using delphi calling the ATLSIP = dll > > > > For the dialed numbers drop down it does not occure everytime. > > > > > > > > > > =20 > > <snip> > > > > > > =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 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 _________________________________________________________________ T=E9l=E9chargez gratuitement l'=E9conomiseur d'=E9cran Les Simson Le film ! http://specials.divertissements.fr.msn.com/lessimpsonlefilm/economiseur.asp= x= |